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 av_cold void swr_free(SwrContext **ss){
 222     SwrContext *s= *ss;
 223     if(s){
 224         free_temp(&s->postin);
 225         free_temp(&s->midbuf);
 226         free_temp(&s->preout);
 227         free_temp(&s->in_buffer);
 228         free_temp(&s->silence);
 229         free_temp(&s->drop_temp);
 230         free_temp(&s->dither.noise);
 231         free_temp(&s->dither.temp);
 232         swri_audio_convert_free(&s-> in_convert);
 233         swri_audio_convert_free(&s->out_convert);
 234         swri_audio_convert_free(&s->full_convert);
 235         if (s->resampler)
 236             s->resampler->free(&s->resample);
 237         swri_rematrix_free(s);
 238     }
 239
 240     av_freep(ss);
 241 }
 242
 243 av_cold int swr_init(struct SwrContext *s){
 244     int ret;
 245     s->in_buffer_index= 0;
 246     s->in_buffer_count= 0;
 247     s->resample_in_constraint= 0;
 248     free_temp(&s->postin);
 249     free_temp(&s->midbuf);
 250     free_temp(&s->preout);
 251     free_temp(&s->in_buffer);
 252     free_temp(&s->silence);
 253     free_temp(&s->drop_temp);
 254     free_temp(&s->dither.noise);
 255     free_temp(&s->dither.temp);
 256     memset(s->in.ch, 0, sizeof(s->in.ch));
 257     memset(s->out.ch, 0, sizeof(s->out.ch));
 258     swri_audio_convert_free(&s-> in_convert);
 259     swri_audio_convert_free(&s->out_convert);
 260     swri_audio_convert_free(&s->full_convert);
 261     swri_rematrix_free(s);
 262
 263     s->flushed = 0;
 264
 265     if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
 266         av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
 267         return AVERROR(EINVAL);
 268     }
 269     if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
 270         av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
 271         return AVERROR(EINVAL);
 272     }
 273
 274     if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) {
 275         av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
 276         s->in_ch_layout = 0;
 277     }
 278
 279     if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) {
 280         av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
 281         s->out_ch_layout = 0;
 282     }
 283
 284     switch(s->engine){
 285 #if CONFIG_LIBSOXR
 286         extern struct Resampler const soxr_resampler;
 287         case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break;
 288 #endif
 289         case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
 290         default:
 291             av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
 292             return AVERROR(EINVAL);
 293     }
 294
 295     if(!s->used_ch_count)
 296         s->used_ch_count= s->in.ch_count;
 297
 298     if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
 299         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
 300         s-> in_ch_layout= 0;
 301     }
 302
 303     if(!s-> in_ch_layout)
 304         s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
 305     if(!s->out_ch_layout)
 306         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
 307
 308     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
 309                  s->rematrix_custom;
 310
 311     if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
 312         if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
 313             s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
 314         }else if(   av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
 315                  && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P
 316                  && !s->rematrix
 317                  && s->engine != SWR_ENGINE_SOXR){
 318             s->int_sample_fmt= AV_SAMPLE_FMT_S32P;
 319         }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
 320             s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
 321         }else{
 322             av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
 323             s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
 324         }
 325     }
 326
 327     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P
 328         &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
 329         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
 330         &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
 331         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));
 332         return AVERROR(EINVAL);
 333     }
 334
 335     set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
 336     set_audiodata_fmt(&s->out, s->out_sample_fmt);
 337
 338     if (s->firstpts_in_samples != AV_NOPTS_VALUE) {
 339         if (!s->async && s->min_compensation >= FLT_MAX/2)
 340             s->async = 1;
 341         s->firstpts =
 342         s->outpts   = s->firstpts_in_samples * s->out_sample_rate;
 343     } else
 344         s->firstpts = AV_NOPTS_VALUE;
 345
 346     if (s->async) {
 347         if (s->min_compensation >= FLT_MAX/2)
 348             s->min_compensation = 0.001;
 349         if (s->async > 1.0001) {
 350             s->max_soft_compensation = s->async / (double) s->in_sample_rate;
 351         }
 352     }
 353
 354     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
 355         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);
 356     }else
 357         s->resampler->free(&s->resample);
 358     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P
 359         && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
 360         && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
 361         && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
 362         && s->resample){
 363         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
 364         return -1;
 365     }
 366
 367 #define RSC 1 //FIXME finetune
 368     if(!s-> in.ch_count)
 369         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
 370     if(!s->used_ch_count)
 371         s->used_ch_count= s->in.ch_count;
 372     if(!s->out.ch_count)
 373         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
 374
 375     if(!s-> in.ch_count){
 376         av_assert0(!s->in_ch_layout);
 377         av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
 378         return -1;
 379     }
 380
 381     if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
 382         char l1[1024], l2[1024];
 383         av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
 384         av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
 385         av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
 386                "but there is not enough information to do it\n", l1, l2);
 387         return -1;
 388     }
 389
 390 av_assert0(s->used_ch_count);
 391 av_assert0(s->out.ch_count);
 392     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
 393
 394     s->in_buffer= s->in;
 395     s->silence  = s->in;
 396     s->drop_temp= s->out;
 397
 398     if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
 399         s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
 400                                                    s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
 401         return 0;
 402     }
 403
 404     s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
 405                                              s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
 406     s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
 407                                              s->int_sample_fmt, s->out.ch_count, NULL, 0);
 408
 409     if (!s->in_convert || !s->out_convert)
 410         return AVERROR(ENOMEM);
 411
 412     s->postin= s->in;
 413     s->preout= s->out;
 414     s->midbuf= s->in;
 415
 416     if(s->channel_map){
 417         s->postin.ch_count=
 418         s->midbuf.ch_count= s->used_ch_count;
 419         if(s->resample)
 420             s->in_buffer.ch_count= s->used_ch_count;
 421     }
 422     if(!s->resample_first){
 423         s->midbuf.ch_count= s->out.ch_count;
 424         if(s->resample)
 425             s->in_buffer.ch_count = s->out.ch_count;
 426     }
 427
 428     set_audiodata_fmt(&s->postin, s->int_sample_fmt);
 429     set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
 430     set_audiodata_fmt(&s->preout, s->int_sample_fmt);
 431
 432     if(s->resample){
 433         set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
 434     }
 435
 436     if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
 437         return ret;
 438
 439     if(s->rematrix || s->dither.method)
 440         return swri_rematrix_init(s);
 441
 442     return 0;
 443 }
 444
 445 int swri_realloc_audio(AudioData *a, int count){
 446     int i, countb;
 447     AudioData old;
 448
 449     if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
 450         return AVERROR(EINVAL);
 451
 452     if(a->count >= count)
 453         return 0;
 454
 455     count*=2;
 456
 457     countb= FFALIGN(count*a->bps, ALIGN);
 458     old= *a;
 459
 460     av_assert0(a->bps);
 461     av_assert0(a->ch_count);
 462
 463     a->data= av_mallocz(countb*a->ch_count);
 464     if(!a->data)
 465         return AVERROR(ENOMEM);
 466     for(i=0; i<a->ch_count; i++){
 467         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
 468         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
 469     }
 470     if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
 471     av_freep(&old.data);
 472     a->count= count;
 473
 474     return 1;
 475 }
 476
 477 static void copy(AudioData *out, AudioData *in,
 478                  int count){
 479     av_assert0(out->planar == in->planar);
 480     av_assert0(out->bps == in->bps);
 481     av_assert0(out->ch_count == in->ch_count);
 482     if(out->planar){
 483         int ch;
 484         for(ch=0; ch<out->ch_count; ch++)
 485             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
 486     }else
 487         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
 488 }
 489
 490 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
 491     int i;
 492     if(!in_arg){
 493         memset(out->ch, 0, sizeof(out->ch));
 494     }else if(out->planar){
 495         for(i=0; i<out->ch_count; i++)
 496             out->ch[i]= in_arg[i];
 497     }else{
 498         for(i=0; i<out->ch_count; i++)
 499             out->ch[i]= in_arg[0] + i*out->bps;
 500     }
 501 }
 502
 503 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
 504     int i;
 505     if(out->planar){
 506         for(i=0; i<out->ch_count; i++)
 507             in_arg[i]= out->ch[i];
 508     }else{
 509         in_arg[0]= out->ch[0];
 510     }
 511 }
 512
 513 /**
 514  *
 515  * out may be equal in.
 516  */
 517 static void buf_set(AudioData *out, AudioData *in, int count){
 518     int ch;
 519     if(in->planar){
 520         for(ch=0; ch<out->ch_count; ch++)
 521             out->ch[ch]= in->ch[ch] + count*out->bps;
 522     }else{
 523         for(ch=out->ch_count-1; ch>=0; ch--)
 524             out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
 525     }
 526 }
 527
 528 /**
 529  *
 530  * @return number of samples output per channel
 531  */
 532 static int resample(SwrContext *s, AudioData *out_param, int out_count,
 533                              const AudioData * in_param, int in_count){
 534     AudioData in, out, tmp;
 535     int ret_sum=0;
 536     int border=0;
 537     int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
 538
 539     av_assert1(s->in_buffer.ch_count == in_param->ch_count);
 540     av_assert1(s->in_buffer.planar   == in_param->planar);
 541     av_assert1(s->in_buffer.fmt      == in_param->fmt);
 542
 543     tmp=out=*out_param;
 544     in =  *in_param;
 545
 546     do{
 547         int ret, size, consumed;
 548         if(!s->resample_in_constraint && s->in_buffer_count){
 549             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 550             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
 551             out_count -= ret;
 552             ret_sum += ret;
 553             buf_set(&out, &out, ret);
 554             s->in_buffer_count -= consumed;
 555             s->in_buffer_index += consumed;
 556
 557             if(!in_count)
 558                 break;
 559             if(s->in_buffer_count <= border){
 560                 buf_set(&in, &in, -s->in_buffer_count);
 561                 in_count += s->in_buffer_count;
 562                 s->in_buffer_count=0;
 563                 s->in_buffer_index=0;
 564                 border = 0;
 565             }
 566         }
 567
 568         if((s->flushed || in_count > padless) && !s->in_buffer_count){
 569             s->in_buffer_index=0;
 570             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
 571             out_count -= ret;
 572             ret_sum += ret;
 573             buf_set(&out, &out, ret);
 574             in_count -= consumed;
 575             buf_set(&in, &in, consumed);
 576         }
 577
 578         //TODO is this check sane considering the advanced copy avoidance below
 579         size= s->in_buffer_index + s->in_buffer_count + in_count;
 580         if(   size > s->in_buffer.count
 581            && s->in_buffer_count + in_count <= s->in_buffer_index){
 582             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 583             copy(&s->in_buffer, &tmp, s->in_buffer_count);
 584             s->in_buffer_index=0;
 585         }else
 586             if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
 587                 return ret;
 588
 589         if(in_count){
 590             int count= in_count;
 591             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
 592
 593             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
 594             copy(&tmp, &in, /*in_*/count);
 595             s->in_buffer_count += count;
 596             in_count -= count;
 597             border += count;
 598             buf_set(&in, &in, count);
 599             s->resample_in_constraint= 0;
 600             if(s->in_buffer_count != count || in_count)
 601                 continue;
 602             if (padless) {
 603                 padless = 0;
 604                 continue;
 605             }
 606         }
 607         break;
 608     }while(1);
 609
 610     s->resample_in_constraint= !!out_count;
 611
 612     return ret_sum;
 613 }
 614
 615 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
 616                                                       AudioData *in , int  in_count){
 617     AudioData *postin, *midbuf, *preout;
 618     int ret/*, in_max*/;
 619     AudioData preout_tmp, midbuf_tmp;
 620
 621     if(s->full_convert){
 622         av_assert0(!s->resample);
 623         swri_audio_convert(s->full_convert, out, in, in_count);
 624         return out_count;
 625     }
 626
 627 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
 628 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
 629
 630     if((ret=swri_realloc_audio(&s->postin, in_count))<0)
 631         return ret;
 632     if(s->resample_first){
 633         av_assert0(s->midbuf.ch_count == s->used_ch_count);
 634         if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
 635             return ret;
 636     }else{
 637         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
 638         if((ret=swri_realloc_audio(&s->midbuf,  in_count))<0)
 639             return ret;
 640     }
 641     if((ret=swri_realloc_audio(&s->preout, out_count))<0)
 642         return ret;
 643
 644     postin= &s->postin;
 645
 646     midbuf_tmp= s->midbuf;
 647     midbuf= &midbuf_tmp;
 648     preout_tmp= s->preout;
 649     preout= &preout_tmp;
 650
 651     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
 652         postin= in;
 653
 654     if(s->resample_first ? !s->resample : !s->rematrix)
 655         midbuf= postin;
 656
 657     if(s->resample_first ? !s->rematrix : !s->resample)
 658         preout= midbuf;
 659
 660     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
 661        && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){
 662         if(preout==in){
 663             out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
 664             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
 665             copy(out, in, out_count);
 666             return out_count;
 667         }
 668         else if(preout==postin) preout= midbuf= postin= out;
 669         else if(preout==midbuf) preout= midbuf= out;
 670         else                    preout= out;
 671     }
 672
 673     if(in != postin){
 674         swri_audio_convert(s->in_convert, postin, in, in_count);
 675     }
 676
 677     if(s->resample_first){
 678         if(postin != midbuf)
 679             out_count= resample(s, midbuf, out_count, postin, in_count);
 680         if(midbuf != preout)
 681             swri_rematrix(s, preout, midbuf, out_count, preout==out);
 682     }else{
 683         if(postin != midbuf)
 684             swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
 685         if(midbuf != preout)
 686             out_count= resample(s, preout, out_count, midbuf, in_count);
 687     }
 688
 689     if(preout != out && out_count){
 690         AudioData *conv_src = preout;
 691         if(s->dither.method){
 692             int ch;
 693             int dither_count= FFMAX(out_count, 1<<16);
 694
 695             if (preout == in) {
 696                 conv_src = &s->dither.temp;
 697                 if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
 698                     return ret;
 699             }
 700
 701             if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
 702                 return ret;
 703             if(ret)
 704                 for(ch=0; ch<s->dither.noise.ch_count; ch++)
 705                     swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);
 706             av_assert0(s->dither.noise.ch_count == preout->ch_count);
 707
 708             if(s->dither.noise_pos + out_count > s->dither.noise.count)
 709                 s->dither.noise_pos = 0;
 710
 711             if (s->dither.method < SWR_DITHER_NS){
 712                 if (s->mix_2_1_simd) {
 713                     int len1= out_count&~15;
 714                     int off = len1 * preout->bps;
 715
 716                     if(len1)
 717                         for(ch=0; ch<preout->ch_count; ch++)
 718                             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);
 719                     if(out_count != len1)
 720                         for(ch=0; ch<preout->ch_count; ch++)
 721                             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);
 722                 } else {
 723                     for(ch=0; ch<preout->ch_count; ch++)
 724                         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);
 725                 }
 726             } else {
 727                 switch(s->int_sample_fmt) {
 728                 case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
 729                 case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
 730                 case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
 731                 case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
 732                 }
 733             }
 734             s->dither.noise_pos += out_count;
 735         }
 736 //FIXME packed doesn't need more than 1 chan here!
 737         swri_audio_convert(s->out_convert, out, conv_src, out_count);
 738     }
 739     return out_count;
 740 }
 741
 742 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
 743                                 const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
 744     AudioData * in= &s->in;
 745     AudioData *out= &s->out;
 746
 747     while(s->drop_output > 0){
 748         int ret;
 749         uint8_t *tmp_arg[SWR_CH_MAX];
 750 #define MAX_DROP_STEP 16384
 751         if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)
 752             return ret;
 753
 754         reversefill_audiodata(&s->drop_temp, tmp_arg);
 755         s->drop_output *= -1; //FIXME find a less hackish solution
 756         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
 757         s->drop_output *= -1;
 758         in_count = 0;
 759         if(ret>0) {
 760             s->drop_output -= ret;
 761             continue;
 762         }
 763
 764         if(s->drop_output || !out_arg)
 765             return 0;
 766     }
 767
 768     if(!in_arg){
 769         if(s->resample){
 770             if (!s->flushed)
 771                 s->resampler->flush(s);
 772             s->resample_in_constraint = 0;
 773             s->flushed = 1;
 774         }else if(!s->in_buffer_count){
 775             return 0;
 776         }
 777     }else
 778         fill_audiodata(in ,  (void*)in_arg);
 779
 780     fill_audiodata(out, out_arg);
 781
 782     if(s->resample){
 783         int ret = swr_convert_internal(s, out, out_count, in, in_count);
 784         if(ret>0 && !s->drop_output)
 785             s->outpts += ret * (int64_t)s->in_sample_rate;
 786         return ret;
 787     }else{
 788         AudioData tmp= *in;
 789         int ret2=0;
 790         int ret, size;
 791         size = FFMIN(out_count, s->in_buffer_count);
 792         if(size){
 793             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 794             ret= swr_convert_internal(s, out, size, &tmp, size);
 795             if(ret<0)
 796                 return ret;
 797             ret2= ret;
 798             s->in_buffer_count -= ret;
 799             s->in_buffer_index += ret;
 800             buf_set(out, out, ret);
 801             out_count -= ret;
 802             if(!s->in_buffer_count)
 803                 s->in_buffer_index = 0;
 804         }
 805
 806         if(in_count){
 807             size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
 808
 809             if(in_count > out_count) { //FIXME move after swr_convert_internal
 810                 if(   size > s->in_buffer.count
 811                 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
 812                     buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
 813                     copy(&s->in_buffer, &tmp, s->in_buffer_count);
 814                     s->in_buffer_index=0;
 815                 }else
 816                     if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
 817                         return ret;
 818             }
 819
 820             if(out_count){
 821                 size = FFMIN(in_count, out_count);
 822                 ret= swr_convert_internal(s, out, size, in, size);
 823                 if(ret<0)
 824                     return ret;
 825                 buf_set(in, in, ret);
 826                 in_count -= ret;
 827                 ret2 += ret;
 828             }
 829             if(in_count){
 830                 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
 831                 copy(&tmp, in, in_count);
 832                 s->in_buffer_count += in_count;
 833             }
 834         }
 835         if(ret2>0 && !s->drop_output)
 836             s->outpts += ret2 * (int64_t)s->in_sample_rate;
 837         return ret2;
 838     }
 839 }
 840
 841 int swr_drop_output(struct SwrContext *s, int count){
 842     s->drop_output += count;
 843
 844     if(s->drop_output <= 0)
 845         return 0;
 846
 847     av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
 848     return swr_convert(s, NULL, s->drop_output, NULL, 0);
 849 }
 850
 851 int swr_inject_silence(struct SwrContext *s, int count){
 852     int ret, i;
 853     uint8_t *tmp_arg[SWR_CH_MAX];
 854
 855     if(count <= 0)
 856         return 0;
 857
 858 #define MAX_SILENCE_STEP 16384
 859     while (count > MAX_SILENCE_STEP) {
 860         if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
 861             return ret;
 862         count -= MAX_SILENCE_STEP;
 863     }
 864
 865     if((ret=swri_realloc_audio(&s->silence, count))<0)
 866         return ret;
 867
 868     if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
 869         memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
 870     } else
 871         memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
 872
 873     reversefill_audiodata(&s->silence, tmp_arg);
 874     av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
 875     ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
 876     return ret;
 877 }
 878
 879 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
 880     if (s->resampler && s->resample){
 881         return s->resampler->get_delay(s, base);
 882     }else{
 883         return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
 884     }
 885 }
 886
 887 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
 888     int ret;
 889
 890     if (!s || compensation_distance < 0)
 891         return AVERROR(EINVAL);
 892     if (!compensation_distance && sample_delta)
 893         return AVERROR(EINVAL);
 894     if (!s->resample) {
 895         s->flags |= SWR_FLAG_RESAMPLE;
 896         ret = swr_init(s);
 897         if (ret < 0)
 898             return ret;
 899     }
 900     if (!s->resampler->set_compensation){
 901         return AVERROR(EINVAL);
 902     }else{
 903         return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
 904     }
 905 }
 906
 907 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
 908     if(pts == INT64_MIN)
 909         return s->outpts;
 910
 911     if (s->firstpts == AV_NOPTS_VALUE)
 912         s->outpts = s->firstpts = pts;
 913
 914     if(s->min_compensation >= FLT_MAX) {
 915         return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
 916     } else {
 917         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;
 918         double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
 919
 920         if(fabs(fdelta) > s->min_compensation) {
 921             if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
 922                 int ret;
 923                 if(delta > 0) ret = swr_inject_silence(s,  delta / s->out_sample_rate);
 924                 else          ret = swr_drop_output   (s, -delta / s-> in_sample_rate);
 925                 if(ret<0){
 926                     av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
 927                 }
 928             } else if(s->soft_compensation_duration && s->max_soft_compensation) {
 929                 int duration = s->out_sample_rate * s->soft_compensation_duration;
 930                 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
 931                 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
 932                 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
 933                 swr_set_compensation(s, comp, duration);
 934             }
 935         }
 936
 937         return s->outpts;
 938     }
 939 }