swresample: include ff_log2_tab for shared builds
[ffmpeg.git] / libswresample / swresample.c
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/audioconvert.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"                  ,  "Input Channel Count"        , OFFSET( in.ch_count   ), AV_OPT_TYPE_INT  , {.i64=2                     }, 0      , SWR_CH_MAX, PARAM},
45 {"in_channel_count"     ,  "Input Channel Count"        , OFFSET( in.ch_count   ), AV_OPT_TYPE_INT  , {.i64=2                     }, 0      , SWR_CH_MAX, PARAM},
46 {"och"                  , "Output Channel Count"        , OFFSET(out.ch_count   ), AV_OPT_TYPE_INT  , {.i64=2                     }, 0      , SWR_CH_MAX, PARAM},
47 {"out_channel_count"    , "Output Channel Count"        , OFFSET(out.ch_count   ), AV_OPT_TYPE_INT  , {.i64=2                     }, 0      , SWR_CH_MAX, PARAM},
48 {"uch"                  ,   "Used Channel Count"        , OFFSET(used_ch_count  ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
49 {"used_channel_count"   ,   "Used Channel Count"        , OFFSET(used_ch_count  ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
50 {"isr"                  ,  "Input Sample Rate"          , OFFSET( in_sample_rate), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , INT_MAX   , PARAM},
51 {"in_sample_rate"       ,  "Input Sample Rate"          , OFFSET( in_sample_rate), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , INT_MAX   , PARAM},
52 {"osr"                  , "Output Sample Rate"          , OFFSET(out_sample_rate), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , INT_MAX   , PARAM},
53 {"out_sample_rate"      , "Output Sample Rate"          , OFFSET(out_sample_rate), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , INT_MAX   , PARAM},
54 {"isf"                  ,    "Input Sample Format"      , OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT  , {.i64=AV_SAMPLE_FMT_NONE    }, -1     , AV_SAMPLE_FMT_NB-1+256, PARAM},
55 {"in_sample_fmt"        ,    "Input Sample Format"      , OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT  , {.i64=AV_SAMPLE_FMT_NONE    }, -1     , AV_SAMPLE_FMT_NB-1+256, PARAM},
56 {"osf"                  ,   "Output Sample Format"      , OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT  , {.i64=AV_SAMPLE_FMT_NONE    }, -1     , AV_SAMPLE_FMT_NB-1+256, PARAM},
57 {"out_sample_fmt"       ,   "Output Sample Format"      , OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT  , {.i64=AV_SAMPLE_FMT_NONE    }, -1     , AV_SAMPLE_FMT_NB-1+256, PARAM},
58 {"tsf"                  , "Internal Sample Format"      , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT  , {.i64=AV_SAMPLE_FMT_NONE    }, -1     , AV_SAMPLE_FMT_FLTP, PARAM},
59 {"internal_sample_fmt"  , "Internal Sample Format"      , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT  , {.i64=AV_SAMPLE_FMT_NONE    }, -1     , AV_SAMPLE_FMT_FLTP, PARAM},
60 {"icl"                  ,   "Input Channel Layout"      , OFFSET( in_ch_layout  ), AV_OPT_TYPE_INT64, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
61 {"in_channel_layout"    ,   "Input Channel Layout"      , OFFSET( in_ch_layout  ), AV_OPT_TYPE_INT64, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
62 {"ocl"                  ,  "Output Channel Layout"      , OFFSET(out_ch_layout  ), AV_OPT_TYPE_INT64, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
63 {"out_channel_layout"   ,  "Output Channel Layout"      , OFFSET(out_ch_layout  ), AV_OPT_TYPE_INT64, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
64 {"clev"                 ,    "Center Mix Level"         , OFFSET(clev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
65 {"center_mix_level"     ,    "Center Mix Level"         , OFFSET(clev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
66 {"slev"                 , "Sourround Mix Level"         , OFFSET(slev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
67 {"surround_mix_level"   , "Sourround Mix Level"         , OFFSET(slev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
68 {"lfe_mix_level"        , "LFE Mix Level"               , OFFSET(lfe_mix_level  ), AV_OPT_TYPE_FLOAT, {.dbl=0                     }, -32    , 32        , PARAM},
69 {"rmvol"                , "Rematrix Volume"             , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0                   }, -1000  , 1000      , PARAM},
70 {"rematrix_volume"      , "Rematrix Volume"             , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0                   }, -1000  , 1000      , PARAM},
71 {"flags"                , NULL                          , OFFSET(flags          ), AV_OPT_TYPE_FLAGS, {.i64=0                     }, 0      , UINT_MAX  , PARAM, "flags"},
72 {"swr_flags"            , NULL                          , OFFSET(flags          ), AV_OPT_TYPE_FLAGS, {.i64=0                     }, 0      , UINT_MAX  , PARAM, "flags"},
73 {"res"                  , "Force Resampling"            , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_FLAG_RESAMPLE     }, INT_MIN, INT_MAX   , PARAM, "flags"},
74 {"dither_scale"         , "Dither Scale"                , OFFSET(dither_scale   ), AV_OPT_TYPE_FLOAT, {.dbl=1                     }, 0      , INT_MAX   , PARAM},
75 {"dither_method"        , "Dither Method"               , OFFSET(dither_method  ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_DITHER_NB-1, PARAM, "dither_method"},
76 {"rectangular"          , "Rectangular Dither"          , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_RECTANGULAR}, INT_MIN, INT_MAX   , PARAM, "dither_method"},
77 {"triangular"           ,  "Triangular Dither"          , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX   , PARAM, "dither_method"},
78 {"triangular_hp"        , "Triangular Dither With High Pass" , 0                 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR_HIGHPASS }, INT_MIN, INT_MAX, PARAM, "dither_method"},
79 {"filter_size"          , "Resampling Filter Size"      , OFFSET(filter_size)    , AV_OPT_TYPE_INT  , {.i64=16                    }, 0      , INT_MAX   , PARAM },
80 {"phase_shift"          , "Resampling Phase Shift"      , OFFSET(phase_shift)    , AV_OPT_TYPE_INT  , {.i64=10                    }, 0      , 30        , PARAM },
81 {"linear_interp"        , "Use Linear Interpolation"    , OFFSET(linear_interp)  , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , 1         , PARAM },
82 {"cutoff"               , "Cutoff Frequency Ratio"      , OFFSET(cutoff)         , AV_OPT_TYPE_DOUBLE,{.dbl=0.8                   }, 0      , 1         , PARAM },
83 {"min_comp"             , "Minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"
84                                                         , OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX               }, 0      , FLT_MAX   , PARAM },
85 {"min_hard_comp"        , "Minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."
86                                                    , OFFSET(min_hard_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0.1                   }, 0      , INT_MAX   , PARAM },
87 {"comp_duration"        , "Duration (in seconds) over which data is stretched/squeezed to make it match the timestamps."
88                                               , OFFSET(soft_compensation_duration),AV_OPT_TYPE_FLOAT ,{.dbl=1                     }, 0      , INT_MAX   , PARAM },
89 {"max_soft_comp"        , "Maximum factor by which data is stretched/squeezed to make it match the timestamps."
90                                                    , OFFSET(max_soft_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0                     }, INT_MIN, INT_MAX   , PARAM },
91 { "matrix_encoding"     , "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" },
92     { "none",  "None",               0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_NONE  }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
93     { "dolby", "Dolby",              0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
94     { "dplii", "Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
95 { "filter_type"         , "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" },
96     { "cubic"           , "Cubic"                       , 0                      , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC            }, INT_MIN, INT_MAX, PARAM, "filter_type" },
97     { "blackman_nuttall", "Blackman Nuttall Windowed Sinc", 0                    , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" },
98     { "kaiser"          , "Kaiser Windowed Sinc"        , 0                      , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER           }, INT_MIN, INT_MAX, PARAM, "filter_type" },
99 { "kaiser_beta"         , "Kaiser Window Beta"          ,OFFSET(kaiser_beta)     , AV_OPT_TYPE_INT  , {.i64=9                     }, 2      , 16        , PARAM },
100
101 {0}
102 };
103
104 static const char* context_to_name(void* ptr) {
105     return "SWR";
106 }
107
108 static const AVClass av_class = {
109     .class_name                = "SWResampler",
110     .item_name                 = context_to_name,
111     .option                    = options,
112     .version                   = LIBAVUTIL_VERSION_INT,
113     .log_level_offset_offset   = OFFSET(log_level_offset),
114     .parent_log_context_offset = OFFSET(log_ctx),
115     .category                  = AV_CLASS_CATEGORY_SWRESAMPLER,
116 };
117
118 unsigned swresample_version(void)
119 {
120     av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
121     return LIBSWRESAMPLE_VERSION_INT;
122 }
123
124 const char *swresample_configuration(void)
125 {
126     return FFMPEG_CONFIGURATION;
127 }
128
129 const char *swresample_license(void)
130 {
131 #define LICENSE_PREFIX "libswresample license: "
132     return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
133 }
134
135 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
136     if(!s || s->in_convert) // s needs to be allocated but not initialized
137         return AVERROR(EINVAL);
138     s->channel_map = channel_map;
139     return 0;
140 }
141
142 const AVClass *swr_get_class(void)
143 {
144     return &av_class;
145 }
146
147 av_cold struct SwrContext *swr_alloc(void){
148     SwrContext *s= av_mallocz(sizeof(SwrContext));
149     if(s){
150         s->av_class= &av_class;
151         av_opt_set_defaults(s);
152     }
153     return s;
154 }
155
156 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
157                                       int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
158                                       int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
159                                       int log_offset, void *log_ctx){
160     if(!s) s= swr_alloc();
161     if(!s) return NULL;
162
163     s->log_level_offset= log_offset;
164     s->log_ctx= log_ctx;
165
166     av_opt_set_int(s, "ocl", out_ch_layout,   0);
167     av_opt_set_int(s, "osf", out_sample_fmt,  0);
168     av_opt_set_int(s, "osr", out_sample_rate, 0);
169     av_opt_set_int(s, "icl", in_ch_layout,    0);
170     av_opt_set_int(s, "isf", in_sample_fmt,   0);
171     av_opt_set_int(s, "isr", in_sample_rate,  0);
172     av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE,   0);
173     av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);
174     av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);
175     av_opt_set_int(s, "uch", 0, 0);
176     return s;
177 }
178
179 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
180     a->fmt   = fmt;
181     a->bps   = av_get_bytes_per_sample(fmt);
182     a->planar= av_sample_fmt_is_planar(fmt);
183 }
184
185 static void free_temp(AudioData *a){
186     av_free(a->data);
187     memset(a, 0, sizeof(*a));
188 }
189
190 av_cold void swr_free(SwrContext **ss){
191     SwrContext *s= *ss;
192     if(s){
193         free_temp(&s->postin);
194         free_temp(&s->midbuf);
195         free_temp(&s->preout);
196         free_temp(&s->in_buffer);
197         free_temp(&s->dither);
198         swri_audio_convert_free(&s-> in_convert);
199         swri_audio_convert_free(&s->out_convert);
200         swri_audio_convert_free(&s->full_convert);
201         swri_resample_free(&s->resample);
202         swri_rematrix_free(s);
203     }
204
205     av_freep(ss);
206 }
207
208 av_cold int swr_init(struct SwrContext *s){
209     s->in_buffer_index= 0;
210     s->in_buffer_count= 0;
211     s->resample_in_constraint= 0;
212     free_temp(&s->postin);
213     free_temp(&s->midbuf);
214     free_temp(&s->preout);
215     free_temp(&s->in_buffer);
216     free_temp(&s->dither);
217     memset(s->in.ch, 0, sizeof(s->in.ch));
218     memset(s->out.ch, 0, sizeof(s->out.ch));
219     swri_audio_convert_free(&s-> in_convert);
220     swri_audio_convert_free(&s->out_convert);
221     swri_audio_convert_free(&s->full_convert);
222     swri_rematrix_free(s);
223
224     s->flushed = 0;
225
226     if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
227         av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
228         return AVERROR(EINVAL);
229     }
230     if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
231         av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
232         return AVERROR(EINVAL);
233     }
234
235     if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
236         if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
237             s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
238         }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
239             s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
240         }else{
241             av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
242             s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
243         }
244     }
245
246     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P
247         &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
248         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
249         &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
250         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));
251         return AVERROR(EINVAL);
252     }
253
254     set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
255     set_audiodata_fmt(&s->out, s->out_sample_fmt);
256
257     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
258         s->resample = swri_resample_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);
259     }else
260         swri_resample_free(&s->resample);
261     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P
262         && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
263         && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
264         && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
265         && s->resample){
266         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
267         return -1;
268     }
269
270     if(!s->used_ch_count)
271         s->used_ch_count= s->in.ch_count;
272
273     if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
274         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
275         s-> in_ch_layout= 0;
276     }
277
278     if(!s-> in_ch_layout)
279         s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
280     if(!s->out_ch_layout)
281         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
282
283     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
284                  s->rematrix_custom;
285
286 #define RSC 1 //FIXME finetune
287     if(!s-> in.ch_count)
288         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
289     if(!s->used_ch_count)
290         s->used_ch_count= s->in.ch_count;
291     if(!s->out.ch_count)
292         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
293
294     if(!s-> in.ch_count){
295         av_assert0(!s->in_ch_layout);
296         av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
297         return -1;
298     }
299
300     if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
301         av_log(s, AV_LOG_ERROR, "Rematrix is needed but there is not enough information to do it\n");
302         return -1;
303     }
304
305 av_assert0(s->used_ch_count);
306 av_assert0(s->out.ch_count);
307     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
308
309     s->in_buffer= s->in;
310
311     if(!s->resample && !s->rematrix && !s->channel_map && !s->dither_method){
312         s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
313                                                    s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
314         return 0;
315     }
316
317     s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
318                                              s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
319     s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
320                                              s->int_sample_fmt, s->out.ch_count, NULL, 0);
321
322
323     s->postin= s->in;
324     s->preout= s->out;
325     s->midbuf= s->in;
326
327     if(s->channel_map){
328         s->postin.ch_count=
329         s->midbuf.ch_count= s->used_ch_count;
330         if(s->resample)
331             s->in_buffer.ch_count= s->used_ch_count;
332     }
333     if(!s->resample_first){
334         s->midbuf.ch_count= s->out.ch_count;
335         if(s->resample)
336             s->in_buffer.ch_count = s->out.ch_count;
337     }
338
339     set_audiodata_fmt(&s->postin, s->int_sample_fmt);
340     set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
341     set_audiodata_fmt(&s->preout, s->int_sample_fmt);
342
343     if(s->resample){
344         set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
345     }
346
347     s->dither = s->preout;
348
349     if(s->rematrix || s->dither_method)
350         return swri_rematrix_init(s);
351
352     return 0;
353 }
354
355 static int realloc_audio(AudioData *a, int count){
356     int i, countb;
357     AudioData old;
358
359     if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
360         return AVERROR(EINVAL);
361
362     if(a->count >= count)
363         return 0;
364
365     count*=2;
366
367     countb= FFALIGN(count*a->bps, ALIGN);
368     old= *a;
369
370     av_assert0(a->bps);
371     av_assert0(a->ch_count);
372
373     a->data= av_mallocz(countb*a->ch_count);
374     if(!a->data)
375         return AVERROR(ENOMEM);
376     for(i=0; i<a->ch_count; i++){
377         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
378         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
379     }
380     if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
381     av_free(old.data);
382     a->count= count;
383
384     return 1;
385 }
386
387 static void copy(AudioData *out, AudioData *in,
388                  int count){
389     av_assert0(out->planar == in->planar);
390     av_assert0(out->bps == in->bps);
391     av_assert0(out->ch_count == in->ch_count);
392     if(out->planar){
393         int ch;
394         for(ch=0; ch<out->ch_count; ch++)
395             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
396     }else
397         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
398 }
399
400 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
401     int i;
402     if(!in_arg){
403         memset(out->ch, 0, sizeof(out->ch));
404     }else if(out->planar){
405         for(i=0; i<out->ch_count; i++)
406             out->ch[i]= in_arg[i];
407     }else{
408         for(i=0; i<out->ch_count; i++)
409             out->ch[i]= in_arg[0] + i*out->bps;
410     }
411 }
412
413 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
414     int i;
415     if(out->planar){
416         for(i=0; i<out->ch_count; i++)
417             in_arg[i]= out->ch[i];
418     }else{
419         in_arg[0]= out->ch[0];
420     }
421 }
422
423 /**
424  *
425  * out may be equal in.
426  */
427 static void buf_set(AudioData *out, AudioData *in, int count){
428     int ch;
429     if(in->planar){
430         for(ch=0; ch<out->ch_count; ch++)
431             out->ch[ch]= in->ch[ch] + count*out->bps;
432     }else{
433         for(ch=out->ch_count-1; ch>=0; ch--)
434             out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
435     }
436 }
437
438 /**
439  *
440  * @return number of samples output per channel
441  */
442 static int resample(SwrContext *s, AudioData *out_param, int out_count,
443                              const AudioData * in_param, int in_count){
444     AudioData in, out, tmp;
445     int ret_sum=0;
446     int border=0;
447
448     av_assert1(s->in_buffer.ch_count == in_param->ch_count);
449     av_assert1(s->in_buffer.planar   == in_param->planar);
450     av_assert1(s->in_buffer.fmt      == in_param->fmt);
451
452     tmp=out=*out_param;
453     in =  *in_param;
454
455     do{
456         int ret, size, consumed;
457         if(!s->resample_in_constraint && s->in_buffer_count){
458             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
459             ret= swri_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
460             out_count -= ret;
461             ret_sum += ret;
462             buf_set(&out, &out, ret);
463             s->in_buffer_count -= consumed;
464             s->in_buffer_index += consumed;
465
466             if(!in_count)
467                 break;
468             if(s->in_buffer_count <= border){
469                 buf_set(&in, &in, -s->in_buffer_count);
470                 in_count += s->in_buffer_count;
471                 s->in_buffer_count=0;
472                 s->in_buffer_index=0;
473                 border = 0;
474             }
475         }
476
477         if(in_count && !s->in_buffer_count){
478             s->in_buffer_index=0;
479             ret= swri_multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
480             out_count -= ret;
481             ret_sum += ret;
482             buf_set(&out, &out, ret);
483             in_count -= consumed;
484             buf_set(&in, &in, consumed);
485         }
486
487         //TODO is this check sane considering the advanced copy avoidance below
488         size= s->in_buffer_index + s->in_buffer_count + in_count;
489         if(   size > s->in_buffer.count
490            && s->in_buffer_count + in_count <= s->in_buffer_index){
491             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
492             copy(&s->in_buffer, &tmp, s->in_buffer_count);
493             s->in_buffer_index=0;
494         }else
495             if((ret=realloc_audio(&s->in_buffer, size)) < 0)
496                 return ret;
497
498         if(in_count){
499             int count= in_count;
500             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
501
502             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
503             copy(&tmp, &in, /*in_*/count);
504             s->in_buffer_count += count;
505             in_count -= count;
506             border += count;
507             buf_set(&in, &in, count);
508             s->resample_in_constraint= 0;
509             if(s->in_buffer_count != count || in_count)
510                 continue;
511         }
512         break;
513     }while(1);
514
515     s->resample_in_constraint= !!out_count;
516
517     return ret_sum;
518 }
519
520 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
521                                                       AudioData *in , int  in_count){
522     AudioData *postin, *midbuf, *preout;
523     int ret/*, in_max*/;
524     AudioData preout_tmp, midbuf_tmp;
525
526     if(s->full_convert){
527         av_assert0(!s->resample);
528         swri_audio_convert(s->full_convert, out, in, in_count);
529         return out_count;
530     }
531
532 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
533 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
534
535     if((ret=realloc_audio(&s->postin, in_count))<0)
536         return ret;
537     if(s->resample_first){
538         av_assert0(s->midbuf.ch_count == s->used_ch_count);
539         if((ret=realloc_audio(&s->midbuf, out_count))<0)
540             return ret;
541     }else{
542         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
543         if((ret=realloc_audio(&s->midbuf,  in_count))<0)
544             return ret;
545     }
546     if((ret=realloc_audio(&s->preout, out_count))<0)
547         return ret;
548
549     postin= &s->postin;
550
551     midbuf_tmp= s->midbuf;
552     midbuf= &midbuf_tmp;
553     preout_tmp= s->preout;
554     preout= &preout_tmp;
555
556     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
557         postin= in;
558
559     if(s->resample_first ? !s->resample : !s->rematrix)
560         midbuf= postin;
561
562     if(s->resample_first ? !s->rematrix : !s->resample)
563         preout= midbuf;
564
565     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
566         if(preout==in){
567             out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
568             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
569             copy(out, in, out_count);
570             return out_count;
571         }
572         else if(preout==postin) preout= midbuf= postin= out;
573         else if(preout==midbuf) preout= midbuf= out;
574         else                    preout= out;
575     }
576
577     if(in != postin){
578         swri_audio_convert(s->in_convert, postin, in, in_count);
579     }
580
581     if(s->resample_first){
582         if(postin != midbuf)
583             out_count= resample(s, midbuf, out_count, postin, in_count);
584         if(midbuf != preout)
585             swri_rematrix(s, preout, midbuf, out_count, preout==out);
586     }else{
587         if(postin != midbuf)
588             swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
589         if(midbuf != preout)
590             out_count= resample(s, preout, out_count, midbuf, in_count);
591     }
592
593     if(preout != out && out_count){
594         if(s->dither_method){
595             int ch;
596             int dither_count= FFMAX(out_count, 1<<16);
597             av_assert0(preout != in);
598
599             if((ret=realloc_audio(&s->dither, dither_count))<0)
600                 return ret;
601             if(ret)
602                 for(ch=0; ch<s->dither.ch_count; ch++)
603                     swri_get_dither(s, s->dither.ch[ch], s->dither.count, 12345678913579<<ch, s->out_sample_fmt, s->int_sample_fmt);
604             av_assert0(s->dither.ch_count == preout->ch_count);
605
606             if(s->dither_pos + out_count > s->dither.count)
607                 s->dither_pos = 0;
608
609             for(ch=0; ch<preout->ch_count; ch++)
610                 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);
611
612             s->dither_pos += out_count;
613         }
614 //FIXME packed doesnt need more than 1 chan here!
615         swri_audio_convert(s->out_convert, out, preout, out_count);
616     }
617     return out_count;
618 }
619
620 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
621                                 const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
622     AudioData * in= &s->in;
623     AudioData *out= &s->out;
624
625     if(s->drop_output > 0){
626         int ret;
627         AudioData tmp = s->out;
628         uint8_t *tmp_arg[SWR_CH_MAX];
629         tmp.count = 0;
630         tmp.data  = NULL;
631         if((ret=realloc_audio(&tmp, s->drop_output))<0)
632             return ret;
633
634         reversefill_audiodata(&tmp, tmp_arg);
635         s->drop_output *= -1; //FIXME find a less hackish solution
636         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
637         s->drop_output *= -1;
638         if(ret>0)
639             s->drop_output -= ret;
640
641         av_freep(&tmp.data);
642         if(s->drop_output || !out_arg)
643             return 0;
644         in_count = 0;
645     }
646
647     if(!in_arg){
648         if(s->in_buffer_count){
649             if (s->resample && !s->flushed) {
650                 AudioData *a= &s->in_buffer;
651                 int i, j, ret;
652                 if((ret=realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
653                     return ret;
654                 av_assert0(a->planar);
655                 for(i=0; i<a->ch_count; i++){
656                     for(j=0; j<s->in_buffer_count; j++){
657                         memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j  )*a->bps,
658                             a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
659                     }
660                 }
661                 s->in_buffer_count += (s->in_buffer_count+1)/2;
662                 s->resample_in_constraint = 0;
663                 s->flushed = 1;
664             }
665         }else{
666             return 0;
667         }
668     }else
669         fill_audiodata(in ,  (void*)in_arg);
670
671     fill_audiodata(out, out_arg);
672
673     if(s->resample){
674         int ret = swr_convert_internal(s, out, out_count, in, in_count);
675         if(ret>0 && !s->drop_output)
676             s->outpts += ret * (int64_t)s->in_sample_rate;
677         return ret;
678     }else{
679         AudioData tmp= *in;
680         int ret2=0;
681         int ret, size;
682         size = FFMIN(out_count, s->in_buffer_count);
683         if(size){
684             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
685             ret= swr_convert_internal(s, out, size, &tmp, size);
686             if(ret<0)
687                 return ret;
688             ret2= ret;
689             s->in_buffer_count -= ret;
690             s->in_buffer_index += ret;
691             buf_set(out, out, ret);
692             out_count -= ret;
693             if(!s->in_buffer_count)
694                 s->in_buffer_index = 0;
695         }
696
697         if(in_count){
698             size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
699
700             if(in_count > out_count) { //FIXME move after swr_convert_internal
701                 if(   size > s->in_buffer.count
702                 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
703                     buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
704                     copy(&s->in_buffer, &tmp, s->in_buffer_count);
705                     s->in_buffer_index=0;
706                 }else
707                     if((ret=realloc_audio(&s->in_buffer, size)) < 0)
708                         return ret;
709             }
710
711             if(out_count){
712                 size = FFMIN(in_count, out_count);
713                 ret= swr_convert_internal(s, out, size, in, size);
714                 if(ret<0)
715                     return ret;
716                 buf_set(in, in, ret);
717                 in_count -= ret;
718                 ret2 += ret;
719             }
720             if(in_count){
721                 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
722                 copy(&tmp, in, in_count);
723                 s->in_buffer_count += in_count;
724             }
725         }
726         if(ret2>0 && !s->drop_output)
727             s->outpts += ret2 * (int64_t)s->in_sample_rate;
728         return ret2;
729     }
730 }
731
732 int swr_drop_output(struct SwrContext *s, int count){
733     s->drop_output += count;
734
735     if(s->drop_output <= 0)
736         return 0;
737
738     av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
739     return swr_convert(s, NULL, s->drop_output, NULL, 0);
740 }
741
742 int swr_inject_silence(struct SwrContext *s, int count){
743     int ret, i;
744     AudioData silence = s->in;
745     uint8_t *tmp_arg[SWR_CH_MAX];
746
747     if(count <= 0)
748         return 0;
749
750     silence.count = 0;
751     silence.data  = NULL;
752     if((ret=realloc_audio(&silence, count))<0)
753         return ret;
754
755     if(silence.planar) for(i=0; i<silence.ch_count; i++) {
756         memset(silence.ch[i], silence.bps==1 ? 0x80 : 0, count*silence.bps);
757     } else
758         memset(silence.ch[0], silence.bps==1 ? 0x80 : 0, count*silence.bps*silence.ch_count);
759
760     reversefill_audiodata(&silence, tmp_arg);
761     av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
762     ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
763     av_freep(&silence.data);
764     return ret;
765 }
766
767 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
768     if(pts == INT64_MIN)
769         return s->outpts;
770     if(s->min_compensation >= FLT_MAX) {
771         return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
772     } else {
773         int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts;
774         double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
775
776         if(fabs(fdelta) > s->min_compensation) {
777             if(!s->outpts || fabs(fdelta) > s->min_hard_compensation){
778                 int ret;
779                 if(delta > 0) ret = swr_inject_silence(s,  delta / s->out_sample_rate);
780                 else          ret = swr_drop_output   (s, -delta / s-> in_sample_rate);
781                 if(ret<0){
782                     av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
783                 }
784             } else if(s->soft_compensation_duration && s->max_soft_compensation) {
785                 int duration = s->out_sample_rate * s->soft_compensation_duration;
786                 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
787                 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
788                 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
789                 swr_set_compensation(s, comp, duration);
790             }
791         }
792
793         return s->outpts;
794     }
795 }