b0bd697e04ab50ce5dd1d71feae4a24b62fca824
[ffmpeg.git] / libswresample / swresample.c
1 /*
2  * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
3  *
4  * This file is part of libswresample
5  *
6  * libswresample is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * libswresample is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with libswresample; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20
21 #include "libavutil/opt.h"
22 #include "swresample_internal.h"
23 #include "audioconvert.h"
24 #include "libavutil/avassert.h"
25 #include "libavutil/channel_layout.h"
26
27 #include <float.h>
28
29 #define ALIGN 32
30
31 unsigned swresample_version(void)
32 {
33     av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
34     return LIBSWRESAMPLE_VERSION_INT;
35 }
36
37 const char *swresample_configuration(void)
38 {
39     return FFMPEG_CONFIGURATION;
40 }
41
42 const char *swresample_license(void)
43 {
44 #define LICENSE_PREFIX "libswresample license: "
45     return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
46 }
47
48 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
49     if(!s || s->in_convert) // s needs to be allocated but not initialized
50         return AVERROR(EINVAL);
51     s->channel_map = channel_map;
52     return 0;
53 }
54
55 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
56                                       int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
57                                       int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
58                                       int log_offset, void *log_ctx){
59     if(!s) s= swr_alloc();
60     if(!s) return NULL;
61
62     s->log_level_offset= log_offset;
63     s->log_ctx= log_ctx;
64
65     if (av_opt_set_int(s, "ocl", out_ch_layout,   0) < 0)
66         goto fail;
67
68     if (av_opt_set_int(s, "osf", out_sample_fmt,  0) < 0)
69         goto fail;
70
71     if (av_opt_set_int(s, "osr", out_sample_rate, 0) < 0)
72         goto fail;
73
74     if (av_opt_set_int(s, "icl", in_ch_layout,    0) < 0)
75         goto fail;
76
77     if (av_opt_set_int(s, "isf", in_sample_fmt,   0) < 0)
78         goto fail;
79
80     if (av_opt_set_int(s, "isr", in_sample_rate,  0) < 0)
81         goto fail;
82
83     if (av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE,   0) < 0)
84         goto fail;
85
86     if (av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0) < 0)
87         goto fail;
88
89     if (av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0) < 0)
90         goto fail;
91
92     av_opt_set_int(s, "uch", 0, 0);
93     return s;
94 fail:
95     av_log(s, AV_LOG_ERROR, "Failed to set option\n");
96     swr_free(&s);
97     return NULL;
98 }
99
100 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
101     a->fmt   = fmt;
102     a->bps   = av_get_bytes_per_sample(fmt);
103     a->planar= av_sample_fmt_is_planar(fmt);
104     if (a->ch_count == 1)
105         a->planar = 1;
106 }
107
108 static void free_temp(AudioData *a){
109     av_free(a->data);
110     memset(a, 0, sizeof(*a));
111 }
112
113 static void clear_context(SwrContext *s){
114     s->in_buffer_index= 0;
115     s->in_buffer_count= 0;
116     s->resample_in_constraint= 0;
117     memset(s->in.ch, 0, sizeof(s->in.ch));
118     memset(s->out.ch, 0, sizeof(s->out.ch));
119     free_temp(&s->postin);
120     free_temp(&s->midbuf);
121     free_temp(&s->preout);
122     free_temp(&s->in_buffer);
123     free_temp(&s->silence);
124     free_temp(&s->drop_temp);
125     free_temp(&s->dither.noise);
126     free_temp(&s->dither.temp);
127     swri_audio_convert_free(&s-> in_convert);
128     swri_audio_convert_free(&s->out_convert);
129     swri_audio_convert_free(&s->full_convert);
130     swri_rematrix_free(s);
131
132     s->flushed = 0;
133 }
134
135 av_cold void swr_free(SwrContext **ss){
136     SwrContext *s= *ss;
137     if(s){
138         clear_context(s);
139         if (s->resampler)
140             s->resampler->free(&s->resample);
141     }
142
143     av_freep(ss);
144 }
145
146 av_cold void swr_close(SwrContext *s){
147     clear_context(s);
148 }
149
150 av_cold int swr_init(struct SwrContext *s){
151     int ret;
152
153     clear_context(s);
154
155     if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
156         av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
157         return AVERROR(EINVAL);
158     }
159     if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
160         av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
161         return AVERROR(EINVAL);
162     }
163
164     if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) {
165         av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
166         s->in_ch_layout = 0;
167     }
168
169     if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) {
170         av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
171         s->out_ch_layout = 0;
172     }
173
174     switch(s->engine){
175 #if CONFIG_LIBSOXR
176         extern struct Resampler const soxr_resampler;
177         case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break;
178 #endif
179         case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
180         default:
181             av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
182             return AVERROR(EINVAL);
183     }
184
185     if(!s->used_ch_count)
186         s->used_ch_count= s->in.ch_count;
187
188     if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
189         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
190         s-> in_ch_layout= 0;
191     }
192
193     if(!s-> in_ch_layout)
194         s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
195     if(!s->out_ch_layout)
196         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
197
198     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
199                  s->rematrix_custom;
200
201     if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
202         if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
203             s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
204         }else if(   av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
205                  && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P
206                  && !s->rematrix
207                  && s->engine != SWR_ENGINE_SOXR){
208             s->int_sample_fmt= AV_SAMPLE_FMT_S32P;
209         }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
210             s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
211         }else{
212             av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
213             s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
214         }
215     }
216
217     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P
218         &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
219         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
220         &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
221         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));
222         return AVERROR(EINVAL);
223     }
224
225     set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
226     set_audiodata_fmt(&s->out, s->out_sample_fmt);
227
228     if (s->firstpts_in_samples != AV_NOPTS_VALUE) {
229         if (!s->async && s->min_compensation >= FLT_MAX/2)
230             s->async = 1;
231         s->firstpts =
232         s->outpts   = s->firstpts_in_samples * s->out_sample_rate;
233     } else
234         s->firstpts = AV_NOPTS_VALUE;
235
236     if (s->async) {
237         if (s->min_compensation >= FLT_MAX/2)
238             s->min_compensation = 0.001;
239         if (s->async > 1.0001) {
240             s->max_soft_compensation = s->async / (double) s->in_sample_rate;
241         }
242     }
243
244     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
245         s->resample = s->resampler->init(s->resample, s->out_sample_rate, s->in_sample_rate, s->filter_size, s->phase_shift, s->linear_interp, s->cutoff, s->int_sample_fmt, s->filter_type, s->kaiser_beta, s->precision, s->cheby);
246     }else
247         s->resampler->free(&s->resample);
248     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P
249         && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
250         && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
251         && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
252         && s->resample){
253         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
254         return -1;
255     }
256
257 #define RSC 1 //FIXME finetune
258     if(!s-> in.ch_count)
259         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
260     if(!s->used_ch_count)
261         s->used_ch_count= s->in.ch_count;
262     if(!s->out.ch_count)
263         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
264
265     if(!s-> in.ch_count){
266         av_assert0(!s->in_ch_layout);
267         av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
268         return -1;
269     }
270
271     if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
272         char l1[1024], l2[1024];
273         av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
274         av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
275         av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
276                "but there is not enough information to do it\n", l1, l2);
277         return -1;
278     }
279
280 av_assert0(s->used_ch_count);
281 av_assert0(s->out.ch_count);
282     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
283
284     s->in_buffer= s->in;
285     s->silence  = s->in;
286     s->drop_temp= s->out;
287
288     if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
289         s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
290                                                    s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
291         return 0;
292     }
293
294     s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
295                                              s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
296     s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
297                                              s->int_sample_fmt, s->out.ch_count, NULL, 0);
298
299     if (!s->in_convert || !s->out_convert)
300         return AVERROR(ENOMEM);
301
302     s->postin= s->in;
303     s->preout= s->out;
304     s->midbuf= s->in;
305
306     if(s->channel_map){
307         s->postin.ch_count=
308         s->midbuf.ch_count= s->used_ch_count;
309         if(s->resample)
310             s->in_buffer.ch_count= s->used_ch_count;
311     }
312     if(!s->resample_first){
313         s->midbuf.ch_count= s->out.ch_count;
314         if(s->resample)
315             s->in_buffer.ch_count = s->out.ch_count;
316     }
317
318     set_audiodata_fmt(&s->postin, s->int_sample_fmt);
319     set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
320     set_audiodata_fmt(&s->preout, s->int_sample_fmt);
321
322     if(s->resample){
323         set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
324     }
325
326     if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
327         return ret;
328
329     if(s->rematrix || s->dither.method)
330         return swri_rematrix_init(s);
331
332     return 0;
333 }
334
335 int swri_realloc_audio(AudioData *a, int count){
336     int i, countb;
337     AudioData old;
338
339     if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
340         return AVERROR(EINVAL);
341
342     if(a->count >= count)
343         return 0;
344
345     count*=2;
346
347     countb= FFALIGN(count*a->bps, ALIGN);
348     old= *a;
349
350     av_assert0(a->bps);
351     av_assert0(a->ch_count);
352
353     a->data= av_mallocz(countb*a->ch_count);
354     if(!a->data)
355         return AVERROR(ENOMEM);
356     for(i=0; i<a->ch_count; i++){
357         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
358         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
359     }
360     if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
361     av_freep(&old.data);
362     a->count= count;
363
364     return 1;
365 }
366
367 static void copy(AudioData *out, AudioData *in,
368                  int count){
369     av_assert0(out->planar == in->planar);
370     av_assert0(out->bps == in->bps);
371     av_assert0(out->ch_count == in->ch_count);
372     if(out->planar){
373         int ch;
374         for(ch=0; ch<out->ch_count; ch++)
375             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
376     }else
377         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
378 }
379
380 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
381     int i;
382     if(!in_arg){
383         memset(out->ch, 0, sizeof(out->ch));
384     }else if(out->planar){
385         for(i=0; i<out->ch_count; i++)
386             out->ch[i]= in_arg[i];
387     }else{
388         for(i=0; i<out->ch_count; i++)
389             out->ch[i]= in_arg[0] + i*out->bps;
390     }
391 }
392
393 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
394     int i;
395     if(out->planar){
396         for(i=0; i<out->ch_count; i++)
397             in_arg[i]= out->ch[i];
398     }else{
399         in_arg[0]= out->ch[0];
400     }
401 }
402
403 /**
404  *
405  * out may be equal in.
406  */
407 static void buf_set(AudioData *out, AudioData *in, int count){
408     int ch;
409     if(in->planar){
410         for(ch=0; ch<out->ch_count; ch++)
411             out->ch[ch]= in->ch[ch] + count*out->bps;
412     }else{
413         for(ch=out->ch_count-1; ch>=0; ch--)
414             out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
415     }
416 }
417
418 /**
419  *
420  * @return number of samples output per channel
421  */
422 static int resample(SwrContext *s, AudioData *out_param, int out_count,
423                              const AudioData * in_param, int in_count){
424     AudioData in, out, tmp;
425     int ret_sum=0;
426     int border=0;
427     int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
428
429     av_assert1(s->in_buffer.ch_count == in_param->ch_count);
430     av_assert1(s->in_buffer.planar   == in_param->planar);
431     av_assert1(s->in_buffer.fmt      == in_param->fmt);
432
433     tmp=out=*out_param;
434     in =  *in_param;
435
436     border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
437                  &in, in_count, &s->in_buffer_index, &s->in_buffer_count);
438     if (border == INT_MAX) {
439         return 0;
440     } else if (border < 0) {
441         return border;
442     } else if (border) {
443         buf_set(&in, &in, border);
444         in_count -= border;
445         s->resample_in_constraint = 0;
446     }
447
448     do{
449         int ret, size, consumed;
450         if(!s->resample_in_constraint && s->in_buffer_count){
451             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
452             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
453             out_count -= ret;
454             ret_sum += ret;
455             buf_set(&out, &out, ret);
456             s->in_buffer_count -= consumed;
457             s->in_buffer_index += consumed;
458
459             if(!in_count)
460                 break;
461             if(s->in_buffer_count <= border){
462                 buf_set(&in, &in, -s->in_buffer_count);
463                 in_count += s->in_buffer_count;
464                 s->in_buffer_count=0;
465                 s->in_buffer_index=0;
466                 border = 0;
467             }
468         }
469
470         if((s->flushed || in_count > padless) && !s->in_buffer_count){
471             s->in_buffer_index=0;
472             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
473             out_count -= ret;
474             ret_sum += ret;
475             buf_set(&out, &out, ret);
476             in_count -= consumed;
477             buf_set(&in, &in, consumed);
478         }
479
480         //TODO is this check sane considering the advanced copy avoidance below
481         size= s->in_buffer_index + s->in_buffer_count + in_count;
482         if(   size > s->in_buffer.count
483            && s->in_buffer_count + in_count <= s->in_buffer_index){
484             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
485             copy(&s->in_buffer, &tmp, s->in_buffer_count);
486             s->in_buffer_index=0;
487         }else
488             if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
489                 return ret;
490
491         if(in_count){
492             int count= in_count;
493             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
494
495             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
496             copy(&tmp, &in, /*in_*/count);
497             s->in_buffer_count += count;
498             in_count -= count;
499             border += count;
500             buf_set(&in, &in, count);
501             s->resample_in_constraint= 0;
502             if(s->in_buffer_count != count || in_count)
503                 continue;
504             if (padless) {
505                 padless = 0;
506                 continue;
507             }
508         }
509         break;
510     }while(1);
511
512     s->resample_in_constraint= !!out_count;
513
514     return ret_sum;
515 }
516
517 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
518                                                       AudioData *in , int  in_count){
519     AudioData *postin, *midbuf, *preout;
520     int ret/*, in_max*/;
521     AudioData preout_tmp, midbuf_tmp;
522
523     if(s->full_convert){
524         av_assert0(!s->resample);
525         swri_audio_convert(s->full_convert, out, in, in_count);
526         return out_count;
527     }
528
529 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
530 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
531
532     if((ret=swri_realloc_audio(&s->postin, in_count))<0)
533         return ret;
534     if(s->resample_first){
535         av_assert0(s->midbuf.ch_count == s->used_ch_count);
536         if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
537             return ret;
538     }else{
539         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
540         if((ret=swri_realloc_audio(&s->midbuf,  in_count))<0)
541             return ret;
542     }
543     if((ret=swri_realloc_audio(&s->preout, out_count))<0)
544         return ret;
545
546     postin= &s->postin;
547
548     midbuf_tmp= s->midbuf;
549     midbuf= &midbuf_tmp;
550     preout_tmp= s->preout;
551     preout= &preout_tmp;
552
553     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
554         postin= in;
555
556     if(s->resample_first ? !s->resample : !s->rematrix)
557         midbuf= postin;
558
559     if(s->resample_first ? !s->rematrix : !s->resample)
560         preout= midbuf;
561
562     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
563        && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){
564         if(preout==in){
565             out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
566             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
567             copy(out, in, out_count);
568             return out_count;
569         }
570         else if(preout==postin) preout= midbuf= postin= out;
571         else if(preout==midbuf) preout= midbuf= out;
572         else                    preout= out;
573     }
574
575     if(in != postin){
576         swri_audio_convert(s->in_convert, postin, in, in_count);
577     }
578
579     if(s->resample_first){
580         if(postin != midbuf)
581             out_count= resample(s, midbuf, out_count, postin, in_count);
582         if(midbuf != preout)
583             swri_rematrix(s, preout, midbuf, out_count, preout==out);
584     }else{
585         if(postin != midbuf)
586             swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
587         if(midbuf != preout)
588             out_count= resample(s, preout, out_count, midbuf, in_count);
589     }
590
591     if(preout != out && out_count){
592         AudioData *conv_src = preout;
593         if(s->dither.method){
594             int ch;
595             int dither_count= FFMAX(out_count, 1<<16);
596
597             if (preout == in) {
598                 conv_src = &s->dither.temp;
599                 if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
600                     return ret;
601             }
602
603             if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
604                 return ret;
605             if(ret)
606                 for(ch=0; ch<s->dither.noise.ch_count; ch++)
607                     swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);
608             av_assert0(s->dither.noise.ch_count == preout->ch_count);
609
610             if(s->dither.noise_pos + out_count > s->dither.noise.count)
611                 s->dither.noise_pos = 0;
612
613             if (s->dither.method < SWR_DITHER_NS){
614                 if (s->mix_2_1_simd) {
615                     int len1= out_count&~15;
616                     int off = len1 * preout->bps;
617
618                     if(len1)
619                         for(ch=0; ch<preout->ch_count; ch++)
620                             s->mix_2_1_simd(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_simd_one, 0, 0, len1);
621                     if(out_count != len1)
622                         for(ch=0; ch<preout->ch_count; ch++)
623                             s->mix_2_1_f(conv_src->ch[ch] + off, preout->ch[ch] + off, s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos + off + len1, s->native_one, 0, 0, out_count - len1);
624                 } else {
625                     for(ch=0; ch<preout->ch_count; ch++)
626                         s->mix_2_1_f(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_one, 0, 0, out_count);
627                 }
628             } else {
629                 switch(s->int_sample_fmt) {
630                 case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
631                 case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
632                 case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
633                 case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
634                 }
635             }
636             s->dither.noise_pos += out_count;
637         }
638 //FIXME packed doesn't need more than 1 chan here!
639         swri_audio_convert(s->out_convert, out, conv_src, out_count);
640     }
641     return out_count;
642 }
643
644 int swr_is_initialized(struct SwrContext *s) {
645     return !!s->in_buffer.ch_count;
646 }
647
648 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
649                                 const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
650     AudioData * in= &s->in;
651     AudioData *out= &s->out;
652
653     if (!swr_is_initialized(s)) {
654         av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
655         return AVERROR(EINVAL);
656     }
657
658     while(s->drop_output > 0){
659         int ret;
660         uint8_t *tmp_arg[SWR_CH_MAX];
661 #define MAX_DROP_STEP 16384
662         if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)
663             return ret;
664
665         reversefill_audiodata(&s->drop_temp, tmp_arg);
666         s->drop_output *= -1; //FIXME find a less hackish solution
667         ret = swr_convert(s, tmp_arg, FFMIN(-s->drop_output, MAX_DROP_STEP), in_arg, in_count); //FIXME optimize but this is as good as never called so maybe it doesn't matter
668         s->drop_output *= -1;
669         in_count = 0;
670         if(ret>0) {
671             s->drop_output -= ret;
672             if (!s->drop_output && !out_arg)
673                 return 0;
674             continue;
675         }
676
677         av_assert0(s->drop_output);
678         return 0;
679     }
680
681     if(!in_arg){
682         if(s->resample){
683             if (!s->flushed)
684                 s->resampler->flush(s);
685             s->resample_in_constraint = 0;
686             s->flushed = 1;
687         }else if(!s->in_buffer_count){
688             return 0;
689         }
690     }else
691         fill_audiodata(in ,  (void*)in_arg);
692
693     fill_audiodata(out, out_arg);
694
695     if(s->resample){
696         int ret = swr_convert_internal(s, out, out_count, in, in_count);
697         if(ret>0 && !s->drop_output)
698             s->outpts += ret * (int64_t)s->in_sample_rate;
699         return ret;
700     }else{
701         AudioData tmp= *in;
702         int ret2=0;
703         int ret, size;
704         size = FFMIN(out_count, s->in_buffer_count);
705         if(size){
706             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
707             ret= swr_convert_internal(s, out, size, &tmp, size);
708             if(ret<0)
709                 return ret;
710             ret2= ret;
711             s->in_buffer_count -= ret;
712             s->in_buffer_index += ret;
713             buf_set(out, out, ret);
714             out_count -= ret;
715             if(!s->in_buffer_count)
716                 s->in_buffer_index = 0;
717         }
718
719         if(in_count){
720             size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
721
722             if(in_count > out_count) { //FIXME move after swr_convert_internal
723                 if(   size > s->in_buffer.count
724                 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
725                     buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
726                     copy(&s->in_buffer, &tmp, s->in_buffer_count);
727                     s->in_buffer_index=0;
728                 }else
729                     if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
730                         return ret;
731             }
732
733             if(out_count){
734                 size = FFMIN(in_count, out_count);
735                 ret= swr_convert_internal(s, out, size, in, size);
736                 if(ret<0)
737                     return ret;
738                 buf_set(in, in, ret);
739                 in_count -= ret;
740                 ret2 += ret;
741             }
742             if(in_count){
743                 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
744                 copy(&tmp, in, in_count);
745                 s->in_buffer_count += in_count;
746             }
747         }
748         if(ret2>0 && !s->drop_output)
749             s->outpts += ret2 * (int64_t)s->in_sample_rate;
750         return ret2;
751     }
752 }
753
754 int swr_drop_output(struct SwrContext *s, int count){
755     const uint8_t *tmp_arg[SWR_CH_MAX];
756     s->drop_output += count;
757
758     if(s->drop_output <= 0)
759         return 0;
760
761     av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
762     return swr_convert(s, NULL, s->drop_output, tmp_arg, 0);
763 }
764
765 int swr_inject_silence(struct SwrContext *s, int count){
766     int ret, i;
767     uint8_t *tmp_arg[SWR_CH_MAX];
768
769     if(count <= 0)
770         return 0;
771
772 #define MAX_SILENCE_STEP 16384
773     while (count > MAX_SILENCE_STEP) {
774         if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
775             return ret;
776         count -= MAX_SILENCE_STEP;
777     }
778
779     if((ret=swri_realloc_audio(&s->silence, count))<0)
780         return ret;
781
782     if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
783         memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
784     } else
785         memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
786
787     reversefill_audiodata(&s->silence, tmp_arg);
788     av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
789     ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
790     return ret;
791 }
792
793 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
794     if (s->resampler && s->resample){
795         return s->resampler->get_delay(s, base);
796     }else{
797         return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
798     }
799 }
800
801 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
802     int ret;
803
804     if (!s || compensation_distance < 0)
805         return AVERROR(EINVAL);
806     if (!compensation_distance && sample_delta)
807         return AVERROR(EINVAL);
808     if (!s->resample) {
809         s->flags |= SWR_FLAG_RESAMPLE;
810         ret = swr_init(s);
811         if (ret < 0)
812             return ret;
813     }
814     if (!s->resampler->set_compensation){
815         return AVERROR(EINVAL);
816     }else{
817         return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
818     }
819 }
820
821 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
822     if(pts == INT64_MIN)
823         return s->outpts;
824
825     if (s->firstpts == AV_NOPTS_VALUE)
826         s->outpts = s->firstpts = pts;
827
828     if(s->min_compensation >= FLT_MAX) {
829         return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
830     } else {
831         int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts + s->drop_output*(int64_t)s->in_sample_rate;
832         double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
833
834         if(fabs(fdelta) > s->min_compensation) {
835             if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
836                 int ret;
837                 if(delta > 0) ret = swr_inject_silence(s,  delta / s->out_sample_rate);
838                 else          ret = swr_drop_output   (s, -delta / s-> in_sample_rate);
839                 if(ret<0){
840                     av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
841                 }
842             } else if(s->soft_compensation_duration && s->max_soft_compensation) {
843                 int duration = s->out_sample_rate * s->soft_compensation_duration;
844                 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
845                 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
846                 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
847                 swr_set_compensation(s, comp, duration);
848             }
849         }
850
851         return s->outpts;
852     }
853 }