55ba7e672a751e3f28607fd2f68f79b7cf49009a
[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 };
107
108 unsigned swresample_version(void)
109 {
110     av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
111     return LIBSWRESAMPLE_VERSION_INT;
112 }
113
114 const char *swresample_configuration(void)
115 {
116     return FFMPEG_CONFIGURATION;
117 }
118
119 const char *swresample_license(void)
120 {
121 #define LICENSE_PREFIX "libswresample license: "
122     return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
123 }
124
125 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
126     if(!s || s->in_convert) // s needs to be allocated but not initialized
127         return AVERROR(EINVAL);
128     s->channel_map = channel_map;
129     return 0;
130 }
131
132 const AVClass *swr_get_class(void)
133 {
134     return &av_class;
135 }
136
137 struct SwrContext *swr_alloc(void){
138     SwrContext *s= av_mallocz(sizeof(SwrContext));
139     if(s){
140         s->av_class= &av_class;
141         av_opt_set_defaults(s);
142     }
143     return s;
144 }
145
146 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
147                                       int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
148                                       int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
149                                       int log_offset, void *log_ctx){
150     if(!s) s= swr_alloc();
151     if(!s) return NULL;
152
153     s->log_level_offset= log_offset;
154     s->log_ctx= log_ctx;
155
156     av_opt_set_int(s, "ocl", out_ch_layout,   0);
157     av_opt_set_int(s, "osf", out_sample_fmt,  0);
158     av_opt_set_int(s, "osr", out_sample_rate, 0);
159     av_opt_set_int(s, "icl", in_ch_layout,    0);
160     av_opt_set_int(s, "isf", in_sample_fmt,   0);
161     av_opt_set_int(s, "isr", in_sample_rate,  0);
162     av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE,   0);
163     av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);
164     av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);
165     av_opt_set_int(s, "uch", 0, 0);
166     return s;
167 }
168
169 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
170     a->fmt   = fmt;
171     a->bps   = av_get_bytes_per_sample(fmt);
172     a->planar= av_sample_fmt_is_planar(fmt);
173 }
174
175 static void free_temp(AudioData *a){
176     av_free(a->data);
177     memset(a, 0, sizeof(*a));
178 }
179
180 void swr_free(SwrContext **ss){
181     SwrContext *s= *ss;
182     if(s){
183         free_temp(&s->postin);
184         free_temp(&s->midbuf);
185         free_temp(&s->preout);
186         free_temp(&s->in_buffer);
187         free_temp(&s->dither);
188         swri_audio_convert_free(&s-> in_convert);
189         swri_audio_convert_free(&s->out_convert);
190         swri_audio_convert_free(&s->full_convert);
191         swri_resample_free(&s->resample);
192         swri_rematrix_free(s);
193     }
194
195     av_freep(ss);
196 }
197
198 int swr_init(struct SwrContext *s){
199     s->in_buffer_index= 0;
200     s->in_buffer_count= 0;
201     s->resample_in_constraint= 0;
202     free_temp(&s->postin);
203     free_temp(&s->midbuf);
204     free_temp(&s->preout);
205     free_temp(&s->in_buffer);
206     free_temp(&s->dither);
207     swri_audio_convert_free(&s-> in_convert);
208     swri_audio_convert_free(&s->out_convert);
209     swri_audio_convert_free(&s->full_convert);
210     swri_rematrix_free(s);
211
212     s->flushed = 0;
213
214     if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
215         av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
216         return AVERROR(EINVAL);
217     }
218     if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
219         av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
220         return AVERROR(EINVAL);
221     }
222
223     if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
224         if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
225             s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
226         }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
227             s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
228         }else{
229             av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
230             s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
231         }
232     }
233
234     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P
235         &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
236         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
237         &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
238         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));
239         return AVERROR(EINVAL);
240     }
241
242     set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
243     set_audiodata_fmt(&s->out, s->out_sample_fmt);
244
245     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
246         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);
247     }else
248         swri_resample_free(&s->resample);
249     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P
250         && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
251         && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
252         && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
253         && s->resample){
254         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
255         return -1;
256     }
257
258     if(!s->used_ch_count)
259         s->used_ch_count= s->in.ch_count;
260
261     if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
262         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
263         s-> in_ch_layout= 0;
264     }
265
266     if(!s-> in_ch_layout)
267         s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
268     if(!s->out_ch_layout)
269         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
270
271     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
272                  s->rematrix_custom;
273
274 #define RSC 1 //FIXME finetune
275     if(!s-> in.ch_count)
276         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
277     if(!s->used_ch_count)
278         s->used_ch_count= s->in.ch_count;
279     if(!s->out.ch_count)
280         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
281
282     if(!s-> in.ch_count){
283         av_assert0(!s->in_ch_layout);
284         av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
285         return -1;
286     }
287
288     if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
289         av_log(s, AV_LOG_ERROR, "Rematrix is needed but there is not enough information to do it\n");
290         return -1;
291     }
292
293 av_assert0(s->used_ch_count);
294 av_assert0(s->out.ch_count);
295     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
296
297     s->in_buffer= s->in;
298
299     if(!s->resample && !s->rematrix && !s->channel_map && !s->dither_method){
300         s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
301                                                    s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
302         return 0;
303     }
304
305     s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
306                                              s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
307     s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
308                                              s->int_sample_fmt, s->out.ch_count, NULL, 0);
309
310
311     s->postin= s->in;
312     s->preout= s->out;
313     s->midbuf= s->in;
314
315     if(s->channel_map){
316         s->postin.ch_count=
317         s->midbuf.ch_count= s->used_ch_count;
318         if(s->resample)
319             s->in_buffer.ch_count= s->used_ch_count;
320     }
321     if(!s->resample_first){
322         s->midbuf.ch_count= s->out.ch_count;
323         if(s->resample)
324             s->in_buffer.ch_count = s->out.ch_count;
325     }
326
327     set_audiodata_fmt(&s->postin, s->int_sample_fmt);
328     set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
329     set_audiodata_fmt(&s->preout, s->int_sample_fmt);
330
331     if(s->resample){
332         set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
333     }
334
335     s->dither = s->preout;
336
337     if(s->rematrix || s->dither_method)
338         return swri_rematrix_init(s);
339
340     return 0;
341 }
342
343 static int realloc_audio(AudioData *a, int count){
344     int i, countb;
345     AudioData old;
346
347     if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
348         return AVERROR(EINVAL);
349
350     if(a->count >= count)
351         return 0;
352
353     count*=2;
354
355     countb= FFALIGN(count*a->bps, ALIGN);
356     old= *a;
357
358     av_assert0(a->bps);
359     av_assert0(a->ch_count);
360
361     a->data= av_mallocz(countb*a->ch_count);
362     if(!a->data)
363         return AVERROR(ENOMEM);
364     for(i=0; i<a->ch_count; i++){
365         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
366         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
367     }
368     if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
369     av_free(old.data);
370     a->count= count;
371
372     return 1;
373 }
374
375 static void copy(AudioData *out, AudioData *in,
376                  int count){
377     av_assert0(out->planar == in->planar);
378     av_assert0(out->bps == in->bps);
379     av_assert0(out->ch_count == in->ch_count);
380     if(out->planar){
381         int ch;
382         for(ch=0; ch<out->ch_count; ch++)
383             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
384     }else
385         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
386 }
387
388 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
389     int i;
390     if(!in_arg){
391         memset(out->ch, 0, sizeof(out->ch));
392     }else if(out->planar){
393         for(i=0; i<out->ch_count; i++)
394             out->ch[i]= in_arg[i];
395     }else{
396         for(i=0; i<out->ch_count; i++)
397             out->ch[i]= in_arg[0] + i*out->bps;
398     }
399 }
400
401 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
402     int i;
403     if(out->planar){
404         for(i=0; i<out->ch_count; i++)
405             in_arg[i]= out->ch[i];
406     }else{
407         in_arg[0]= out->ch[0];
408     }
409 }
410
411 /**
412  *
413  * out may be equal in.
414  */
415 static void buf_set(AudioData *out, AudioData *in, int count){
416     int ch;
417     if(in->planar){
418         for(ch=0; ch<out->ch_count; ch++)
419             out->ch[ch]= in->ch[ch] + count*out->bps;
420     }else{
421         for(ch=out->ch_count-1; ch>=0; ch--)
422             out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
423     }
424 }
425
426 /**
427  *
428  * @return number of samples output per channel
429  */
430 static int resample(SwrContext *s, AudioData *out_param, int out_count,
431                              const AudioData * in_param, int in_count){
432     AudioData in, out, tmp;
433     int ret_sum=0;
434     int border=0;
435
436     av_assert1(s->in_buffer.ch_count == in_param->ch_count);
437     av_assert1(s->in_buffer.planar   == in_param->planar);
438     av_assert1(s->in_buffer.fmt      == in_param->fmt);
439
440     tmp=out=*out_param;
441     in =  *in_param;
442
443     do{
444         int ret, size, consumed;
445         if(!s->resample_in_constraint && s->in_buffer_count){
446             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
447             ret= swri_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
448             out_count -= ret;
449             ret_sum += ret;
450             buf_set(&out, &out, ret);
451             s->in_buffer_count -= consumed;
452             s->in_buffer_index += consumed;
453
454             if(!in_count)
455                 break;
456             if(s->in_buffer_count <= border){
457                 buf_set(&in, &in, -s->in_buffer_count);
458                 in_count += s->in_buffer_count;
459                 s->in_buffer_count=0;
460                 s->in_buffer_index=0;
461                 border = 0;
462             }
463         }
464
465         if(in_count && !s->in_buffer_count){
466             s->in_buffer_index=0;
467             ret= swri_multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
468             out_count -= ret;
469             ret_sum += ret;
470             buf_set(&out, &out, ret);
471             in_count -= consumed;
472             buf_set(&in, &in, consumed);
473         }
474
475         //TODO is this check sane considering the advanced copy avoidance below
476         size= s->in_buffer_index + s->in_buffer_count + in_count;
477         if(   size > s->in_buffer.count
478            && s->in_buffer_count + in_count <= s->in_buffer_index){
479             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
480             copy(&s->in_buffer, &tmp, s->in_buffer_count);
481             s->in_buffer_index=0;
482         }else
483             if((ret=realloc_audio(&s->in_buffer, size)) < 0)
484                 return ret;
485
486         if(in_count){
487             int count= in_count;
488             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
489
490             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
491             copy(&tmp, &in, /*in_*/count);
492             s->in_buffer_count += count;
493             in_count -= count;
494             border += count;
495             buf_set(&in, &in, count);
496             s->resample_in_constraint= 0;
497             if(s->in_buffer_count != count || in_count)
498                 continue;
499         }
500         break;
501     }while(1);
502
503     s->resample_in_constraint= !!out_count;
504
505     return ret_sum;
506 }
507
508 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
509                                                       AudioData *in , int  in_count){
510     AudioData *postin, *midbuf, *preout;
511     int ret/*, in_max*/;
512     AudioData preout_tmp, midbuf_tmp;
513
514     if(s->full_convert){
515         av_assert0(!s->resample);
516         swri_audio_convert(s->full_convert, out, in, in_count);
517         return out_count;
518     }
519
520 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
521 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
522
523     if((ret=realloc_audio(&s->postin, in_count))<0)
524         return ret;
525     if(s->resample_first){
526         av_assert0(s->midbuf.ch_count == s->used_ch_count);
527         if((ret=realloc_audio(&s->midbuf, out_count))<0)
528             return ret;
529     }else{
530         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
531         if((ret=realloc_audio(&s->midbuf,  in_count))<0)
532             return ret;
533     }
534     if((ret=realloc_audio(&s->preout, out_count))<0)
535         return ret;
536
537     postin= &s->postin;
538
539     midbuf_tmp= s->midbuf;
540     midbuf= &midbuf_tmp;
541     preout_tmp= s->preout;
542     preout= &preout_tmp;
543
544     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar)
545         postin= in;
546
547     if(s->resample_first ? !s->resample : !s->rematrix)
548         midbuf= postin;
549
550     if(s->resample_first ? !s->rematrix : !s->resample)
551         preout= midbuf;
552
553     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
554         if(preout==in){
555             out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
556             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
557             copy(out, in, out_count);
558             return out_count;
559         }
560         else if(preout==postin) preout= midbuf= postin= out;
561         else if(preout==midbuf) preout= midbuf= out;
562         else                    preout= out;
563     }
564
565     if(in != postin){
566         swri_audio_convert(s->in_convert, postin, in, in_count);
567     }
568
569     if(s->resample_first){
570         if(postin != midbuf)
571             out_count= resample(s, midbuf, out_count, postin, in_count);
572         if(midbuf != preout)
573             swri_rematrix(s, preout, midbuf, out_count, preout==out);
574     }else{
575         if(postin != midbuf)
576             swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
577         if(midbuf != preout)
578             out_count= resample(s, preout, out_count, midbuf, in_count);
579     }
580
581     if(preout != out && out_count){
582         if(s->dither_method){
583             int ch;
584             int dither_count= FFMAX(out_count, 1<<16);
585             av_assert0(preout != in);
586
587             if((ret=realloc_audio(&s->dither, dither_count))<0)
588                 return ret;
589             if(ret)
590                 for(ch=0; ch<s->dither.ch_count; ch++)
591                     swri_get_dither(s, s->dither.ch[ch], s->dither.count, 12345678913579<<ch, s->out_sample_fmt, s->int_sample_fmt);
592             av_assert0(s->dither.ch_count == preout->ch_count);
593
594             if(s->dither_pos + out_count > s->dither.count)
595                 s->dither_pos = 0;
596
597             for(ch=0; ch<preout->ch_count; ch++)
598                 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);
599
600             s->dither_pos += out_count;
601         }
602 //FIXME packed doesnt need more than 1 chan here!
603         swri_audio_convert(s->out_convert, out, preout, out_count);
604     }
605     return out_count;
606 }
607
608 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
609                                 const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
610     AudioData * in= &s->in;
611     AudioData *out= &s->out;
612
613     if(s->drop_output > 0){
614         int ret;
615         AudioData tmp = s->out;
616         uint8_t *tmp_arg[SWR_CH_MAX];
617         tmp.count = 0;
618         tmp.data  = NULL;
619         if((ret=realloc_audio(&tmp, s->drop_output))<0)
620             return ret;
621
622         reversefill_audiodata(&tmp, tmp_arg);
623         s->drop_output *= -1; //FIXME find a less hackish solution
624         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
625         s->drop_output *= -1;
626         if(ret>0)
627             s->drop_output -= ret;
628
629         av_freep(&tmp.data);
630         if(s->drop_output || !out_arg)
631             return 0;
632         in_count = 0;
633     }
634
635     if(!in_arg){
636         if(s->in_buffer_count){
637             if (s->resample && !s->flushed) {
638                 AudioData *a= &s->in_buffer;
639                 int i, j, ret;
640                 if((ret=realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
641                     return ret;
642                 av_assert0(a->planar);
643                 for(i=0; i<a->ch_count; i++){
644                     for(j=0; j<s->in_buffer_count; j++){
645                         memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j  )*a->bps,
646                             a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
647                     }
648                 }
649                 s->in_buffer_count += (s->in_buffer_count+1)/2;
650                 s->resample_in_constraint = 0;
651                 s->flushed = 1;
652             }
653         }else{
654             return 0;
655         }
656     }else
657         fill_audiodata(in ,  (void*)in_arg);
658
659     fill_audiodata(out, out_arg);
660
661     if(s->resample){
662         int ret = swr_convert_internal(s, out, out_count, in, in_count);
663         if(ret>0 && !s->drop_output)
664             s->outpts += ret * (int64_t)s->in_sample_rate;
665         return ret;
666     }else{
667         AudioData tmp= *in;
668         int ret2=0;
669         int ret, size;
670         size = FFMIN(out_count, s->in_buffer_count);
671         if(size){
672             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
673             ret= swr_convert_internal(s, out, size, &tmp, size);
674             if(ret<0)
675                 return ret;
676             ret2= ret;
677             s->in_buffer_count -= ret;
678             s->in_buffer_index += ret;
679             buf_set(out, out, ret);
680             out_count -= ret;
681             if(!s->in_buffer_count)
682                 s->in_buffer_index = 0;
683         }
684
685         if(in_count){
686             size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
687
688             if(in_count > out_count) { //FIXME move after swr_convert_internal
689                 if(   size > s->in_buffer.count
690                 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
691                     buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
692                     copy(&s->in_buffer, &tmp, s->in_buffer_count);
693                     s->in_buffer_index=0;
694                 }else
695                     if((ret=realloc_audio(&s->in_buffer, size)) < 0)
696                         return ret;
697             }
698
699             if(out_count){
700                 size = FFMIN(in_count, out_count);
701                 ret= swr_convert_internal(s, out, size, in, size);
702                 if(ret<0)
703                     return ret;
704                 buf_set(in, in, ret);
705                 in_count -= ret;
706                 ret2 += ret;
707             }
708             if(in_count){
709                 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
710                 copy(&tmp, in, in_count);
711                 s->in_buffer_count += in_count;
712             }
713         }
714         if(ret2>0 && !s->drop_output)
715             s->outpts += ret2 * (int64_t)s->in_sample_rate;
716         return ret2;
717     }
718 }
719
720 int swr_drop_output(struct SwrContext *s, int count){
721     s->drop_output += count;
722
723     if(s->drop_output <= 0)
724         return 0;
725
726     av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
727     return swr_convert(s, NULL, s->drop_output, NULL, 0);
728 }
729
730 int swr_inject_silence(struct SwrContext *s, int count){
731     int ret, i;
732     AudioData silence = s->out;
733     uint8_t *tmp_arg[SWR_CH_MAX];
734
735     if(count <= 0)
736         return 0;
737
738     silence.count = 0;
739     silence.data  = NULL;
740     if((ret=realloc_audio(&silence, count))<0)
741         return ret;
742
743     if(silence.planar) for(i=0; i<silence.ch_count; i++) {
744         memset(silence.ch[i], silence.bps==1 ? 0x80 : 0, count*silence.bps);
745     } else
746         memset(silence.ch[0], silence.bps==1 ? 0x80 : 0, count*silence.bps*silence.ch_count);
747
748     reversefill_audiodata(&silence, tmp_arg);
749     av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
750     ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
751     av_freep(&silence.data);
752     return ret;
753 }
754
755 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
756     if(pts == INT64_MIN)
757         return s->outpts;
758     if(s->min_compensation >= FLT_MAX) {
759         return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
760     } else {
761         int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts;
762         double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
763
764         if(fabs(fdelta) > s->min_compensation) {
765             if(!s->outpts || fabs(fdelta) > s->min_hard_compensation){
766                 int ret;
767                 if(delta > 0) ret = swr_inject_silence(s,  delta / s->out_sample_rate);
768                 else          ret = swr_drop_output   (s, -delta / s-> in_sample_rate);
769                 if(ret<0){
770                     av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
771                 }
772             } else if(s->soft_compensation_duration && s->max_soft_compensation) {
773                 int duration = s->out_sample_rate * s->soft_compensation_duration;
774                 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
775                 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
776                 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
777                 swr_set_compensation(s, comp, duration);
778             }
779         }
780
781         return s->outpts;
782     }
783 }