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

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