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  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_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.  
  538.     av_assert1(s->in_buffer.ch_count == in_param->ch_count);
  539.     av_assert1(s->in_buffer.planar   == in_param->planar);
  540.     av_assert1(s->in_buffer.fmt      == in_param->fmt);
  541.  
  542.     tmp=out=*out_param;
  543.     in =  *in_param;
  544.  
  545.     do{
  546.         int ret, size, consumed;
  547.         if(!s->resample_in_constraint && s->in_buffer_count){
  548.             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
  549.             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
  550.             out_count -= ret;
  551.             ret_sum += ret;
  552.             buf_set(&out, &out, ret);
  553.             s->in_buffer_count -= consumed;
  554.             s->in_buffer_index += consumed;
  555.  
  556.             if(!in_count)
  557.                 break;
  558.             if(s->in_buffer_count <= border){
  559.                 buf_set(&in, &in, -s->in_buffer_count);
  560.                 in_count += s->in_buffer_count;
  561.                 s->in_buffer_count=0;
  562.                 s->in_buffer_index=0;
  563.                 border = 0;
  564.             }
  565.         }
  566.  
  567.         if((s->flushed || in_count) && !s->in_buffer_count){
  568.             s->in_buffer_index=0;
  569.             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
  570.             out_count -= ret;
  571.             ret_sum += ret;
  572.             buf_set(&out, &out, ret);
  573.             in_count -= consumed;
  574.             buf_set(&in, &in, consumed);
  575.         }
  576.  
  577.         //TODO is this check sane considering the advanced copy avoidance below
  578.         size= s->in_buffer_index + s->in_buffer_count + in_count;
  579.         if(   size > s->in_buffer.count
  580.            && s->in_buffer_count + in_count <= s->in_buffer_index){
  581.             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
  582.             copy(&s->in_buffer, &tmp, s->in_buffer_count);
  583.             s->in_buffer_index=0;
  584.         }else
  585.             if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
  586.                 return ret;
  587.  
  588.         if(in_count){
  589.             int count= in_count;
  590.             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
  591.  
  592.             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
  593.             copy(&tmp, &in, /*in_*/count);
  594.             s->in_buffer_count += count;
  595.             in_count -= count;
  596.             border += count;
  597.             buf_set(&in, &in, count);
  598.             s->resample_in_constraint= 0;
  599.             if(s->in_buffer_count != count || in_count)
  600.                 continue;
  601.         }
  602.         break;
  603.     }while(1);
  604.  
  605.     s->resample_in_constraint= !!out_count;
  606.  
  607.     return ret_sum;
  608. }
  609.  
  610. static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
  611.                                                       AudioData *in , int  in_count){
  612.     AudioData *postin, *midbuf, *preout;
  613.     int ret/*, in_max*/;
  614.     AudioData preout_tmp, midbuf_tmp;
  615.  
  616.     if(s->full_convert){
  617.         av_assert0(!s->resample);
  618.         swri_audio_convert(s->full_convert, out, in, in_count);
  619.         return out_count;
  620.     }
  621.  
  622. //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
  623. //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
  624.  
  625.     if((ret=swri_realloc_audio(&s->postin, in_count))<0)
  626.         return ret;
  627.     if(s->resample_first){
  628.         av_assert0(s->midbuf.ch_count == s->used_ch_count);
  629.         if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
  630.             return ret;
  631.     }else{
  632.         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
  633.         if((ret=swri_realloc_audio(&s->midbuf,  in_count))<0)
  634.             return ret;
  635.     }
  636.     if((ret=swri_realloc_audio(&s->preout, out_count))<0)
  637.         return ret;
  638.  
  639.     postin= &s->postin;
  640.  
  641.     midbuf_tmp= s->midbuf;
  642.     midbuf= &midbuf_tmp;
  643.     preout_tmp= s->preout;
  644.     preout= &preout_tmp;
  645.  
  646.     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
  647.         postin= in;
  648.  
  649.     if(s->resample_first ? !s->resample : !s->rematrix)
  650.         midbuf= postin;
  651.  
  652.     if(s->resample_first ? !s->rematrix : !s->resample)
  653.         preout= midbuf;
  654.  
  655.     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
  656.        && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){
  657.         if(preout==in){
  658.             out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
  659.             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
  660.             copy(out, in, out_count);
  661.             return out_count;
  662.         }
  663.         else if(preout==postin) preout= midbuf= postin= out;
  664.         else if(preout==midbuf) preout= midbuf= out;
  665.         else                    preout= out;
  666.     }
  667.  
  668.     if(in != postin){
  669.         swri_audio_convert(s->in_convert, postin, in, in_count);
  670.     }
  671.  
  672.     if(s->resample_first){
  673.         if(postin != midbuf)
  674.             out_count= resample(s, midbuf, out_count, postin, in_count);
  675.         if(midbuf != preout)
  676.             swri_rematrix(s, preout, midbuf, out_count, preout==out);
  677.     }else{
  678.         if(postin != midbuf)
  679.             swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
  680.         if(midbuf != preout)
  681.             out_count= resample(s, preout, out_count, midbuf, in_count);
  682.     }
  683.  
  684.     if(preout != out && out_count){
  685.         AudioData *conv_src = preout;
  686.         if(s->dither.method){
  687.             int ch;
  688.             int dither_count= FFMAX(out_count, 1<<16);
  689.  
  690.             if (preout == in) {
  691.                 conv_src = &s->dither.temp;
  692.                 if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
  693.                     return ret;
  694.             }
  695.  
  696.             if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
  697.                 return ret;
  698.             if(ret)
  699.                 for(ch=0; ch<s->dither.noise.ch_count; ch++)
  700.                     swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);
  701.             av_assert0(s->dither.noise.ch_count == preout->ch_count);
  702.  
  703.             if(s->dither.noise_pos + out_count > s->dither.noise.count)
  704.                 s->dither.noise_pos = 0;
  705.  
  706.             if (s->dither.method < SWR_DITHER_NS){
  707.                 if (s->mix_2_1_simd) {
  708.                     int len1= out_count&~15;
  709.                     int off = len1 * preout->bps;
  710.  
  711.                     if(len1)
  712.                         for(ch=0; ch<preout->ch_count; ch++)
  713.                             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);
  714.                     if(out_count != len1)
  715.                         for(ch=0; ch<preout->ch_count; ch++)
  716.                             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);
  717.                 } else {
  718.                     for(ch=0; ch<preout->ch_count; ch++)
  719.                         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);
  720.                 }
  721.             } else {
  722.                 switch(s->int_sample_fmt) {
  723.                 case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
  724.                 case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
  725.                 case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
  726.                 case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
  727.                 }
  728.             }
  729.             s->dither.noise_pos += out_count;
  730.         }
  731. //FIXME packed doesn't need more than 1 chan here!
  732.         swri_audio_convert(s->out_convert, out, conv_src, out_count);
  733.     }
  734.     return out_count;
  735. }
  736.  
  737. int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
  738.                                 const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
  739.     AudioData * in= &s->in;
  740.     AudioData *out= &s->out;
  741.  
  742.     while(s->drop_output > 0){
  743.         int ret;
  744.         uint8_t *tmp_arg[SWR_CH_MAX];
  745. #define MAX_DROP_STEP 16384
  746.         if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)
  747.             return ret;
  748.  
  749.         reversefill_audiodata(&s->drop_temp, tmp_arg);
  750.         s->drop_output *= -1; //FIXME find a less hackish solution
  751.         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
  752.         s->drop_output *= -1;
  753.         in_count = 0;
  754.         if(ret>0) {
  755.             s->drop_output -= ret;
  756.             continue;
  757.         }
  758.  
  759.         if(s->drop_output || !out_arg)
  760.             return 0;
  761.     }
  762.  
  763.     if(!in_arg){
  764.         if(s->resample){
  765.             if (!s->flushed)
  766.                 s->resampler->flush(s);
  767.             s->resample_in_constraint = 0;
  768.             s->flushed = 1;
  769.         }else if(!s->in_buffer_count){
  770.             return 0;
  771.         }
  772.     }else
  773.         fill_audiodata(in ,  (void*)in_arg);
  774.  
  775.     fill_audiodata(out, out_arg);
  776.  
  777.     if(s->resample){
  778.         int ret = swr_convert_internal(s, out, out_count, in, in_count);
  779.         if(ret>0 && !s->drop_output)
  780.             s->outpts += ret * (int64_t)s->in_sample_rate;
  781.         return ret;
  782.     }else{
  783.         AudioData tmp= *in;
  784.         int ret2=0;
  785.         int ret, size;
  786.         size = FFMIN(out_count, s->in_buffer_count);
  787.         if(size){
  788.             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
  789.             ret= swr_convert_internal(s, out, size, &tmp, size);
  790.             if(ret<0)
  791.                 return ret;
  792.             ret2= ret;
  793.             s->in_buffer_count -= ret;
  794.             s->in_buffer_index += ret;
  795.             buf_set(out, out, ret);
  796.             out_count -= ret;
  797.             if(!s->in_buffer_count)
  798.                 s->in_buffer_index = 0;
  799.         }
  800.  
  801.         if(in_count){
  802.             size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
  803.  
  804.             if(in_count > out_count) { //FIXME move after swr_convert_internal
  805.                 if(   size > s->in_buffer.count
  806.                 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
  807.                     buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
  808.                     copy(&s->in_buffer, &tmp, s->in_buffer_count);
  809.                     s->in_buffer_index=0;
  810.                 }else
  811.                     if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
  812.                         return ret;
  813.             }
  814.  
  815.             if(out_count){
  816.                 size = FFMIN(in_count, out_count);
  817.                 ret= swr_convert_internal(s, out, size, in, size);
  818.                 if(ret<0)
  819.                     return ret;
  820.                 buf_set(in, in, ret);
  821.                 in_count -= ret;
  822.                 ret2 += ret;
  823.             }
  824.             if(in_count){
  825.                 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
  826.                 copy(&tmp, in, in_count);
  827.                 s->in_buffer_count += in_count;
  828.             }
  829.         }
  830.         if(ret2>0 && !s->drop_output)
  831.             s->outpts += ret2 * (int64_t)s->in_sample_rate;
  832.         return ret2;
  833.     }
  834. }
  835.  
  836. int swr_drop_output(struct SwrContext *s, int count){
  837.     s->drop_output += count;
  838.  
  839.     if(s->drop_output <= 0)
  840.         return 0;
  841.  
  842.     av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
  843.     return swr_convert(s, NULL, s->drop_output, NULL, 0);
  844. }
  845.  
  846. int swr_inject_silence(struct SwrContext *s, int count){
  847.     int ret, i;
  848.     uint8_t *tmp_arg[SWR_CH_MAX];
  849.  
  850.     if(count <= 0)
  851.         return 0;
  852.  
  853. #define MAX_SILENCE_STEP 16384
  854.     while (count > MAX_SILENCE_STEP) {
  855.         if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
  856.             return ret;
  857.         count -= MAX_SILENCE_STEP;
  858.     }
  859.  
  860.     if((ret=swri_realloc_audio(&s->silence, count))<0)
  861.         return ret;
  862.  
  863.     if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
  864.         memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
  865.     } else
  866.         memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
  867.  
  868.     reversefill_audiodata(&s->silence, tmp_arg);
  869.     av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
  870.     ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
  871.     return ret;
  872. }
  873.  
  874. int64_t swr_get_delay(struct SwrContext *s, int64_t base){
  875.     if (s->resampler && s->resample){
  876.         return s->resampler->get_delay(s, base);
  877.     }else{
  878.         return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
  879.     }
  880. }
  881.  
  882. int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
  883.     int ret;
  884.  
  885.     if (!s || compensation_distance < 0)
  886.         return AVERROR(EINVAL);
  887.     if (!compensation_distance && sample_delta)
  888.         return AVERROR(EINVAL);
  889.     if (!s->resample) {
  890.         s->flags |= SWR_FLAG_RESAMPLE;
  891.         ret = swr_init(s);
  892.         if (ret < 0)
  893.             return ret;
  894.     }
  895.     if (!s->resampler->set_compensation){
  896.         return AVERROR(EINVAL);
  897.     }else{
  898.         return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
  899.     }
  900. }
  901.  
  902. int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
  903.     if(pts == INT64_MIN)
  904.         return s->outpts;
  905.  
  906.     if (s->firstpts == AV_NOPTS_VALUE)
  907.         s->outpts = s->firstpts = pts;
  908.  
  909.     if(s->min_compensation >= FLT_MAX) {
  910.         return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
  911.     } else {
  912.         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;
  913.         double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
  914.  
  915.         if(fabs(fdelta) > s->min_compensation) {
  916.             if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
  917.                 int ret;
  918.                 if(delta > 0) ret = swr_inject_silence(s,  delta / s->out_sample_rate);
  919.                 else          ret = swr_drop_output   (s, -delta / s-> in_sample_rate);
  920.                 if(ret<0){
  921.                     av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
  922.                 }
  923.             } else if(s->soft_compensation_duration && s->max_soft_compensation) {
  924.                 int duration = s->out_sample_rate * s->soft_compensation_duration;
  925.                 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
  926.                 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
  927.                 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
  928.                 swr_set_compensation(s, comp, duration);
  929.             }
  930.         }
  931.  
  932.         return s->outpts;
  933.     }
  934. }
  935.