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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   , INT_MAX, PARAM},

  55. {"in_sample_fmt"        , "set input sample format"     , OFFSET( in_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , INT_MAX, PARAM},

  56. {"osf"                  , "set output sample format"    , OFFSET(out_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , INT_MAX, PARAM},

  57. {"out_sample_fmt"       , "set output sample format"    , OFFSET(out_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , INT_MAX, PARAM},

  58. {"tsf"                  , "set internal sample format"  , OFFSET(int_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , INT_MAX, PARAM},

  59. {"internal_sample_fmt"  , "set internal sample format"  , OFFSET(int_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , INT_MAX, PARAM},

  60. {"icl"                  , "set input channel layout"    , OFFSET( in_ch_layout  ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},

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

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

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

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

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

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

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

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

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

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

  71. {"rematrix_maxval"      , "set rematrix maxval"         , OFFSET(rematrix_maxval), AV_OPT_TYPE_FLOAT, {.dbl=0.0                   }, 0      , 1000      , PARAM},

  72. 

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

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

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

  76. 

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

  78. 

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

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

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

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

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

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

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

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

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

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

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

  90. 

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

  92. {"phase_shift"          , "set swr resampling phase shift", OFFSET(phase_shift)  , AV_OPT_TYPE_INT  , {.i64=10                    }, 0      , 24        , PARAM },

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

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

  95. 

  96. /* duplicate option in order to work with avconv */

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

  98. 

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

  118. 

  119. { "matrix_encoding"     , "set matrixed stereo encoding" , OFFSET(matrix_encoding), AV_OPT_TYPE_INT   ,{.i64 = AV_MATRIX_ENCODING_NONE}, AV_MATRIX_ENCODING_NONE,     AV_MATRIX_ENCODING_NB-1, PARAM, "matrix_encoding" },

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

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

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

  123. 

  124. { "filter_type"         , "select swr filter type"      , OFFSET(filter_type)    , AV_OPT_TYPE_INT  , { .i64 = SWR_FILTER_TYPE_KAISER }, SWR_FILTER_TYPE_CUBIC, SWR_FILTER_TYPE_KAISER, PARAM, "filter_type" },

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

  126.     { "blackman_nuttall", "select Blackman Nuttall Windowed Sinc", 0             , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" },

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

  128. 

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

  130. 

  131. { "output_sample_bits"  , "set swr number of output sample bits", OFFSET(dither.output_sample_bits), AV_OPT_TYPE_INT  , {.i64=0   }, 0      , 64        , PARAM },

  132. {0}

  133. };

  134. 

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

  136.     return "SWR";

  137. }

  138. 

  139. static const AVClass av_class = {

  140.     .class_name                = "SWResampler",

  141.     .item_name                 = context_to_name,

  142.     .option                    = options,

  143.     .version                   = LIBAVUTIL_VERSION_INT,

  144.     .log_level_offset_offset   = OFFSET(log_level_offset),

  145.     .parent_log_context_offset = OFFSET(log_ctx),

  146.     .category                  = AV_CLASS_CATEGORY_SWRESAMPLER,

  147. };

  148. 

  149. unsigned swresample_version(void)

  150. {

  151.     av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);

  152.     return LIBSWRESAMPLE_VERSION_INT;

  153. }

  154. 

  155. const char *swresample_configuration(void)

  156. {

  157.     return FFMPEG_CONFIGURATION;

  158. }

  159. 

  160. const char *swresample_license(void)

  161. {

  162. #define LICENSE_PREFIX "libswresample license: "

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

  164. }

  165. 

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

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

  168.         return AVERROR(EINVAL);

  169.     s->channel_map = channel_map;

  170.     return 0;

  171. }

  172. 

  173. const AVClass *swr_get_class(void)

  174. {

  175.     return &av_class;

  176. }

  177. 

  178. av_cold struct SwrContext *swr_alloc(void){

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

  180.     if(s){

  181.         s->av_class= &av_class;

  182.         av_opt_set_defaults(s);

  183.     }

  184.     return s;

  185. }

  186. 

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

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

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

  190.                                       int log_offset, void *log_ctx){

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

  192.     if(!s) return NULL;

  193. 

  194.     s->log_level_offset= log_offset;

  195.     s->log_ctx= log_ctx;

  196. 

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

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

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

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

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

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

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

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

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

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

  207.     return s;

  208. }

  209. 

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

  211.     a->fmt   = fmt;

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

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

  214. }

  215. 

  216. static void free_temp(AudioData *a){

  217.     av_free(a->data);

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

  219. }

  220. 

  221. static void clear_context(SwrContext *s){

  222.     s->in_buffer_index= 0;

  223.     s->in_buffer_count= 0;

  224.     s->resample_in_constraint= 0;

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

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

  227.     free_temp(&s->postin);

  228.     free_temp(&s->midbuf);

  229.     free_temp(&s->preout);

  230.     free_temp(&s->in_buffer);

  231.     free_temp(&s->silence);

  232.     free_temp(&s->drop_temp);

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

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

  235.     swri_audio_convert_free(&s-> in_convert);

  236.     swri_audio_convert_free(&s->out_convert);

  237.     swri_audio_convert_free(&s->full_convert);

  238.     swri_rematrix_free(s);

  239. 

  240.     s->flushed = 0;

  241. }

  242. 

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

  244.     SwrContext *s= *ss;

  245.     if(s){

  246.         clear_context(s);

  247.         if (s->resampler)

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

  249.     }

  250. 

  251.     av_freep(ss);

  252. }

  253. 

  254. av_cold void swr_close(SwrContext *s){

  255.     clear_context(s);

  256. }

  257. 

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

  259.     int ret;

  260. 

  261.     clear_context(s);

  262. 

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

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

  265.         return AVERROR(EINVAL);

  266.     }

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

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

  269.         return AVERROR(EINVAL);

  270.     }

  271. 

  272.     if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) {

  273.         av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);

  274.         s->in_ch_layout = 0;

  275.     }

  276. 

  277.     if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) {

  278.         av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);

  279.         s->out_ch_layout = 0;

  280.     }

  281. 

  282.     switch(s->engine){

  283. #if CONFIG_LIBSOXR

  284.         extern struct Resampler const soxr_resampler;

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

  286. #endif

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

  288.         default:

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

  290.             return AVERROR(EINVAL);

  291.     }

  292. 

  293.     if(!s->used_ch_count)

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

  295. 

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

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

  298.         s-> in_ch_layout= 0;

  299.     }

  300. 

  301.     if(!s-> in_ch_layout)

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

  303.     if(!s->out_ch_layout)

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

  305. 

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

  307.                  s->rematrix_custom;

  308. 

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

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

  311.             s->int_sample_fmt= AV_SAMPLE_FMT_S16P;

  312.         }else if(   av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P

  313.                  && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P

  314.                  && !s->rematrix

  315.                  && s->engine != SWR_ENGINE_SOXR){

  316.             s->int_sample_fmt= AV_SAMPLE_FMT_S32P;

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

  318.             s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;

  319.         }else{

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

  321.             s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;

  322.         }

  323.     }

  324. 

  325.     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P

  326.         &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P

  327.         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP

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

  329.         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));

  330.         return AVERROR(EINVAL);

  331.     }

  332. 

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

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

  335. 

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

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

  338.             s->async = 1;

  339.         s->firstpts =

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

  341.     } else

  342.         s->firstpts = AV_NOPTS_VALUE;

  343. 

  344.     if (s->async) {

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

  346.             s->min_compensation = 0.001;

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

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

  349.         }

  350.     }

  351. 

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

  353.         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);

  354.     }else

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

  356.     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P

  357.         && s->int_sample_fmt != AV_SAMPLE_FMT_S32P

  358.         && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP

  359.         && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP

  360.         && s->resample){

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

  362.         return -1;

  363.     }

  364. 

  365. #define RSC 1 //FIXME finetune

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

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

  368.     if(!s->used_ch_count)

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

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

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

  372. 

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

  374.         av_assert0(!s->in_ch_layout);

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

  376.         return -1;

  377.     }

  378. 

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

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

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

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

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

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

  385.         return -1;

  386.     }

  387. 

  388. av_assert0(s->used_ch_count);

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

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

  391. 

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

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

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

  395. 

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

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

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

  399.         return 0;

  400.     }

  401. 

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

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

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

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

  406. 

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

  408.         return AVERROR(ENOMEM);

  409. 

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

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

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

  413. 

  414.     if(s->channel_map){

  415.         s->postin.ch_count=

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

  417.         if(s->resample)

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

  419.     }

  420.     if(!s->resample_first){

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

  422.         if(s->resample)

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

  424.     }

  425. 

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

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

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

  429. 

  430.     if(s->resample){

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

  432.     }

  433. 

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

  435.         return ret;

  436. 

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

  438.         return swri_rematrix_init(s);

  439. 

  440.     return 0;

  441. }

  442. 

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

  444.     int i, countb;

  445.     AudioData old;

  446. 

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

  448.         return AVERROR(EINVAL);

  449. 

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

  451.         return 0;

  452. 

  453.     count*=2;

  454. 

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

  456.     old= *a;

  457. 

  458.     av_assert0(a->bps);

  459.     av_assert0(a->ch_count);

  460. 

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

  462.     if(!a->data)

  463.         return AVERROR(ENOMEM);

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

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

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

  467.     }

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

  469.     av_freep(&old.data);

  470.     a->count= count;

  471. 

  472.     return 1;

  473. }

  474. 

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

  476.                  int count){

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

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

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

  480.     if(out->planar){

  481.         int ch;

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

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

  484.     }else

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

  486. }

  487. 

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

  489.     int i;

  490.     if(!in_arg){

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

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

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

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

  495.     }else{

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

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

  498.     }

  499. }

  500. 

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

  502.     int i;

  503.     if(out->planar){

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

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

  506.     }else{

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

  508.     }

  509. }

  510. 

  511. /**

  512.  *

  513.  * out may be equal in.

  514.  */

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

  516.     int ch;

  517.     if(in->planar){

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

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

  520.     }else{

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

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

  523.     }

  524. }

  525. 

  526. /**

  527.  *

  528.  * @return number of samples output per channel

  529.  */

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

  531.                              const AudioData * in_param, int in_count){

  532.     AudioData in, out, tmp;

  533.     int ret_sum=0;

  534.     int border=0;

  535.     int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;

  536. 

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

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

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

  540. 

  541.     tmp=out=*out_param;

  542.     in =  *in_param;

  543. 

  544.     do{

  545.         int ret, size, consumed;

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

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

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

  549.             out_count -= ret;

  550.             ret_sum += ret;

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

  552.             s->in_buffer_count -= consumed;

  553.             s->in_buffer_index += consumed;

  554. 

  555.             if(!in_count)

  556.                 break;

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

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

  559.                 in_count += s->in_buffer_count;

  560.                 s->in_buffer_count=0;

  561.                 s->in_buffer_index=0;

  562.                 border = 0;

  563.             }

  564.         }

  565. 

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

  567.             s->in_buffer_index=0;

  568.             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);

  569.             out_count -= ret;

  570.             ret_sum += ret;

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

  572.             in_count -= consumed;

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

  574.         }

  575. 

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

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

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

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

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

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

  582.             s->in_buffer_index=0;

  583.         }else

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

  585.                 return ret;

  586. 

  587.         if(in_count){

  588.             int count= in_count;

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

  590. 

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

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

  593.             s->in_buffer_count += count;

  594.             in_count -= count;

  595.             border += count;

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

  597.             s->resample_in_constraint= 0;

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

  599.                 continue;

  600.             if (padless) {

  601.                 padless = 0;

  602.                 continue;

  603.             }

  604.         }

  605.         break;

  606.     }while(1);

  607. 

  608.     s->resample_in_constraint= !!out_count;

  609. 

  610.     return ret_sum;

  611. }

  612. 

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

  614.                                                       AudioData *in , int  in_count){

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

  616.     int ret/*, in_max*/;

  617.     AudioData preout_tmp, midbuf_tmp;

  618. 

  619.     if(s->full_convert){

  620.         av_assert0(!s->resample);

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

  622.         return out_count;

  623.     }

  624. 

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

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

  627. 

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

  629.         return ret;

  630.     if(s->resample_first){

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

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

  633.             return ret;

  634.     }else{

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

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

  637.             return ret;

  638.     }

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

  640.         return ret;

  641. 

  642.     postin= &s->postin;

  643. 

  644.     midbuf_tmp= s->midbuf;

  645.     midbuf= &midbuf_tmp;

  646.     preout_tmp= s->preout;

  647.     preout= &preout_tmp;

  648. 

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

  650.         postin= in;

  651. 

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

  653.         midbuf= postin;

  654. 

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

  656.         preout= midbuf;

  657. 

  658.     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar

  659.        && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){

  660.         if(preout==in){

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

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

  663.             copy(out, in, out_count);

  664.             return out_count;

  665.         }

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

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

  668.         else                    preout= out;

  669.     }

  670. 

  671.     if(in != postin){

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

  673.     }

  674. 

  675.     if(s->resample_first){

  676.         if(postin != midbuf)

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

  678.         if(midbuf != preout)

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

  680.     }else{

  681.         if(postin != midbuf)

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

  683.         if(midbuf != preout)

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

  685.     }

  686. 

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

  688.         AudioData *conv_src = preout;

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

  690.             int ch;

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

  692. 

  693.             if (preout == in) {

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

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

  696.                     return ret;

  697.             }

  698. 

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

  700.                 return ret;

  701.             if(ret)

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

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

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

  705. 

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

  707.                 s->dither.noise_pos = 0;

  708. 

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

  710.                 if (s->mix_2_1_simd) {

  711.                     int len1= out_count&~15;

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

  713. 

  714.                     if(len1)

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

  716.                             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);

  717.                     if(out_count != len1)

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

  719.                             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);

  720.                 } else {

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

  722.                         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);

  723.                 }

  724.             } else {

  725.                 switch(s->int_sample_fmt) {

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

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

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

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

  730.                 }

  731.             }

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

  733.         }

  734. //FIXME packed doesn't need more than 1 chan here!

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

  736.     }

  737.     return out_count;

  738. }

  739. 

  740. int swr_is_initialized(struct SwrContext *s) {

  741.     return !!s->in_buffer.ch_count;

  742. }

  743. 

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

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

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

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

  748. 

  749.     if (!swr_is_initialized(s)) {

  750.         av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");

  751.         return AVERROR(EINVAL);

  752.     }

  753. 

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

  755.         int ret;

  756.         uint8_t *tmp_arg[SWR_CH_MAX];

  757. #define MAX_DROP_STEP 16384

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

  759.             return ret;

  760. 

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

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

  763.         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

  764.         s->drop_output *= -1;

  765.         in_count = 0;

  766.         if(ret>0) {

  767.             s->drop_output -= ret;

  768.             continue;

  769.         }

  770. 

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

  772.             return 0;

  773.     }

  774. 

  775.     if(!in_arg){

  776.         if(s->resample){

  777.             if (!s->flushed)

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

  779.             s->resample_in_constraint = 0;

  780.             s->flushed = 1;

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

  782.             return 0;

  783.         }

  784.     }else

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

  786. 

  787.     fill_audiodata(out, out_arg);

  788. 

  789.     if(s->resample){

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

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

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

  793.         return ret;

  794.     }else{

  795.         AudioData tmp= *in;

  796.         int ret2=0;

  797.         int ret, size;

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

  799.         if(size){

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

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

  802.             if(ret<0)

  803.                 return ret;

  804.             ret2= ret;

  805.             s->in_buffer_count -= ret;

  806.             s->in_buffer_index += ret;

  807.             buf_set(out, out, ret);

  808.             out_count -= ret;

  809.             if(!s->in_buffer_count)

  810.                 s->in_buffer_index = 0;

  811.         }

  812. 

  813.         if(in_count){

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

  815. 

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

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

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

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

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

  821.                     s->in_buffer_index=0;

  822.                 }else

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

  824.                         return ret;

  825.             }

  826. 

  827.             if(out_count){

  828.                 size = FFMIN(in_count, out_count);

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

  830.                 if(ret<0)

  831.                     return ret;

  832.                 buf_set(in, in, ret);

  833.                 in_count -= ret;

  834.                 ret2 += ret;

  835.             }

  836.             if(in_count){

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

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

  839.                 s->in_buffer_count += in_count;

  840.             }

  841.         }

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

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

  844.         return ret2;

  845.     }

  846. }

  847. 

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

  849.     s->drop_output += count;

  850. 

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

  852.         return 0;

  853. 

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

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

  856. }

  857. 

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

  859.     int ret, i;

  860.     uint8_t *tmp_arg[SWR_CH_MAX];

  861. 

  862.     if(count <= 0)

  863.         return 0;

  864. 

  865. #define MAX_SILENCE_STEP 16384

  866.     while (count > MAX_SILENCE_STEP) {

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

  868.             return ret;

  869.         count -= MAX_SILENCE_STEP;

  870.     }

  871. 

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

  873.         return ret;

  874. 

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

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

  877.     } else

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

  879. 

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

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

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

  883.     return ret;

  884. }

  885. 

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

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

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

  889.     }else{

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

  891.     }

  892. }

  893. 

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

  895.     int ret;

  896. 

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

  898.         return AVERROR(EINVAL);

  899.     if (!compensation_distance && sample_delta)

  900.         return AVERROR(EINVAL);

  901.     if (!s->resample) {

  902.         s->flags |= SWR_FLAG_RESAMPLE;

  903.         ret = swr_init(s);

  904.         if (ret < 0)

  905.             return ret;

  906.     }

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

  908.         return AVERROR(EINVAL);

  909.     }else{

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

  911.     }

  912. }

  913. 

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

  915.     if(pts == INT64_MIN)

  916.         return s->outpts;

  917. 

  918.     if (s->firstpts == AV_NOPTS_VALUE)

  919.         s->outpts = s->firstpts = pts;

  920. 

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

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

  923.     } else {

  924.         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;

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

  926. 

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

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

  929.                 int ret;

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

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

  932.                 if(ret<0){

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

  934.                 }

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

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

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

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

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

  940.                 swr_set_compensation(s, comp, duration);

  941.             }

  942.         }

  943. 

  944.         return s->outpts;

  945.     }

  946. }