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