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

   1 /*
   2  * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
   3  *
   4  * This file is part of libswresample
   5  *
   6  * libswresample is free software; you can redistribute it and/or
   7  * modify it under the terms of the GNU Lesser General Public
   8  * License as published by the Free Software Foundation; either
   9  * version 2.1 of the License, or (at your option) any later version.
  10  *
  11  * libswresample is distributed in the hope that it will be useful,
  12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  14  * Lesser General Public License for more details.
  15  *
  16  * You should have received a copy of the GNU Lesser General Public
  17  * License along with libswresample; if not, write to the Free Software
  18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  19  */
  20
  21 #include "libavutil/opt.h"
  22 #include "swresample_internal.h"
  23 #include "audioconvert.h"
  24 #include "libavutil/avassert.h"
  25 #include "libavutil/channel_layout.h"
  26
  27 #include <float.h>
  28
  29 #define  C30DB  M_SQRT2
  30 #define  C15DB  1.189207115
  31 #define C__0DB  1.0
  32 #define C_15DB  0.840896415
  33 #define C_30DB  M_SQRT1_2
  34 #define C_45DB  0.594603558
  35 #define C_60DB  0.5
  36
  37 #define ALIGN 32
  38
  39 //TODO split options array out?
  40 #define OFFSET(x) offsetof(SwrContext,x)
  41 #define PARAM AV_OPT_FLAG_AUDIO_PARAM
  42
  43 static const AVOption options[]={
  44 {"ich"                  , "set input channel count"     , OFFSET( in.ch_count   ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
  45 {"in_channel_count"     , "set input channel count"     , OFFSET( in.ch_count   ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
  46 {"och"                  , "set output channel count"    , OFFSET(out.ch_count   ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
  47 {"out_channel_count"    , "set output channel count"    , OFFSET(out.ch_count   ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
  48 {"uch"                  , "set used channel count"      , OFFSET(used_ch_count  ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
  49 {"used_channel_count"   , "set used channel count"      , OFFSET(used_ch_count  ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
  50 {"isr"                  , "set input sample rate"       , OFFSET( in_sample_rate), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , INT_MAX   , PARAM},
  51 {"in_sample_rate"       , "set input sample rate"       , OFFSET( in_sample_rate), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , INT_MAX   , PARAM},
  52 {"osr"                  , "set output sample rate"      , OFFSET(out_sample_rate), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , INT_MAX   , PARAM},
  53 {"out_sample_rate"      , "set output sample rate"      , OFFSET(out_sample_rate), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , INT_MAX   , PARAM},
  54 {"isf"                  , "set input sample format"     , OFFSET( in_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , INT_MAX, PARAM},
  55 {"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},
  56 {"osf"                  , "set output sample format"    , OFFSET(out_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , INT_MAX, PARAM},
  57 {"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},
  58 {"tsf"                  , "set internal sample format"  , OFFSET(int_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , INT_MAX, PARAM},
  59 {"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},
  60 {"icl"                  , "set input channel layout"    , OFFSET( in_ch_layout  ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
  61 {"in_channel_layout"    , "set input channel layout"    , OFFSET( in_ch_layout  ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
  62 {"ocl"                  , "set output channel layout"   , OFFSET(out_ch_layout  ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
  63 {"out_channel_layout"   , "set output channel layout"   , OFFSET(out_ch_layout  ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
  64 {"clev"                 , "set center mix level"        , OFFSET(clev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
  65 {"center_mix_level"     , "set center mix level"        , OFFSET(clev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
  66 {"slev"                 , "set surround mix level"      , OFFSET(slev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
  67 {"surround_mix_level"   , "set surround mix Level"      , OFFSET(slev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
  68 {"lfe_mix_level"        , "set LFE mix level"           , OFFSET(lfe_mix_level  ), AV_OPT_TYPE_FLOAT, {.dbl=0                     }, -32    , 32        , PARAM},
  69 {"rmvol"                , "set rematrix volume"         , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0                   }, -1000  , 1000      , PARAM},
  70 {"rematrix_volume"      , "set rematrix volume"         , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0                   }, -1000  , 1000      , PARAM},
  71 {"rematrix_maxval"      , "set rematrix maxval"         , OFFSET(rematrix_maxval), AV_OPT_TYPE_FLOAT, {.dbl=0.0                   }, 0      , 1000      , PARAM},
  72
  73 {"flags"                , "set flags"                   , OFFSET(flags          ), AV_OPT_TYPE_FLAGS, {.i64=0                     }, 0      , UINT_MAX  , PARAM, "flags"},
  74 {"swr_flags"            , "set flags"                   , OFFSET(flags          ), AV_OPT_TYPE_FLAGS, {.i64=0                     }, 0      , UINT_MAX  , PARAM, "flags"},
  75 {"res"                  , "force resampling"            , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_FLAG_RESAMPLE     }, INT_MIN, INT_MAX   , PARAM, "flags"},
  76
  77 {"dither_scale"         , "set dither scale"            , OFFSET(dither.scale   ), AV_OPT_TYPE_FLOAT, {.dbl=1                     }, 0      , INT_MAX   , PARAM},
  78
  79 {"dither_method"        , "set dither method"           , OFFSET(dither.method  ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_DITHER_NB-1, PARAM, "dither_method"},
  80 {"rectangular"          , "select rectangular dither"   , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_RECTANGULAR}, INT_MIN, INT_MAX   , PARAM, "dither_method"},
  81 {"triangular"           , "select triangular dither"    , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX   , PARAM, "dither_method"},
  82 {"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"},
  83 {"lipshitz"             , "select lipshitz noise shaping dither" , 0             , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_LIPSHITZ}, INT_MIN, INT_MAX, PARAM, "dither_method"},
  84 {"shibata"              , "select shibata noise shaping dither" , 0              , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"},
  85 {"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"},
  86 {"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"},
  87 {"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"},
  88 {"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"},
  89 {"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"},
  90
  91 {"filter_size"          , "set swr resampling filter size", OFFSET(filter_size)  , AV_OPT_TYPE_INT  , {.i64=32                    }, 0      , INT_MAX   , PARAM },
  92 {"phase_shift"          , "set swr resampling phase shift", OFFSET(phase_shift)  , AV_OPT_TYPE_INT  , {.i64=10                    }, 0      , 24        , PARAM },
  93 {"linear_interp"        , "enable linear interpolation" , OFFSET(linear_interp)  , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , 1         , PARAM },
  94 {"cutoff"               , "set cutoff frequency ratio"  , OFFSET(cutoff)         , AV_OPT_TYPE_DOUBLE,{.dbl=0.                    }, 0      , 1         , PARAM },
  95
  96 /* duplicate option in order to work with avconv */
  97 {"resample_cutoff"      , "set cutoff frequency ratio"  , OFFSET(cutoff)         , AV_OPT_TYPE_DOUBLE,{.dbl=0.                    }, 0      , 1         , PARAM },
  98
  99 {"resampler"            , "set resampling Engine"       , OFFSET(engine)         , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_ENGINE_NB-1, PARAM, "resampler"},
 100 {"swr"                  , "select SW Resampler"         , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SWR        }, INT_MIN, INT_MAX   , PARAM, "resampler"},
 101 {"soxr"                 , "select SoX Resampler"        , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SOXR       }, INT_MIN, INT_MAX   , PARAM, "resampler"},
 102 {"precision"            , "set soxr resampling precision (in bits)"
 103                                                         , OFFSET(precision)      , AV_OPT_TYPE_DOUBLE,{.dbl=20.0                  }, 15.0   , 33.0      , PARAM },
 104 {"cheby"                , "enable soxr Chebyshev passband & higher-precision irrational ratio approximation"
 105                                                         , OFFSET(cheby)          , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , 1         , PARAM },
 106 {"min_comp"             , "set minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"
 107                                                         , OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX               }, 0      , FLT_MAX   , PARAM },
 108 {"min_hard_comp"        , "set minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."
 109                                                         , OFFSET(min_hard_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0.1                   }, 0      , INT_MAX   , PARAM },
 110 {"comp_duration"        , "set duration (in seconds) over which data is stretched/squeezed to make it match the timestamps."
 111                                                         , OFFSET(soft_compensation_duration),AV_OPT_TYPE_FLOAT ,{.dbl=1                     }, 0      , INT_MAX   , PARAM },
 112 {"max_soft_comp"        , "set maximum factor by which data is stretched/squeezed to make it match the timestamps."
 113                                                         , OFFSET(max_soft_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0                     }, INT_MIN, INT_MAX   , PARAM },
 114 {"async"                , "simplified 1 parameter audio timestamp matching, 0(disabled), 1(filling and trimming), >1(maximum stretch/squeeze in samples per second)"
 115                                                         , OFFSET(async)          , AV_OPT_TYPE_FLOAT ,{.dbl=0                     }, INT_MIN, INT_MAX   , PARAM },
 116 {"first_pts"            , "Assume the first pts should be this value (in samples)."
 117                                                         , OFFSET(firstpts_in_samples), AV_OPT_TYPE_INT64 ,{.i64=AV_NOPTS_VALUE    }, INT64_MIN,INT64_MAX, PARAM },
 118
 119 { "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" },
 120     { "none",  "select none",               0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_NONE  }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
 121     { "dolby", "select Dolby",              0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
 122     { "dplii", "select Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
 123
 124 { "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" },
 125     { "cubic"           , "select cubic"                , 0                      , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC            }, INT_MIN, INT_MAX, PARAM, "filter_type" },
 126     { "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" },
 127     { "kaiser"          , "select Kaiser Windowed Sinc" , 0                      , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER           }, INT_MIN, INT_MAX, PARAM, "filter_type" },
 128
 129 { "kaiser_beta"         , "set swr Kaiser Window Beta"  , OFFSET(kaiser_beta)    , AV_OPT_TYPE_INT  , {.i64=9                     }, 2      , 16        , PARAM },
 130
 131 { "output_sample_bits"  , "set swr number of output sample bits", OFFSET(dither.output_sample_bits), AV_OPT_TYPE_INT  , {.i64=0   }, 0      , 64        , PARAM },
 132 {0}
 133 };
 134
 135 static const char* context_to_name(void* ptr) {
 136     return "SWR";
 137 }
 138
 139 static const AVClass av_class = {
 140     .class_name                = "SWResampler",
 141     .item_name                 = context_to_name,
 142     .option                    = options,
 143     .version                   = LIBAVUTIL_VERSION_INT,
 144     .log_level_offset_offset   = OFFSET(log_level_offset),
 145     .parent_log_context_offset = OFFSET(log_ctx),
 146     .category                  = AV_CLASS_CATEGORY_SWRESAMPLER,
 147 };
 148
 149 unsigned swresample_version(void)
 150 {
 151     av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
 152     return LIBSWRESAMPLE_VERSION_INT;
 153 }
 154
 155 const char *swresample_configuration(void)
 156 {
 157     return FFMPEG_CONFIGURATION;
 158 }
 159
 160 const char *swresample_license(void)
 161 {
 162 #define LICENSE_PREFIX "libswresample license: "
 163     return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
 164 }
 165
 166 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
 167     if(!s || s->in_convert) // s needs to be allocated but not initialized
 168         return AVERROR(EINVAL);
 169     s->channel_map = channel_map;
 170     return 0;
 171 }
 172
 173 const AVClass *swr_get_class(void)
 174 {
 175     return &av_class;
 176 }
 177
 178 av_cold struct SwrContext *swr_alloc(void){
 179     SwrContext *s= av_mallocz(sizeof(SwrContext));
 180     if(s){
 181         s->av_class= &av_class;
 182         av_opt_set_defaults(s);
 183     }
 184     return s;
 185 }
 186
 187 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
 188                                       int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
 189                                       int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
 190                                       int log_offset, void *log_ctx){
 191     if(!s) s= swr_alloc();
 192     if(!s) return NULL;
 193
 194     s->log_level_offset= log_offset;
 195     s->log_ctx= log_ctx;
 196
 197     av_opt_set_int(s, "ocl", out_ch_layout,   0);
 198     av_opt_set_int(s, "osf", out_sample_fmt,  0);
 199     av_opt_set_int(s, "osr", out_sample_rate, 0);
 200     av_opt_set_int(s, "icl", in_ch_layout,    0);
 201     av_opt_set_int(s, "isf", in_sample_fmt,   0);
 202     av_opt_set_int(s, "isr", in_sample_rate,  0);
 203     av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE,   0);
 204     av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);
 205     av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);
 206     av_opt_set_int(s, "uch", 0, 0);
 207     return s;
 208 }
 209
 210 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
 211     a->fmt   = fmt;
 212     a->bps   = av_get_bytes_per_sample(fmt);
 213     a->planar= av_sample_fmt_is_planar(fmt);
 214 }
 215
 216 static void free_temp(AudioData *a){
 217     av_free(a->data);
 218     memset(a, 0, sizeof(*a));
 219 }
 220
 221 static void clear_context(SwrContext *s){
 222     s->in_buffer_index= 0;
 223     s->in_buffer_count= 0;
 224     s->resample_in_constraint= 0;
 225     memset(s->in.ch, 0, sizeof(s->in.ch));
 226     memset(s->out.ch, 0, sizeof(s->out.ch));
 227     free_temp(&s->postin);
 228     free_temp(&s->midbuf);
 229     free_temp(&s->preout);
 230     free_temp(&s->in_buffer);
 231     free_temp(&s->silence);
 232     free_temp(&s->drop_temp);
 233     free_temp(&s->dither.noise);
 234     free_temp(&s->dither.temp);
 235     swri_audio_convert_free(&s-> in_convert);
 236     swri_audio_convert_free(&s->out_convert);
 237     swri_audio_convert_free(&s->full_convert);
 238     swri_rematrix_free(s);
 239
 240     s->flushed = 0;
 241 }
 242
 243 av_cold void swr_free(SwrContext **ss){
 244     SwrContext *s= *ss;
 245     if(s){
 246         clear_context(s);
 247         if (s->resampler)
 248             s->resampler->free(&s->resample);
 249     }
 250
 251     av_freep(ss);
 252 }
 253
 254 av_cold int swr_init(struct SwrContext *s){
 255     int ret;
 256
 257     clear_context(s);
 258
 259     if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
 260         av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
 261         return AVERROR(EINVAL);
 262     }
 263     if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
 264         av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
 265         return AVERROR(EINVAL);
 266     }
 267
 268     if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) {
 269         av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
 270         s->in_ch_layout = 0;
 271     }
 272
 273     if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) {
 274         av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
 275         s->out_ch_layout = 0;
 276     }
 277
 278     switch(s->engine){
 279 #if CONFIG_LIBSOXR
 280         extern struct Resampler const soxr_resampler;
 281         case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break;
 282 #endif
 283         case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
 284         default:
 285             av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
 286             return AVERROR(EINVAL);
 287     }
 288
 289     if(!s->used_ch_count)
 290         s->used_ch_count= s->in.ch_count;
 291
 292     if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
 293         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
 294         s-> in_ch_layout= 0;
 295     }
 296
 297     if(!s-> in_ch_layout)
 298         s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
 299     if(!s->out_ch_layout)
 300         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
 301
 302     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
 303                  s->rematrix_custom;
 304
 305     if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
 306         if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
 307             s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
 308         }else if(   av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
 309                  && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P
 310                  && !s->rematrix
 311                  && s->engine != SWR_ENGINE_SOXR){
 312             s->int_sample_fmt= AV_SAMPLE_FMT_S32P;
 313         }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
 314             s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
 315         }else{
 316             av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
 317             s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
 318         }
 319     }
 320
 321     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P
 322         &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
 323         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
 324         &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
 325         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));
 326         return AVERROR(EINVAL);
 327     }
 328
 329     set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
 330     set_audiodata_fmt(&s->out, s->out_sample_fmt);
 331
 332     if (s->firstpts_in_samples != AV_NOPTS_VALUE) {
 333         if (!s->async && s->min_compensation >= FLT_MAX/2)
 334             s->async = 1;
 335         s->firstpts =
 336         s->outpts   = s->firstpts_in_samples * s->out_sample_rate;
 337     } else
 338         s->firstpts = AV_NOPTS_VALUE;
 339
 340     if (s->async) {
 341         if (s->min_compensation >= FLT_MAX/2)
 342             s->min_compensation = 0.001;
 343         if (s->async > 1.0001) {
 344             s->max_soft_compensation = s->async / (double) s->in_sample_rate;
 345         }
 346     }
 347
 348     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
 349         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);
 350     }else
 351         s->resampler->free(&s->resample);
 352     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P
 353         && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
 354         && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
 355         && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
 356         && s->resample){
 357         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
 358         return -1;
 359     }
 360
 361 #define RSC 1 //FIXME finetune
 362     if(!s-> in.ch_count)
 363         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
 364     if(!s->used_ch_count)
 365         s->used_ch_count= s->in.ch_count;
 366     if(!s->out.ch_count)
 367         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
 368
 369     if(!s-> in.ch_count){
 370         av_assert0(!s->in_ch_layout);
 371         av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
 372         return -1;
 373     }
 374
 375     if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
 376         char l1[1024], l2[1024];
 377         av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
 378         av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
 379         av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
 380                "but there is not enough information to do it\n", l1, l2);
 381         return -1;
 382     }
 383
 384 av_assert0(s->used_ch_count);
 385 av_assert0(s->out.ch_count);
 386     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
 387
 388     s->in_buffer= s->in;
 389     s->silence  = s->in;
 390     s->drop_temp= s->out;
 391
 392     if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
 393         s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
 394                                                    s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
 395         return 0;
 396     }
 397
 398     s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
 399                                              s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
 400     s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
 401                                              s->int_sample_fmt, s->out.ch_count, NULL, 0);
 402
 403     if (!s->in_convert || !s->out_convert)
 404         return AVERROR(ENOMEM);
 405
 406     s->postin= s->in;
 407     s->preout= s->out;
 408     s->midbuf= s->in;
 409
 410     if(s->channel_map){
 411         s->postin.ch_count=
 412         s->midbuf.ch_count= s->used_ch_count;
 413         if(s->resample)
 414             s->in_buffer.ch_count= s->used_ch_count;
 415     }
 416     if(!s->resample_first){
 417         s->midbuf.ch_count= s->out.ch_count;
 418         if(s->resample)
 419             s->in_buffer.ch_count = s->out.ch_count;
 420     }
 421
 422     set_audiodata_fmt(&s->postin, s->int_sample_fmt);
 423     set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
 424     set_audiodata_fmt(&s->preout, s->int_sample_fmt);
 425
 426     if(s->resample){
 427         set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
 428     }
 429
 430     if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
 431         return ret;
 432
 433     if(s->rematrix || s->dither.method)
 434         return swri_rematrix_init(s);
 435
 436     return 0;
 437 }
 438
 439 int swri_realloc_audio(AudioData *a, int count){
 440     int i, countb;
 441     AudioData old;
 442
 443     if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
 444         return AVERROR(EINVAL);
 445
 446     if(a->count >= count)
 447         return 0;
 448
 449     count*=2;
 450
 451     countb= FFALIGN(count*a->bps, ALIGN);
 452     old= *a;
 453
 454     av_assert0(a->bps);
 455     av_assert0(a->ch_count);
 456
 457     a->data= av_mallocz(countb*a->ch_count);
 458     if(!a->data)
 459         return AVERROR(ENOMEM);
 460     for(i=0; i<a->ch_count; i++){
 461         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
 462         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
 463     }
 464     if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
 465     av_freep(&old.data);
 466     a->count= count;
 467
 468     return 1;
 469 }
 470
 471 static void copy(AudioData *out, AudioData *in,
 472                  int count){
 473     av_assert0(out->planar == in->planar);
 474     av_assert0(out->bps == in->bps);
 475     av_assert0(out->ch_count == in->ch_count);
 476     if(out->planar){
 477         int ch;
 478         for(ch=0; ch<out->ch_count; ch++)
 479             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
 480     }else
 481         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
 482 }
 483
 484 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
 485     int i;
 486     if(!in_arg){
 487         memset(out->ch, 0, sizeof(out->ch));
 488     }else if(out->planar){
 489         for(i=0; i<out->ch_count; i++)
 490             out->ch[i]= in_arg[i];
 491     }else{
 492         for(i=0; i<out->ch_count; i++)
 493             out->ch[i]= in_arg[0] + i*out->bps;
 494     }
 495 }
 496
 497 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
 498     int i;
 499     if(out->planar){
 500         for(i=0; i<out->ch_count; i++)
 501             in_arg[i]= out->ch[i];
 502     }else{
 503         in_arg[0]= out->ch[0];
 504     }
 505 }
 506
 507 /**
 508  *
 509  * out may be equal in.
 510  */
 511 static void buf_set(AudioData *out, AudioData *in, int count){
 512     int ch;
 513     if(in->planar){
 514         for(ch=0; ch<out->ch_count; ch++)
 515             out->ch[ch]= in->ch[ch] + count*out->bps;
 516     }else{
 517         for(ch=out->ch_count-1; ch>=0; ch--)
 518             out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
 519     }
 520 }
 521
 522 /**
 523  *
 524  * @return number of samples output per channel
 525  */
 526 static int resample(SwrContext *s, AudioData *out_param, int out_count,
 527                              const AudioData * in_param, int in_count){
 528     AudioData in, out, tmp;
 529     int ret_sum=0;
 530     int border=0;
 531     int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
 532
 533     av_assert1(s->in_buffer.ch_count == in_param->ch_count);
 534     av_assert1(s->in_buffer.planar   == in_param->planar);
 535     av_assert1(s->in_buffer.fmt      == in_param->fmt);
 536
 537     tmp=out=*out_param;
 538     in =  *in_param;
 539
 540     do{
 541         int ret, size, consumed;
 542         if(!s->resample_in_constraint && s->in_buffer_count){
 543             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 544             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
 545             out_count -= ret;
 546             ret_sum += ret;
 547             buf_set(&out, &out, ret);
 548             s->in_buffer_count -= consumed;
 549             s->in_buffer_index += consumed;
 550
 551             if(!in_count)
 552                 break;
 553             if(s->in_buffer_count <= border){
 554                 buf_set(&in, &in, -s->in_buffer_count);
 555                 in_count += s->in_buffer_count;
 556                 s->in_buffer_count=0;
 557                 s->in_buffer_index=0;
 558                 border = 0;
 559             }
 560         }
 561
 562         if((s->flushed || in_count > padless) && !s->in_buffer_count){
 563             s->in_buffer_index=0;
 564             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
 565             out_count -= ret;
 566             ret_sum += ret;
 567             buf_set(&out, &out, ret);
 568             in_count -= consumed;
 569             buf_set(&in, &in, consumed);
 570         }
 571
 572         //TODO is this check sane considering the advanced copy avoidance below
 573         size= s->in_buffer_index + s->in_buffer_count + in_count;
 574         if(   size > s->in_buffer.count
 575            && s->in_buffer_count + in_count <= s->in_buffer_index){
 576             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 577             copy(&s->in_buffer, &tmp, s->in_buffer_count);
 578             s->in_buffer_index=0;
 579         }else
 580             if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
 581                 return ret;
 582
 583         if(in_count){
 584             int count= in_count;
 585             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
 586
 587             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
 588             copy(&tmp, &in, /*in_*/count);
 589             s->in_buffer_count += count;
 590             in_count -= count;
 591             border += count;
 592             buf_set(&in, &in, count);
 593             s->resample_in_constraint= 0;
 594             if(s->in_buffer_count != count || in_count)
 595                 continue;
 596             if (padless) {
 597                 padless = 0;
 598                 continue;
 599             }
 600         }
 601         break;
 602     }while(1);
 603
 604     s->resample_in_constraint= !!out_count;
 605
 606     return ret_sum;
 607 }
 608
 609 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
 610                                                       AudioData *in , int  in_count){
 611     AudioData *postin, *midbuf, *preout;
 612     int ret/*, in_max*/;
 613     AudioData preout_tmp, midbuf_tmp;
 614
 615     if(s->full_convert){
 616         av_assert0(!s->resample);
 617         swri_audio_convert(s->full_convert, out, in, in_count);
 618         return out_count;
 619     }
 620
 621 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
 622 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
 623
 624     if((ret=swri_realloc_audio(&s->postin, in_count))<0)
 625         return ret;
 626     if(s->resample_first){
 627         av_assert0(s->midbuf.ch_count == s->used_ch_count);
 628         if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
 629             return ret;
 630     }else{
 631         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
 632         if((ret=swri_realloc_audio(&s->midbuf,  in_count))<0)
 633             return ret;
 634     }
 635     if((ret=swri_realloc_audio(&s->preout, out_count))<0)
 636         return ret;
 637
 638     postin= &s->postin;
 639
 640     midbuf_tmp= s->midbuf;
 641     midbuf= &midbuf_tmp;
 642     preout_tmp= s->preout;
 643     preout= &preout_tmp;
 644
 645     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
 646         postin= in;
 647
 648     if(s->resample_first ? !s->resample : !s->rematrix)
 649         midbuf= postin;
 650
 651     if(s->resample_first ? !s->rematrix : !s->resample)
 652         preout= midbuf;
 653
 654     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
 655        && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){
 656         if(preout==in){
 657             out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
 658             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
 659             copy(out, in, out_count);
 660             return out_count;
 661         }
 662         else if(preout==postin) preout= midbuf= postin= out;
 663         else if(preout==midbuf) preout= midbuf= out;
 664         else                    preout= out;
 665     }
 666
 667     if(in != postin){
 668         swri_audio_convert(s->in_convert, postin, in, in_count);
 669     }
 670
 671     if(s->resample_first){
 672         if(postin != midbuf)
 673             out_count= resample(s, midbuf, out_count, postin, in_count);
 674         if(midbuf != preout)
 675             swri_rematrix(s, preout, midbuf, out_count, preout==out);
 676     }else{
 677         if(postin != midbuf)
 678             swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
 679         if(midbuf != preout)
 680             out_count= resample(s, preout, out_count, midbuf, in_count);
 681     }
 682
 683     if(preout != out && out_count){
 684         AudioData *conv_src = preout;
 685         if(s->dither.method){
 686             int ch;
 687             int dither_count= FFMAX(out_count, 1<<16);
 688
 689             if (preout == in) {
 690                 conv_src = &s->dither.temp;
 691                 if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
 692                     return ret;
 693             }
 694
 695             if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
 696                 return ret;
 697             if(ret)
 698                 for(ch=0; ch<s->dither.noise.ch_count; ch++)
 699                     swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);
 700             av_assert0(s->dither.noise.ch_count == preout->ch_count);
 701
 702             if(s->dither.noise_pos + out_count > s->dither.noise.count)
 703                 s->dither.noise_pos = 0;
 704
 705             if (s->dither.method < SWR_DITHER_NS){
 706                 if (s->mix_2_1_simd) {
 707                     int len1= out_count&~15;
 708                     int off = len1 * preout->bps;
 709
 710                     if(len1)
 711                         for(ch=0; ch<preout->ch_count; ch++)
 712                             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);
 713                     if(out_count != len1)
 714                         for(ch=0; ch<preout->ch_count; ch++)
 715                             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);
 716                 } else {
 717                     for(ch=0; ch<preout->ch_count; ch++)
 718                         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);
 719                 }
 720             } else {
 721                 switch(s->int_sample_fmt) {
 722                 case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
 723                 case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
 724                 case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
 725                 case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
 726                 }
 727             }
 728             s->dither.noise_pos += out_count;
 729         }
 730 //FIXME packed doesn't need more than 1 chan here!
 731         swri_audio_convert(s->out_convert, out, conv_src, out_count);
 732     }
 733     return out_count;
 734 }
 735
 736 int swr_is_initialized(struct SwrContext *s) {
 737     return !!s->in_buffer.ch_count;
 738 }
 739
 740 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
 741                                 const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
 742     AudioData * in= &s->in;
 743     AudioData *out= &s->out;
 744
 745     if (!swr_is_initialized(s)) {
 746         av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
 747         return AVERROR(EINVAL);
 748     }
 749
 750     while(s->drop_output > 0){
 751         int ret;
 752         uint8_t *tmp_arg[SWR_CH_MAX];
 753 #define MAX_DROP_STEP 16384
 754         if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)
 755             return ret;
 756
 757         reversefill_audiodata(&s->drop_temp, tmp_arg);
 758         s->drop_output *= -1; //FIXME find a less hackish solution
 759         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
 760         s->drop_output *= -1;
 761         in_count = 0;
 762         if(ret>0) {
 763             s->drop_output -= ret;
 764             continue;
 765         }
 766
 767         if(s->drop_output || !out_arg)
 768             return 0;
 769     }
 770
 771     if(!in_arg){
 772         if(s->resample){
 773             if (!s->flushed)
 774                 s->resampler->flush(s);
 775             s->resample_in_constraint = 0;
 776             s->flushed = 1;
 777         }else if(!s->in_buffer_count){
 778             return 0;
 779         }
 780     }else
 781         fill_audiodata(in ,  (void*)in_arg);
 782
 783     fill_audiodata(out, out_arg);
 784
 785     if(s->resample){
 786         int ret = swr_convert_internal(s, out, out_count, in, in_count);
 787         if(ret>0 && !s->drop_output)
 788             s->outpts += ret * (int64_t)s->in_sample_rate;
 789         return ret;
 790     }else{
 791         AudioData tmp= *in;
 792         int ret2=0;
 793         int ret, size;
 794         size = FFMIN(out_count, s->in_buffer_count);
 795         if(size){
 796             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 797             ret= swr_convert_internal(s, out, size, &tmp, size);
 798             if(ret<0)
 799                 return ret;
 800             ret2= ret;
 801             s->in_buffer_count -= ret;
 802             s->in_buffer_index += ret;
 803             buf_set(out, out, ret);
 804             out_count -= ret;
 805             if(!s->in_buffer_count)
 806                 s->in_buffer_index = 0;
 807         }
 808
 809         if(in_count){
 810             size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
 811
 812             if(in_count > out_count) { //FIXME move after swr_convert_internal
 813                 if(   size > s->in_buffer.count
 814                 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
 815                     buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 816                     copy(&s->in_buffer, &tmp, s->in_buffer_count);
 817                     s->in_buffer_index=0;
 818                 }else
 819                     if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
 820                         return ret;
 821             }
 822
 823             if(out_count){
 824                 size = FFMIN(in_count, out_count);
 825                 ret= swr_convert_internal(s, out, size, in, size);
 826                 if(ret<0)
 827                     return ret;
 828                 buf_set(in, in, ret);
 829                 in_count -= ret;
 830                 ret2 += ret;
 831             }
 832             if(in_count){
 833                 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
 834                 copy(&tmp, in, in_count);
 835                 s->in_buffer_count += in_count;
 836             }
 837         }
 838         if(ret2>0 && !s->drop_output)
 839             s->outpts += ret2 * (int64_t)s->in_sample_rate;
 840         return ret2;
 841     }
 842 }
 843
 844 int swr_drop_output(struct SwrContext *s, int count){
 845     s->drop_output += count;
 846
 847     if(s->drop_output <= 0)
 848         return 0;
 849
 850     av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
 851     return swr_convert(s, NULL, s->drop_output, NULL, 0);
 852 }
 853
 854 int swr_inject_silence(struct SwrContext *s, int count){
 855     int ret, i;
 856     uint8_t *tmp_arg[SWR_CH_MAX];
 857
 858     if(count <= 0)
 859         return 0;
 860
 861 #define MAX_SILENCE_STEP 16384
 862     while (count > MAX_SILENCE_STEP) {
 863         if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
 864             return ret;
 865         count -= MAX_SILENCE_STEP;
 866     }
 867
 868     if((ret=swri_realloc_audio(&s->silence, count))<0)
 869         return ret;
 870
 871     if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
 872         memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
 873     } else
 874         memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
 875
 876     reversefill_audiodata(&s->silence, tmp_arg);
 877     av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
 878     ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
 879     return ret;
 880 }
 881
 882 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
 883     if (s->resampler && s->resample){
 884         return s->resampler->get_delay(s, base);
 885     }else{
 886         return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
 887     }
 888 }
 889
 890 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
 891     int ret;
 892
 893     if (!s || compensation_distance < 0)
 894         return AVERROR(EINVAL);
 895     if (!compensation_distance && sample_delta)
 896         return AVERROR(EINVAL);
 897     if (!s->resample) {
 898         s->flags |= SWR_FLAG_RESAMPLE;
 899         ret = swr_init(s);
 900         if (ret < 0)
 901             return ret;
 902     }
 903     if (!s->resampler->set_compensation){
 904         return AVERROR(EINVAL);
 905     }else{
 906         return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
 907     }
 908 }
 909
 910 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
 911     if(pts == INT64_MIN)
 912         return s->outpts;
 913
 914     if (s->firstpts == AV_NOPTS_VALUE)
 915         s->outpts = s->firstpts = pts;
 916
 917     if(s->min_compensation >= FLT_MAX) {
 918         return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
 919     } else {
 920         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;
 921         double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
 922
 923         if(fabs(fdelta) > s->min_compensation) {
 924             if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
 925                 int ret;
 926                 if(delta > 0) ret = swr_inject_silence(s,  delta / s->out_sample_rate);
 927                 else          ret = swr_drop_output   (s, -delta / s-> in_sample_rate);
 928                 if(ret<0){
 929                     av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
 930                 }
 931             } else if(s->soft_compensation_duration && s->max_soft_compensation) {
 932                 int duration = s->out_sample_rate * s->soft_compensation_duration;
 933                 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
 934                 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
 935                 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
 936                 swr_set_compensation(s, comp, duration);
 937             }
 938         }
 939
 940         return s->outpts;
 941     }
 942 }