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