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FFmpeg/libswresample/swresample.c

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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_INT64, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},

  61. {"in_channel_layout"    , "set input channel layout"    , OFFSET( in_ch_layout  ), AV_OPT_TYPE_INT64, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},

  62. {"ocl"                  , "set output channel layout"   , OFFSET(out_ch_layout  ), AV_OPT_TYPE_INT64, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},

  63. {"out_channel_layout"   , "set output channel layout"   , OFFSET(out_ch_layout  ), AV_OPT_TYPE_INT64, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},

  64. {"clev"                 , "set center mix level"        , OFFSET(clev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},

  65. {"center_mix_level"     , "set center mix level"        , OFFSET(clev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},

  66. {"slev"                 , "set surround mix level"      , OFFSET(slev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},

  67. {"surround_mix_level"   , "set surround mix Level"      , OFFSET(slev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},

  68. {"lfe_mix_level"        , "set LFE mix level"           , OFFSET(lfe_mix_level  ), AV_OPT_TYPE_FLOAT, {.dbl=0                     }, -32    , 32        , PARAM},

  69. {"rmvol"                , "set rematrix volume"         , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0                   }, -1000  , 1000      , PARAM},

  70. {"rematrix_volume"      , "set rematrix volume"         , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0                   }, -1000  , 1000      , PARAM},

  71. 

  72. {"flags"                , "set flags"                   , OFFSET(flags          ), AV_OPT_TYPE_FLAGS, {.i64=0                     }, 0      , UINT_MAX  , PARAM, "flags"},

  73. {"swr_flags"            , "set flags"                   , OFFSET(flags          ), AV_OPT_TYPE_FLAGS, {.i64=0                     }, 0      , UINT_MAX  , PARAM, "flags"},

  74. {"res"                  , "force resampling"            , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_FLAG_RESAMPLE     }, INT_MIN, INT_MAX   , PARAM, "flags"},

  75. 

  76. {"dither_scale"         , "set dither scale"            , OFFSET(dither.scale   ), AV_OPT_TYPE_FLOAT, {.dbl=1                     }, 0      , INT_MAX   , PARAM},

  77. 

  78. {"dither_method"        , "set dither method"           , OFFSET(dither.method  ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_DITHER_NB-1, PARAM, "dither_method"},

  79. {"rectangular"          , "select rectangular dither"   , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_RECTANGULAR}, INT_MIN, INT_MAX   , PARAM, "dither_method"},

  80. {"triangular"           , "select triangular dither"    , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX   , PARAM, "dither_method"},

  81. {"triangular_hp"        , "select triangular dither with high pass" , 0          , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR_HIGHPASS }, INT_MIN, INT_MAX, PARAM, "dither_method"},

  82. {"lipshitz"             , "select lipshitz noise shaping dither" , 0             , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_LIPSHITZ}, INT_MIN, INT_MAX, PARAM, "dither_method"},

  83. {"shibata"              , "select shibata noise shaping dither" , 0              , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"},

  84. {"low_shibata"          , "select low shibata noise shaping dither" , 0          , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_LOW_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"},

  85. {"high_shibata"         , "select high shibata noise shaping dither" , 0         , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_HIGH_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"},

  86. {"f_weighted"           , "select f-weighted noise shaping dither" , 0           , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_F_WEIGHTED }, INT_MIN, INT_MAX, PARAM, "dither_method"},

  87. {"modified_e_weighted"  , "select modified-e-weighted noise shaping dither" , 0  , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_MODIFIED_E_WEIGHTED }, INT_MIN, INT_MAX, PARAM, "dither_method"},

  88. {"improved_e_weighted"  , "select improved-e-weighted noise shaping dither" , 0  , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_IMPROVED_E_WEIGHTED }, INT_MIN, INT_MAX, PARAM, "dither_method"},

  89. 

  90. {"filter_size"          , "set swr resampling filter size", OFFSET(filter_size)  , AV_OPT_TYPE_INT  , {.i64=32                    }, 0      , INT_MAX   , PARAM },

  91. {"phase_shift"          , "set swr resampling phase shift", OFFSET(phase_shift)  , AV_OPT_TYPE_INT  , {.i64=10                    }, 0      , 30        , PARAM },

  92. {"linear_interp"        , "enable linear interpolation" , OFFSET(linear_interp)  , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , 1         , PARAM },

  93. {"cutoff"               , "set cutoff frequency ratio"  , OFFSET(cutoff)         , AV_OPT_TYPE_DOUBLE,{.dbl=0.                    }, 0      , 1         , PARAM },

  94. {"resampler"            , "set resampling Engine"       , OFFSET(engine)         , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_ENGINE_NB-1, PARAM, "resampler"},

  95. {"swr"                  , "select SW Resampler"         , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SWR        }, INT_MIN, INT_MAX   , PARAM, "resampler"},

  96. {"soxr"                 , "select SoX Resampler"        , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SOXR       }, INT_MIN, INT_MAX   , PARAM, "resampler"},

  97. {"precision"            , "set soxr resampling precision (in bits)"

  98.                                                         , OFFSET(precision)      , AV_OPT_TYPE_DOUBLE,{.dbl=20.0                  }, 15.0   , 33.0      , PARAM },

  99. {"cheby"                , "enable soxr Chebyshev passband & higher-precision irrational ratio approximation"

  100.                                                         , OFFSET(cheby)          , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , 1         , PARAM },

  101. {"min_comp"             , "set minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"

  102.                                                         , OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX               }, 0      , FLT_MAX   , PARAM },

  103. {"min_hard_comp"        , "set minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."

  104.                                                         , OFFSET(min_hard_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0.1                   }, 0      , INT_MAX   , PARAM },

  105. {"comp_duration"        , "set duration (in seconds) over which data is stretched/squeezed to make it match the timestamps."

  106.                                                         , OFFSET(soft_compensation_duration),AV_OPT_TYPE_FLOAT ,{.dbl=1                     }, 0      , INT_MAX   , PARAM },

  107. {"max_soft_comp"        , "set maximum factor by which data is stretched/squeezed to make it match the timestamps."

  108.                                                         , OFFSET(max_soft_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0                     }, INT_MIN, INT_MAX   , PARAM },

  109. {"async"                , "simplified 1 parameter audio timestamp matching, 0(disabled), 1(filling and trimming), >1(maximum stretch/squeeze in samples per second)"

  110.                                                         , OFFSET(async)          , AV_OPT_TYPE_FLOAT ,{.dbl=0                     }, INT_MIN, INT_MAX   , PARAM },

  111. {"first_pts"            , "Assume the first pts should be this value (in samples)."

  112.                                                         , OFFSET(firstpts_in_samples), AV_OPT_TYPE_INT64 ,{.i64=AV_NOPTS_VALUE    }, INT64_MIN,INT64_MAX, PARAM },

  113. 

  114. { "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" },

  115.     { "none",  "select none",               0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_NONE  }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },

  116.     { "dolby", "select Dolby",              0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },

  117.     { "dplii", "select Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },

  118. 

  119. { "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" },

  120.     { "cubic"           , "select cubic"                , 0                      , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC            }, INT_MIN, INT_MAX, PARAM, "filter_type" },

  121.     { "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" },

  122.     { "kaiser"          , "select Kaiser Windowed Sinc" , 0                      , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER           }, INT_MIN, INT_MAX, PARAM, "filter_type" },

  123. 

  124. { "kaiser_beta"         , "set swr Kaiser Window Beta"  , OFFSET(kaiser_beta)    , AV_OPT_TYPE_INT  , {.i64=9                     }, 2      , 16        , PARAM },

  125. 

  126. {0}

  127. };

  128. 

  129. static const char* context_to_name(void* ptr) {

  130.     return "SWR";

  131. }

  132. 

  133. static const AVClass av_class = {

  134.     .class_name                = "SWResampler",

  135.     .item_name                 = context_to_name,

  136.     .option                    = options,

  137.     .version                   = LIBAVUTIL_VERSION_INT,

  138.     .log_level_offset_offset   = OFFSET(log_level_offset),

  139.     .parent_log_context_offset = OFFSET(log_ctx),

  140.     .category                  = AV_CLASS_CATEGORY_SWRESAMPLER,

  141. };

  142. 

  143. unsigned swresample_version(void)

  144. {

  145.     av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);

  146.     return LIBSWRESAMPLE_VERSION_INT;

  147. }

  148. 

  149. const char *swresample_configuration(void)

  150. {

  151.     return FFMPEG_CONFIGURATION;

  152. }

  153. 

  154. const char *swresample_license(void)

  155. {

  156. #define LICENSE_PREFIX "libswresample license: "

  157.     return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;

  158. }

  159. 

  160. int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){

  161.     if(!s || s->in_convert) // s needs to be allocated but not initialized

  162.         return AVERROR(EINVAL);

  163.     s->channel_map = channel_map;

  164.     return 0;

  165. }

  166. 

  167. const AVClass *swr_get_class(void)

  168. {

  169.     return &av_class;

  170. }

  171. 

  172. av_cold struct SwrContext *swr_alloc(void){

  173.     SwrContext *s= av_mallocz(sizeof(SwrContext));

  174.     if(s){

  175.         s->av_class= &av_class;

  176.         av_opt_set_defaults(s);

  177.     }

  178.     return s;

  179. }

  180. 

  181. struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,

  182.                                       int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,

  183.                                       int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,

  184.                                       int log_offset, void *log_ctx){

  185.     if(!s) s= swr_alloc();

  186.     if(!s) return NULL;

  187. 

  188.     s->log_level_offset= log_offset;

  189.     s->log_ctx= log_ctx;

  190. 

  191.     av_opt_set_int(s, "ocl", out_ch_layout,   0);

  192.     av_opt_set_int(s, "osf", out_sample_fmt,  0);

  193.     av_opt_set_int(s, "osr", out_sample_rate, 0);

  194.     av_opt_set_int(s, "icl", in_ch_layout,    0);

  195.     av_opt_set_int(s, "isf", in_sample_fmt,   0);

  196.     av_opt_set_int(s, "isr", in_sample_rate,  0);

  197.     av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE,   0);

  198.     av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);

  199.     av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);

  200.     av_opt_set_int(s, "uch", 0, 0);

  201.     return s;

  202. }

  203. 

  204. static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){

  205.     a->fmt   = fmt;

  206.     a->bps   = av_get_bytes_per_sample(fmt);

  207.     a->planar= av_sample_fmt_is_planar(fmt);

  208. }

  209. 

  210. static void free_temp(AudioData *a){

  211.     av_free(a->data);

  212.     memset(a, 0, sizeof(*a));

  213. }

  214. 

  215. av_cold void swr_free(SwrContext **ss){

  216.     SwrContext *s= *ss;

  217.     if(s){

  218.         free_temp(&s->postin);

  219.         free_temp(&s->midbuf);

  220.         free_temp(&s->preout);

  221.         free_temp(&s->in_buffer);

  222.         free_temp(&s->silence);

  223.         free_temp(&s->drop_temp);

  224.         free_temp(&s->dither.noise);

  225.         free_temp(&s->dither.temp);

  226.         swri_audio_convert_free(&s-> in_convert);

  227.         swri_audio_convert_free(&s->out_convert);

  228.         swri_audio_convert_free(&s->full_convert);

  229.         if (s->resampler)

  230.             s->resampler->free(&s->resample);

  231.         swri_rematrix_free(s);

  232.     }

  233. 

  234.     av_freep(ss);

  235. }

  236. 

  237. av_cold int swr_init(struct SwrContext *s){

  238.     int ret;

  239.     s->in_buffer_index= 0;

  240.     s->in_buffer_count= 0;

  241.     s->resample_in_constraint= 0;

  242.     free_temp(&s->postin);

  243.     free_temp(&s->midbuf);

  244.     free_temp(&s->preout);

  245.     free_temp(&s->in_buffer);

  246.     free_temp(&s->silence);

  247.     free_temp(&s->drop_temp);

  248.     free_temp(&s->dither.noise);

  249.     free_temp(&s->dither.temp);

  250.     memset(s->in.ch, 0, sizeof(s->in.ch));

  251.     memset(s->out.ch, 0, sizeof(s->out.ch));

  252.     swri_audio_convert_free(&s-> in_convert);

  253.     swri_audio_convert_free(&s->out_convert);

  254.     swri_audio_convert_free(&s->full_convert);

  255.     swri_rematrix_free(s);

  256. 

  257.     s->flushed = 0;

  258. 

  259.     if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){

  260.         av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);

  261.         return AVERROR(EINVAL);

  262.     }

  263.     if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){

  264.         av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);

  265.         return AVERROR(EINVAL);

  266.     }

  267. 

  268.     if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){

  269.         if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){

  270.             s->int_sample_fmt= AV_SAMPLE_FMT_S16P;

  271.         }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){

  272.             s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;

  273.         }else{

  274.             av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");

  275.             s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;

  276.         }

  277.     }

  278. 

  279.     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P

  280.         &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P

  281.         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP

  282.         &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){

  283.         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));

  284.         return AVERROR(EINVAL);

  285.     }

  286. 

  287.     switch(s->engine){

  288. #if CONFIG_LIBSOXR

  289.         extern struct Resampler const soxr_resampler;

  290.         case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break;

  291. #endif

  292.         case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;

  293.         default:

  294.             av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");

  295.             return AVERROR(EINVAL);

  296.     }

  297. 

  298.     set_audiodata_fmt(&s-> in, s-> in_sample_fmt);

  299.     set_audiodata_fmt(&s->out, s->out_sample_fmt);

  300. 

  301.     if (s->firstpts_in_samples != AV_NOPTS_VALUE) {

  302.         if (!s->async && s->min_compensation >= FLT_MAX/2)

  303.             s->async = 1;

  304.         s->firstpts =

  305.         s->outpts   = s->firstpts_in_samples * s->out_sample_rate;

  306.     }

  307. 

  308.     if (s->async) {

  309.         if (s->min_compensation >= FLT_MAX/2)

  310.             s->min_compensation = 0.001;

  311.         if (s->async > 1.0001) {

  312.             s->max_soft_compensation = s->async / (double) s->in_sample_rate;

  313.         }

  314.     }

  315. 

  316.     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){

  317.         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);

  318.     }else

  319.         s->resampler->free(&s->resample);

  320.     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P

  321.         && s->int_sample_fmt != AV_SAMPLE_FMT_S32P

  322.         && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP

  323.         && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP

  324.         && s->resample){

  325.         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");

  326.         return -1;

  327.     }

  328. 

  329.     if(!s->used_ch_count)

  330.         s->used_ch_count= s->in.ch_count;

  331. 

  332.     if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){

  333.         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");

  334.         s-> in_ch_layout= 0;

  335.     }

  336. 

  337.     if(!s-> in_ch_layout)

  338.         s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);

  339.     if(!s->out_ch_layout)

  340.         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);

  341. 

  342.     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0 ||

  343.                  s->rematrix_custom;

  344. 

  345. #define RSC 1 //FIXME finetune

  346.     if(!s-> in.ch_count)

  347.         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);

  348.     if(!s->used_ch_count)

  349.         s->used_ch_count= s->in.ch_count;

  350.     if(!s->out.ch_count)

  351.         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);

  352. 

  353.     if(!s-> in.ch_count){

  354.         av_assert0(!s->in_ch_layout);

  355.         av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");

  356.         return -1;

  357.     }

  358. 

  359.     if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {

  360.         char l1[1024], l2[1024];

  361.         av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);

  362.         av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);

  363.         av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "

  364.                "but there is not enough information to do it\n", l1, l2);

  365.         return -1;

  366.     }

  367. 

  368. av_assert0(s->used_ch_count);

  369. av_assert0(s->out.ch_count);

  370.     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;

  371. 

  372.     s->in_buffer= s->in;

  373.     s->silence  = s->in;

  374.     s->drop_temp= s->out;

  375. 

  376.     if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){

  377.         s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,

  378.                                                    s-> in_sample_fmt, s-> in.ch_count, NULL, 0);

  379.         return 0;

  380.     }

  381. 

  382.     s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,

  383.                                              s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);

  384.     s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,

  385.                                              s->int_sample_fmt, s->out.ch_count, NULL, 0);

  386. 

  387.     if (!s->in_convert || !s->out_convert)

  388.         return AVERROR(ENOMEM);

  389. 

  390.     s->postin= s->in;

  391.     s->preout= s->out;

  392.     s->midbuf= s->in;

  393. 

  394.     if(s->channel_map){

  395.         s->postin.ch_count=

  396.         s->midbuf.ch_count= s->used_ch_count;

  397.         if(s->resample)

  398.             s->in_buffer.ch_count= s->used_ch_count;

  399.     }

  400.     if(!s->resample_first){

  401.         s->midbuf.ch_count= s->out.ch_count;

  402.         if(s->resample)

  403.             s->in_buffer.ch_count = s->out.ch_count;

  404.     }

  405. 

  406.     set_audiodata_fmt(&s->postin, s->int_sample_fmt);

  407.     set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);

  408.     set_audiodata_fmt(&s->preout, s->int_sample_fmt);

  409. 

  410.     if(s->resample){

  411.         set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);

  412.     }

  413. 

  414.     if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)

  415.         return ret;

  416. 

  417.     if(s->rematrix || s->dither.method)

  418.         return swri_rematrix_init(s);

  419. 

  420.     return 0;

  421. }

  422. 

  423. int swri_realloc_audio(AudioData *a, int count){

  424.     int i, countb;

  425.     AudioData old;

  426. 

  427.     if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)

  428.         return AVERROR(EINVAL);

  429. 

  430.     if(a->count >= count)

  431.         return 0;

  432. 

  433.     count*=2;

  434. 

  435.     countb= FFALIGN(count*a->bps, ALIGN);

  436.     old= *a;

  437. 

  438.     av_assert0(a->bps);

  439.     av_assert0(a->ch_count);

  440. 

  441.     a->data= av_mallocz(countb*a->ch_count);

  442.     if(!a->data)

  443.         return AVERROR(ENOMEM);

  444.     for(i=0; i<a->ch_count; i++){

  445.         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);

  446.         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);

  447.     }

  448.     if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);

  449.     av_free(old.data);

  450.     a->count= count;

  451. 

  452.     return 1;

  453. }

  454. 

  455. static void copy(AudioData *out, AudioData *in,

  456.                  int count){

  457.     av_assert0(out->planar == in->planar);

  458.     av_assert0(out->bps == in->bps);

  459.     av_assert0(out->ch_count == in->ch_count);

  460.     if(out->planar){

  461.         int ch;

  462.         for(ch=0; ch<out->ch_count; ch++)

  463.             memcpy(out->ch[ch], in->ch[ch], count*out->bps);

  464.     }else

  465.         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);

  466. }

  467. 

  468. static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){

  469.     int i;

  470.     if(!in_arg){

  471.         memset(out->ch, 0, sizeof(out->ch));

  472.     }else if(out->planar){

  473.         for(i=0; i<out->ch_count; i++)

  474.             out->ch[i]= in_arg[i];

  475.     }else{

  476.         for(i=0; i<out->ch_count; i++)

  477.             out->ch[i]= in_arg[0] + i*out->bps;

  478.     }

  479. }

  480. 

  481. static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){

  482.     int i;

  483.     if(out->planar){

  484.         for(i=0; i<out->ch_count; i++)

  485.             in_arg[i]= out->ch[i];

  486.     }else{

  487.         in_arg[0]= out->ch[0];

  488.     }

  489. }

  490. 

  491. /**

  492.  *

  493.  * out may be equal in.

  494.  */

  495. static void buf_set(AudioData *out, AudioData *in, int count){

  496.     int ch;

  497.     if(in->planar){

  498.         for(ch=0; ch<out->ch_count; ch++)

  499.             out->ch[ch]= in->ch[ch] + count*out->bps;

  500.     }else{

  501.         for(ch=out->ch_count-1; ch>=0; ch--)

  502.             out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;

  503.     }

  504. }

  505. 

  506. /**

  507.  *

  508.  * @return number of samples output per channel

  509.  */

  510. static int resample(SwrContext *s, AudioData *out_param, int out_count,

  511.                              const AudioData * in_param, int in_count){

  512.     AudioData in, out, tmp;

  513.     int ret_sum=0;

  514.     int border=0;

  515. 

  516.     av_assert1(s->in_buffer.ch_count == in_param->ch_count);

  517.     av_assert1(s->in_buffer.planar   == in_param->planar);

  518.     av_assert1(s->in_buffer.fmt      == in_param->fmt);

  519. 

  520.     tmp=out=*out_param;

  521.     in =  *in_param;

  522. 

  523.     do{

  524.         int ret, size, consumed;

  525.         if(!s->resample_in_constraint && s->in_buffer_count){

  526.             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);

  527.             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);

  528.             out_count -= ret;

  529.             ret_sum += ret;

  530.             buf_set(&out, &out, ret);

  531.             s->in_buffer_count -= consumed;

  532.             s->in_buffer_index += consumed;

  533. 

  534.             if(!in_count)

  535.                 break;

  536.             if(s->in_buffer_count <= border){

  537.                 buf_set(&in, &in, -s->in_buffer_count);

  538.                 in_count += s->in_buffer_count;

  539.                 s->in_buffer_count=0;

  540.                 s->in_buffer_index=0;

  541.                 border = 0;

  542.             }

  543.         }

  544. 

  545.         if((s->flushed || in_count) && !s->in_buffer_count){

  546.             s->in_buffer_index=0;

  547.             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);

  548.             out_count -= ret;

  549.             ret_sum += ret;

  550.             buf_set(&out, &out, ret);

  551.             in_count -= consumed;

  552.             buf_set(&in, &in, consumed);

  553.         }

  554. 

  555.         //TODO is this check sane considering the advanced copy avoidance below

  556.         size= s->in_buffer_index + s->in_buffer_count + in_count;

  557.         if(   size > s->in_buffer.count

  558.            && s->in_buffer_count + in_count <= s->in_buffer_index){

  559.             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);

  560.             copy(&s->in_buffer, &tmp, s->in_buffer_count);

  561.             s->in_buffer_index=0;

  562.         }else

  563.             if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)

  564.                 return ret;

  565. 

  566.         if(in_count){

  567.             int count= in_count;

  568.             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;

  569. 

  570.             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);

  571.             copy(&tmp, &in, /*in_*/count);

  572.             s->in_buffer_count += count;

  573.             in_count -= count;

  574.             border += count;

  575.             buf_set(&in, &in, count);

  576.             s->resample_in_constraint= 0;

  577.             if(s->in_buffer_count != count || in_count)

  578.                 continue;

  579.         }

  580.         break;

  581.     }while(1);

  582. 

  583.     s->resample_in_constraint= !!out_count;

  584. 

  585.     return ret_sum;

  586. }

  587. 

  588. static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,

  589.                                                       AudioData *in , int  in_count){

  590.     AudioData *postin, *midbuf, *preout;

  591.     int ret/*, in_max*/;

  592.     AudioData preout_tmp, midbuf_tmp;

  593. 

  594.     if(s->full_convert){

  595.         av_assert0(!s->resample);

  596.         swri_audio_convert(s->full_convert, out, in, in_count);

  597.         return out_count;

  598.     }

  599. 

  600. //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;

  601. //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);

  602. 

  603.     if((ret=swri_realloc_audio(&s->postin, in_count))<0)

  604.         return ret;

  605.     if(s->resample_first){

  606.         av_assert0(s->midbuf.ch_count == s->used_ch_count);

  607.         if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)

  608.             return ret;

  609.     }else{

  610.         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);

  611.         if((ret=swri_realloc_audio(&s->midbuf,  in_count))<0)

  612.             return ret;

  613.     }

  614.     if((ret=swri_realloc_audio(&s->preout, out_count))<0)

  615.         return ret;

  616. 

  617.     postin= &s->postin;

  618. 

  619.     midbuf_tmp= s->midbuf;

  620.     midbuf= &midbuf_tmp;

  621.     preout_tmp= s->preout;

  622.     preout= &preout_tmp;

  623. 

  624.     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)

  625.         postin= in;

  626. 

  627.     if(s->resample_first ? !s->resample : !s->rematrix)

  628.         midbuf= postin;

  629. 

  630.     if(s->resample_first ? !s->rematrix : !s->resample)

  631.         preout= midbuf;

  632. 

  633.     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){

  634.         if(preout==in){

  635.             out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant

  636.             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though

  637.             copy(out, in, out_count);

  638.             return out_count;

  639.         }

  640.         else if(preout==postin) preout= midbuf= postin= out;

  641.         else if(preout==midbuf) preout= midbuf= out;

  642.         else                    preout= out;

  643.     }

  644. 

  645.     if(in != postin){

  646.         swri_audio_convert(s->in_convert, postin, in, in_count);

  647.     }

  648. 

  649.     if(s->resample_first){

  650.         if(postin != midbuf)

  651.             out_count= resample(s, midbuf, out_count, postin, in_count);

  652.         if(midbuf != preout)

  653.             swri_rematrix(s, preout, midbuf, out_count, preout==out);

  654.     }else{

  655.         if(postin != midbuf)

  656.             swri_rematrix(s, midbuf, postin, in_count, midbuf==out);

  657.         if(midbuf != preout)

  658.             out_count= resample(s, preout, out_count, midbuf, in_count);

  659.     }

  660. 

  661.     if(preout != out && out_count){

  662.         AudioData *conv_src = preout;

  663.         if(s->dither.method){

  664.             int ch;

  665.             int dither_count= FFMAX(out_count, 1<<16);

  666. 

  667.             if (preout == in) {

  668.                 conv_src = &s->dither.temp;

  669.                 if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)

  670.                     return ret;

  671.             }

  672. 

  673.             if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)

  674.                 return ret;

  675.             if(ret)

  676.                 for(ch=0; ch<s->dither.noise.ch_count; ch++)

  677.                     swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);

  678.             av_assert0(s->dither.noise.ch_count == preout->ch_count);

  679. 

  680.             if(s->dither.noise_pos + out_count > s->dither.noise.count)

  681.                 s->dither.noise_pos = 0;

  682. 

  683.             if (s->dither.method < SWR_DITHER_NS){

  684.                 if (s->mix_2_1_simd) {

  685.                     int len1= out_count&~15;

  686.                     int off = len1 * preout->bps;

  687. 

  688.                     if(len1)

  689.                         for(ch=0; ch<preout->ch_count; ch++)

  690.                             s->mix_2_1_simd(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_one, 0, 0, len1);

  691.                     if(out_count != len1)

  692.                         for(ch=0; ch<preout->ch_count; ch++)

  693.                             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);

  694.                 } else {

  695.                     for(ch=0; ch<preout->ch_count; ch++)

  696.                         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);

  697.                 }

  698.             } else {

  699.                 switch(s->int_sample_fmt) {

  700.                 case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;

  701.                 case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;

  702.                 case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;

  703.                 case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;

  704.                 }

  705.             }

  706.             s->dither.noise_pos += out_count;

  707.         }

  708. //FIXME packed doesnt need more than 1 chan here!

  709.         swri_audio_convert(s->out_convert, out, conv_src, out_count);

  710.     }

  711.     return out_count;

  712. }

  713. 

  714. int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,

  715.                                 const uint8_t *in_arg [SWR_CH_MAX], int  in_count){

  716.     AudioData * in= &s->in;

  717.     AudioData *out= &s->out;

  718. 

  719.     while(s->drop_output > 0){

  720.         int ret;

  721.         uint8_t *tmp_arg[SWR_CH_MAX];

  722. #define MAX_DROP_STEP 16384

  723.         if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)

  724.             return ret;

  725. 

  726.         reversefill_audiodata(&s->drop_temp, tmp_arg);

  727.         s->drop_output *= -1; //FIXME find a less hackish solution

  728.         ret = swr_convert(s, tmp_arg, FFMIN(-s->drop_output, MAX_DROP_STEP), in_arg, in_count); //FIXME optimize but this is as good as never called so maybe it doesnt matter

  729.         s->drop_output *= -1;

  730.         in_count = 0;

  731.         if(ret>0) {

  732.             s->drop_output -= ret;

  733.             continue;

  734.         }

  735. 

  736.         if(s->drop_output || !out_arg)

  737.             return 0;

  738.     }

  739. 

  740.     if(!in_arg){

  741.         if(s->resample){

  742.             if (!s->flushed)

  743.                 s->resampler->flush(s);

  744.             s->resample_in_constraint = 0;

  745.             s->flushed = 1;

  746.         }else if(!s->in_buffer_count){

  747.             return 0;

  748.         }

  749.     }else

  750.         fill_audiodata(in ,  (void*)in_arg);

  751. 

  752.     fill_audiodata(out, out_arg);

  753. 

  754.     if(s->resample){

  755.         int ret = swr_convert_internal(s, out, out_count, in, in_count);

  756.         if(ret>0 && !s->drop_output)

  757.             s->outpts += ret * (int64_t)s->in_sample_rate;

  758.         return ret;

  759.     }else{

  760.         AudioData tmp= *in;

  761.         int ret2=0;

  762.         int ret, size;

  763.         size = FFMIN(out_count, s->in_buffer_count);

  764.         if(size){

  765.             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);

  766.             ret= swr_convert_internal(s, out, size, &tmp, size);

  767.             if(ret<0)

  768.                 return ret;

  769.             ret2= ret;

  770.             s->in_buffer_count -= ret;

  771.             s->in_buffer_index += ret;

  772.             buf_set(out, out, ret);

  773.             out_count -= ret;

  774.             if(!s->in_buffer_count)

  775.                 s->in_buffer_index = 0;

  776.         }

  777. 

  778.         if(in_count){

  779.             size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;

  780. 

  781.             if(in_count > out_count) { //FIXME move after swr_convert_internal

  782.                 if(   size > s->in_buffer.count

  783.                 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){

  784.                     buf_set(&tmp, &s->in_buffer, s->in_buffer_index);

  785.                     copy(&s->in_buffer, &tmp, s->in_buffer_count);

  786.                     s->in_buffer_index=0;

  787.                 }else

  788.                     if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)

  789.                         return ret;

  790.             }

  791. 

  792.             if(out_count){

  793.                 size = FFMIN(in_count, out_count);

  794.                 ret= swr_convert_internal(s, out, size, in, size);

  795.                 if(ret<0)

  796.                     return ret;

  797.                 buf_set(in, in, ret);

  798.                 in_count -= ret;

  799.                 ret2 += ret;

  800.             }

  801.             if(in_count){

  802.                 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);

  803.                 copy(&tmp, in, in_count);

  804.                 s->in_buffer_count += in_count;

  805.             }

  806.         }

  807.         if(ret2>0 && !s->drop_output)

  808.             s->outpts += ret2 * (int64_t)s->in_sample_rate;

  809.         return ret2;

  810.     }

  811. }

  812. 

  813. int swr_drop_output(struct SwrContext *s, int count){

  814.     s->drop_output += count;

  815. 

  816.     if(s->drop_output <= 0)

  817.         return 0;

  818. 

  819.     av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);

  820.     return swr_convert(s, NULL, s->drop_output, NULL, 0);

  821. }

  822. 

  823. int swr_inject_silence(struct SwrContext *s, int count){

  824.     int ret, i;

  825.     uint8_t *tmp_arg[SWR_CH_MAX];

  826. 

  827.     if(count <= 0)

  828.         return 0;

  829. 

  830. #define MAX_SILENCE_STEP 16384

  831.     while (count > MAX_SILENCE_STEP) {

  832.         if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)

  833.             return ret;

  834.         count -= MAX_SILENCE_STEP;

  835.     }

  836. 

  837.     if((ret=swri_realloc_audio(&s->silence, count))<0)

  838.         return ret;

  839. 

  840.     if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {

  841.         memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);

  842.     } else

  843.         memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);

  844. 

  845.     reversefill_audiodata(&s->silence, tmp_arg);

  846.     av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);

  847.     ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);

  848.     return ret;

  849. }

  850. 

  851. int64_t swr_get_delay(struct SwrContext *s, int64_t base){

  852.     if (s->resampler && s->resample){

  853.         return s->resampler->get_delay(s, base);

  854.     }else{

  855.         return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;

  856.     }

  857. }

  858. 

  859. int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){

  860.     int ret;

  861. 

  862.     if (!s || compensation_distance < 0)

  863.         return AVERROR(EINVAL);

  864.     if (!compensation_distance && sample_delta)

  865.         return AVERROR(EINVAL);

  866.     if (!s->resample) {

  867.         s->flags |= SWR_FLAG_RESAMPLE;

  868.         ret = swr_init(s);

  869.         if (ret < 0)

  870.             return ret;

  871.     }

  872.     if (!s->resampler->set_compensation){

  873.         return AVERROR(EINVAL);

  874.     }else{

  875.         return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);

  876.     }

  877. }

  878. 

  879. int64_t swr_next_pts(struct SwrContext *s, int64_t pts){

  880.     if(pts == INT64_MIN)

  881.         return s->outpts;

  882.     if(s->min_compensation >= FLT_MAX) {

  883.         return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));

  884.     } else {

  885.         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;

  886.         double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);

  887. 

  888.         if(fabs(fdelta) > s->min_compensation) {

  889.             if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){

  890.                 int ret;

  891.                 if(delta > 0) ret = swr_inject_silence(s,  delta / s->out_sample_rate);

  892.                 else          ret = swr_drop_output   (s, -delta / s-> in_sample_rate);

  893.                 if(ret<0){

  894.                     av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);

  895.                 }

  896.             } else if(s->soft_compensation_duration && s->max_soft_compensation) {

  897.                 int duration = s->out_sample_rate * s->soft_compensation_duration;

  898.                 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);

  899.                 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;

  900.                 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);

  901.                 swr_set_compensation(s, comp, duration);

  902.             }

  903.         }

  904. 

  905.         return s->outpts;

  906.     }

  907. }