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FFmpeg/libswresample/swresample.c

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  1. /*

  2.  * Copyright (C) 2011-2012 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. 

  83. {"filter_size"          , "set swr resampling filter size", OFFSET(filter_size)  , AV_OPT_TYPE_INT  , {.i64=16                    }, 0      , INT_MAX   , PARAM },

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

  104. 

  105. { "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" },

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

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

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

  109. 

  110. { "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" },

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

  112.     { "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" },

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

  114. 

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

  116. 

  117. {0}

  118. };

  119. 

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

  121.     return "SWR";

  122. }

  123. 

  124. static const AVClass av_class = {

  125.     .class_name                = "SWResampler",

  126.     .item_name                 = context_to_name,

  127.     .option                    = options,

  128.     .version                   = LIBAVUTIL_VERSION_INT,

  129.     .log_level_offset_offset   = OFFSET(log_level_offset),

  130.     .parent_log_context_offset = OFFSET(log_ctx),

  131.     .category                  = AV_CLASS_CATEGORY_SWRESAMPLER,

  132. };

  133. 

  134. unsigned swresample_version(void)

  135. {

  136.     av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);

  137.     return LIBSWRESAMPLE_VERSION_INT;

  138. }

  139. 

  140. const char *swresample_configuration(void)

  141. {

  142.     return FFMPEG_CONFIGURATION;

  143. }

  144. 

  145. const char *swresample_license(void)

  146. {

  147. #define LICENSE_PREFIX "libswresample license: "

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

  149. }

  150. 

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

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

  153.         return AVERROR(EINVAL);

  154.     s->channel_map = channel_map;

  155.     return 0;

  156. }

  157. 

  158. const AVClass *swr_get_class(void)

  159. {

  160.     return &av_class;

  161. }

  162. 

  163. av_cold struct SwrContext *swr_alloc(void){

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

  165.     if(s){

  166.         s->av_class= &av_class;

  167.         av_opt_set_defaults(s);

  168.     }

  169.     return s;

  170. }

  171. 

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

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

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

  175.                                       int log_offset, void *log_ctx){

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

  177.     if(!s) return NULL;

  178. 

  179.     s->log_level_offset= log_offset;

  180.     s->log_ctx= log_ctx;

  181. 

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

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

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

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

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

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

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

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

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

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

  192.     return s;

  193. }

  194. 

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

  196.     a->fmt   = fmt;

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

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

  199. }

  200. 

  201. static void free_temp(AudioData *a){

  202.     av_free(a->data);

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

  204. }

  205. 

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

  207.     SwrContext *s= *ss;

  208.     if(s){

  209.         free_temp(&s->postin);

  210.         free_temp(&s->midbuf);

  211.         free_temp(&s->preout);

  212.         free_temp(&s->in_buffer);

  213.         free_temp(&s->dither);

  214.         swri_audio_convert_free(&s-> in_convert);

  215.         swri_audio_convert_free(&s->out_convert);

  216.         swri_audio_convert_free(&s->full_convert);

  217.         if (s->resampler)

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

  219.         swri_rematrix_free(s);

  220.     }

  221. 

  222.     av_freep(ss);

  223. }

  224. 

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

  226.     s->in_buffer_index= 0;

  227.     s->in_buffer_count= 0;

  228.     s->resample_in_constraint= 0;

  229.     free_temp(&s->postin);

  230.     free_temp(&s->midbuf);

  231.     free_temp(&s->preout);

  232.     free_temp(&s->in_buffer);

  233.     free_temp(&s->dither);

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

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

  236.     swri_audio_convert_free(&s-> in_convert);

  237.     swri_audio_convert_free(&s->out_convert);

  238.     swri_audio_convert_free(&s->full_convert);

  239.     swri_rematrix_free(s);

  240. 

  241.     s->flushed = 0;

  242. 

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

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

  245.         return AVERROR(EINVAL);

  246.     }

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

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

  249.         return AVERROR(EINVAL);

  250.     }

  251. 

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

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

  254.             s->int_sample_fmt= AV_SAMPLE_FMT_S16P;

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

  256.             s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;

  257.         }else{

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

  259.             s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;

  260.         }

  261.     }

  262. 

  263.     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P

  264.         &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P

  265.         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP

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

  267.         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));

  268.         return AVERROR(EINVAL);

  269.     }

  270. 

  271.     switch(s->engine){

  272. #if CONFIG_LIBSOXR

  273.         extern struct Resampler const soxr_resampler;

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

  275. #endif

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

  277.         default:

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

  279.             return AVERROR(EINVAL);

  280.     }

  281. 

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

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

  284. 

  285.     if (s->async) {

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

  287.             s->min_compensation = 0.001;

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

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

  290.         }

  291.     }

  292. 

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

  294.         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);

  295.     }else

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

  297.     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P

  298.         && s->int_sample_fmt != AV_SAMPLE_FMT_S32P

  299.         && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP

  300.         && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP

  301.         && s->resample){

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

  303.         return -1;

  304.     }

  305. 

  306.     if(!s->used_ch_count)

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

  308. 

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

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

  311.         s-> in_ch_layout= 0;

  312.     }

  313. 

  314.     if(!s-> in_ch_layout)

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

  316.     if(!s->out_ch_layout)

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

  318. 

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

  320.                  s->rematrix_custom;

  321. 

  322. #define RSC 1 //FIXME finetune

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

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

  325.     if(!s->used_ch_count)

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

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

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

  329. 

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

  331.         av_assert0(!s->in_ch_layout);

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

  333.         return -1;

  334.     }

  335. 

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

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

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

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

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

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

  342.         return -1;

  343.     }

  344. 

  345. av_assert0(s->used_ch_count);

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

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

  348. 

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

  350. 

  351.     if(!s->resample && !s->rematrix && !s->channel_map && !s->dither_method){

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

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

  354.         return 0;

  355.     }

  356. 

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

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

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

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

  361. 

  362. 

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

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

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

  366. 

  367.     if(s->channel_map){

  368.         s->postin.ch_count=

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

  370.         if(s->resample)

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

  372.     }

  373.     if(!s->resample_first){

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

  375.         if(s->resample)

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

  377.     }

  378. 

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

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

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

  382. 

  383.     if(s->resample){

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

  385.     }

  386. 

  387.     s->dither = s->preout;

  388. 

  389.     if(s->rematrix || s->dither_method)

  390.         return swri_rematrix_init(s);

  391. 

  392.     return 0;

  393. }

  394. 

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

  396.     int i, countb;

  397.     AudioData old;

  398. 

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

  400.         return AVERROR(EINVAL);

  401. 

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

  403.         return 0;

  404. 

  405.     count*=2;

  406. 

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

  408.     old= *a;

  409. 

  410.     av_assert0(a->bps);

  411.     av_assert0(a->ch_count);

  412. 

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

  414.     if(!a->data)

  415.         return AVERROR(ENOMEM);

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

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

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

  419.     }

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

  421.     av_free(old.data);

  422.     a->count= count;

  423. 

  424.     return 1;

  425. }

  426. 

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

  428.                  int count){

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

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

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

  432.     if(out->planar){

  433.         int ch;

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

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

  436.     }else

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

  438. }

  439. 

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

  441.     int i;

  442.     if(!in_arg){

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

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

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

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

  447.     }else{

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

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

  450.     }

  451. }

  452. 

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

  454.     int i;

  455.     if(out->planar){

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

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

  458.     }else{

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

  460.     }

  461. }

  462. 

  463. /**

  464.  *

  465.  * out may be equal in.

  466.  */

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

  468.     int ch;

  469.     if(in->planar){

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

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

  472.     }else{

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

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

  475.     }

  476. }

  477. 

  478. /**

  479.  *

  480.  * @return number of samples output per channel

  481.  */

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

  483.                              const AudioData * in_param, int in_count){

  484.     AudioData in, out, tmp;

  485.     int ret_sum=0;

  486.     int border=0;

  487. 

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

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

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

  491. 

  492.     tmp=out=*out_param;

  493.     in =  *in_param;

  494. 

  495.     do{

  496.         int ret, size, consumed;

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

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

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

  500.             out_count -= ret;

  501.             ret_sum += ret;

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

  503.             s->in_buffer_count -= consumed;

  504.             s->in_buffer_index += consumed;

  505. 

  506.             if(!in_count)

  507.                 break;

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

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

  510.                 in_count += s->in_buffer_count;

  511.                 s->in_buffer_count=0;

  512.                 s->in_buffer_index=0;

  513.                 border = 0;

  514.             }

  515.         }

  516. 

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

  518.             s->in_buffer_index=0;

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

  520.             out_count -= ret;

  521.             ret_sum += ret;

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

  523.             in_count -= consumed;

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

  525.         }

  526. 

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

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

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

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

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

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

  533.             s->in_buffer_index=0;

  534.         }else

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

  536.                 return ret;

  537. 

  538.         if(in_count){

  539.             int count= in_count;

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

  541. 

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

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

  544.             s->in_buffer_count += count;

  545.             in_count -= count;

  546.             border += count;

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

  548.             s->resample_in_constraint= 0;

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

  550.                 continue;

  551.         }

  552.         break;

  553.     }while(1);

  554. 

  555.     s->resample_in_constraint= !!out_count;

  556. 

  557.     return ret_sum;

  558. }

  559. 

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

  561.                                                       AudioData *in , int  in_count){

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

  563.     int ret/*, in_max*/;

  564.     AudioData preout_tmp, midbuf_tmp;

  565. 

  566.     if(s->full_convert){

  567.         av_assert0(!s->resample);

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

  569.         return out_count;

  570.     }

  571. 

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

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

  574. 

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

  576.         return ret;

  577.     if(s->resample_first){

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

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

  580.             return ret;

  581.     }else{

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

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

  584.             return ret;

  585.     }

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

  587.         return ret;

  588. 

  589.     postin= &s->postin;

  590. 

  591.     midbuf_tmp= s->midbuf;

  592.     midbuf= &midbuf_tmp;

  593.     preout_tmp= s->preout;

  594.     preout= &preout_tmp;

  595. 

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

  597.         postin= in;

  598. 

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

  600.         midbuf= postin;

  601. 

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

  603.         preout= midbuf;

  604. 

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

  606.         if(preout==in){

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

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

  609.             copy(out, in, out_count);

  610.             return out_count;

  611.         }

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

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

  614.         else                    preout= out;

  615.     }

  616. 

  617.     if(in != postin){

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

  619.     }

  620. 

  621.     if(s->resample_first){

  622.         if(postin != midbuf)

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

  624.         if(midbuf != preout)

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

  626.     }else{

  627.         if(postin != midbuf)

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

  629.         if(midbuf != preout)

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

  631.     }

  632. 

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

  634.         if(s->dither_method){

  635.             int ch;

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

  637.             av_assert0(preout != in);

  638. 

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

  640.                 return ret;

  641.             if(ret)

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

  643.                     swri_get_dither(s, s->dither.ch[ch], s->dither.count, 12345678913579<<ch, s->out_sample_fmt, s->int_sample_fmt);

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

  645. 

  646.             if(s->dither_pos + out_count > s->dither.count)

  647.                 s->dither_pos = 0;

  648. 

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

  650.                 s->mix_2_1_f(preout->ch[ch], preout->ch[ch], s->dither.ch[ch] + s->dither.bps * s->dither_pos, s->native_one, 0, 0, out_count);

  651. 

  652.             s->dither_pos += out_count;

  653.         }

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

  655.         swri_audio_convert(s->out_convert, out, preout, out_count);

  656.     }

  657.     return out_count;

  658. }

  659. 

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

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

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

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

  664. 

  665.     if(s->drop_output > 0){

  666.         int ret;

  667.         AudioData tmp = s->out;

  668.         uint8_t *tmp_arg[SWR_CH_MAX];

  669.         tmp.count = 0;

  670.         tmp.data  = NULL;

  671.         if((ret=swri_realloc_audio(&tmp, s->drop_output))<0)

  672.             return ret;

  673. 

  674.         reversefill_audiodata(&tmp, tmp_arg);

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

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

  677.         s->drop_output *= -1;

  678.         if(ret>0)

  679.             s->drop_output -= ret;

  680. 

  681.         av_freep(&tmp.data);

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

  683.             return 0;

  684.         in_count = 0;

  685.     }

  686. 

  687.     if(!in_arg){

  688.         if(s->resample){

  689.             if (!s->flushed)

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

  691.             s->resample_in_constraint = 0;

  692.             s->flushed = 1;

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

  694.             return 0;

  695.         }

  696.     }else

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

  698. 

  699.     fill_audiodata(out, out_arg);

  700. 

  701.     if(s->resample){

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

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

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

  705.         return ret;

  706.     }else{

  707.         AudioData tmp= *in;

  708.         int ret2=0;

  709.         int ret, size;

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

  711.         if(size){

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

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

  714.             if(ret<0)

  715.                 return ret;

  716.             ret2= ret;

  717.             s->in_buffer_count -= ret;

  718.             s->in_buffer_index += ret;

  719.             buf_set(out, out, ret);

  720.             out_count -= ret;

  721.             if(!s->in_buffer_count)

  722.                 s->in_buffer_index = 0;

  723.         }

  724. 

  725.         if(in_count){

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

  727. 

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

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

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

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

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

  733.                     s->in_buffer_index=0;

  734.                 }else

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

  736.                         return ret;

  737.             }

  738. 

  739.             if(out_count){

  740.                 size = FFMIN(in_count, out_count);

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

  742.                 if(ret<0)

  743.                     return ret;

  744.                 buf_set(in, in, ret);

  745.                 in_count -= ret;

  746.                 ret2 += ret;

  747.             }

  748.             if(in_count){

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

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

  751.                 s->in_buffer_count += in_count;

  752.             }

  753.         }

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

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

  756.         return ret2;

  757.     }

  758. }

  759. 

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

  761.     s->drop_output += count;

  762. 

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

  764.         return 0;

  765. 

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

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

  768. }

  769. 

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

  771.     int ret, i;

  772.     AudioData silence = s->in;

  773.     uint8_t *tmp_arg[SWR_CH_MAX];

  774. 

  775.     if(count <= 0)

  776.         return 0;

  777. 

  778.     silence.count = 0;

  779.     silence.data  = NULL;

  780.     if((ret=swri_realloc_audio(&silence, count))<0)

  781.         return ret;

  782. 

  783.     if(silence.planar) for(i=0; i<silence.ch_count; i++) {

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

  785.     } else

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

  787. 

  788.     reversefill_audiodata(&silence, tmp_arg);

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

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

  791.     av_freep(&silence.data);

  792.     return ret;

  793. }

  794. 

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

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

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

  798.     }else{

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

  800.     }

  801. }

  802. 

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

  804.     int ret;

  805. 

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

  807.         return AVERROR(EINVAL);

  808.     if (!compensation_distance && sample_delta)

  809.         return AVERROR(EINVAL);

  810.     if (!s->resample) {

  811.         s->flags |= SWR_FLAG_RESAMPLE;

  812.         ret = swr_init(s);

  813.         if (ret < 0)

  814.             return ret;

  815.     }

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

  817.         return AVERROR(EINVAL);

  818.     }else{

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

  820.     }

  821. }

  822. 

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

  824.     if(pts == INT64_MIN)

  825.         return s->outpts;

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

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

  828.     } else {

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

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

  831. 

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

  833.             if(!s->outpts || fabs(fdelta) > s->min_hard_compensation){

  834.                 int ret;

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

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

  837.                 if(ret<0){

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

  839.                 }

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

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

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

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

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

  845.                 swr_set_compensation(s, comp, duration);

  846.             }

  847.         }

  848. 

  849.         return s->outpts;

  850.     }

  851. }