swresample/swresample: Use av_mallocz_array()
[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 #include "libavutil/ffversion.h"
32 const char swr_ffversion[] = "FFmpeg version " FFMPEG_VERSION;
33
34 unsigned swresample_version(void)
35 {
36     av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
37     return LIBSWRESAMPLE_VERSION_INT;
38 }
39
40 const char *swresample_configuration(void)
41 {
42     return FFMPEG_CONFIGURATION;
43 }
44
45 const char *swresample_license(void)
46 {
47 #define LICENSE_PREFIX "libswresample license: "
48     return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
49 }
50
51 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
52     if(!s || s->in_convert) // s needs to be allocated but not initialized
53         return AVERROR(EINVAL);
54     s->channel_map = channel_map;
55     return 0;
56 }
57
58 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
59                                       int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
60                                       int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
61                                       int log_offset, void *log_ctx){
62     if(!s) s= swr_alloc();
63     if(!s) return NULL;
64
65     s->log_level_offset= log_offset;
66     s->log_ctx= log_ctx;
67
68     if (av_opt_set_int(s, "ocl", out_ch_layout,   0) < 0)
69         goto fail;
70
71     if (av_opt_set_int(s, "osf", out_sample_fmt,  0) < 0)
72         goto fail;
73
74     if (av_opt_set_int(s, "osr", out_sample_rate, 0) < 0)
75         goto fail;
76
77     if (av_opt_set_int(s, "icl", in_ch_layout,    0) < 0)
78         goto fail;
79
80     if (av_opt_set_int(s, "isf", in_sample_fmt,   0) < 0)
81         goto fail;
82
83     if (av_opt_set_int(s, "isr", in_sample_rate,  0) < 0)
84         goto fail;
85
86     if (av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE,   0) < 0)
87         goto fail;
88
89     if (av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0) < 0)
90         goto fail;
91
92     if (av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0) < 0)
93         goto fail;
94
95     av_opt_set_int(s, "uch", 0, 0);
96     return s;
97 fail:
98     av_log(s, AV_LOG_ERROR, "Failed to set option\n");
99     swr_free(&s);
100     return NULL;
101 }
102
103 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
104     a->fmt   = fmt;
105     a->bps   = av_get_bytes_per_sample(fmt);
106     a->planar= av_sample_fmt_is_planar(fmt);
107     if (a->ch_count == 1)
108         a->planar = 1;
109 }
110
111 static void free_temp(AudioData *a){
112     av_free(a->data);
113     memset(a, 0, sizeof(*a));
114 }
115
116 static void clear_context(SwrContext *s){
117     s->in_buffer_index= 0;
118     s->in_buffer_count= 0;
119     s->resample_in_constraint= 0;
120     memset(s->in.ch, 0, sizeof(s->in.ch));
121     memset(s->out.ch, 0, sizeof(s->out.ch));
122     free_temp(&s->postin);
123     free_temp(&s->midbuf);
124     free_temp(&s->preout);
125     free_temp(&s->in_buffer);
126     free_temp(&s->silence);
127     free_temp(&s->drop_temp);
128     free_temp(&s->dither.noise);
129     free_temp(&s->dither.temp);
130     swri_audio_convert_free(&s-> in_convert);
131     swri_audio_convert_free(&s->out_convert);
132     swri_audio_convert_free(&s->full_convert);
133     swri_rematrix_free(s);
134
135     s->flushed = 0;
136 }
137
138 av_cold void swr_free(SwrContext **ss){
139     SwrContext *s= *ss;
140     if(s){
141         clear_context(s);
142         if (s->resampler)
143             s->resampler->free(&s->resample);
144     }
145
146     av_freep(ss);
147 }
148
149 av_cold void swr_close(SwrContext *s){
150     clear_context(s);
151 }
152
153 av_cold int swr_init(struct SwrContext *s){
154     int ret;
155
156     clear_context(s);
157
158     if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
159         av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
160         return AVERROR(EINVAL);
161     }
162     if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
163         av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
164         return AVERROR(EINVAL);
165     }
166
167     if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) {
168         av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
169         s->in_ch_layout = 0;
170     }
171
172     if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) {
173         av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
174         s->out_ch_layout = 0;
175     }
176
177     switch(s->engine){
178 #if CONFIG_LIBSOXR
179         case SWR_ENGINE_SOXR: s->resampler = &swri_soxr_resampler; break;
180 #endif
181         case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
182         default:
183             av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
184             return AVERROR(EINVAL);
185     }
186
187     if(!s->used_ch_count)
188         s->used_ch_count= s->in.ch_count;
189
190     if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
191         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
192         s-> in_ch_layout= 0;
193     }
194
195     if(!s-> in_ch_layout)
196         s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
197     if(!s->out_ch_layout)
198         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
199
200     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
201                  s->rematrix_custom;
202
203     if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
204         if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
205             s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
206         }else if(   av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
207                  && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P
208                  && !s->rematrix
209                  && s->engine != SWR_ENGINE_SOXR){
210             s->int_sample_fmt= AV_SAMPLE_FMT_S32P;
211         }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
212             s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
213         }else{
214             av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
215             s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
216         }
217     }
218
219     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P
220         &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
221         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
222         &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
223         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));
224         return AVERROR(EINVAL);
225     }
226
227     set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
228     set_audiodata_fmt(&s->out, s->out_sample_fmt);
229
230     if (s->firstpts_in_samples != AV_NOPTS_VALUE) {
231         if (!s->async && s->min_compensation >= FLT_MAX/2)
232             s->async = 1;
233         s->firstpts =
234         s->outpts   = s->firstpts_in_samples * s->out_sample_rate;
235     } else
236         s->firstpts = AV_NOPTS_VALUE;
237
238     if (s->async) {
239         if (s->min_compensation >= FLT_MAX/2)
240             s->min_compensation = 0.001;
241         if (s->async > 1.0001) {
242             s->max_soft_compensation = s->async / (double) s->in_sample_rate;
243         }
244     }
245
246     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
247         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);
248     }else
249         s->resampler->free(&s->resample);
250     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P
251         && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
252         && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
253         && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
254         && s->resample){
255         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
256         return -1;
257     }
258
259 #define RSC 1 //FIXME finetune
260     if(!s-> in.ch_count)
261         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
262     if(!s->used_ch_count)
263         s->used_ch_count= s->in.ch_count;
264     if(!s->out.ch_count)
265         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
266
267     if(!s-> in.ch_count){
268         av_assert0(!s->in_ch_layout);
269         av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
270         return -1;
271     }
272
273     if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
274         char l1[1024], l2[1024];
275         av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
276         av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
277         av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
278                "but there is not enough information to do it\n", l1, l2);
279         return -1;
280     }
281
282 av_assert0(s->used_ch_count);
283 av_assert0(s->out.ch_count);
284     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
285
286     s->in_buffer= s->in;
287     s->silence  = s->in;
288     s->drop_temp= s->out;
289
290     if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
291         s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
292                                                    s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
293         return 0;
294     }
295
296     s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
297                                              s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
298     s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
299                                              s->int_sample_fmt, s->out.ch_count, NULL, 0);
300
301     if (!s->in_convert || !s->out_convert)
302         return AVERROR(ENOMEM);
303
304     s->postin= s->in;
305     s->preout= s->out;
306     s->midbuf= s->in;
307
308     if(s->channel_map){
309         s->postin.ch_count=
310         s->midbuf.ch_count= s->used_ch_count;
311         if(s->resample)
312             s->in_buffer.ch_count= s->used_ch_count;
313     }
314     if(!s->resample_first){
315         s->midbuf.ch_count= s->out.ch_count;
316         if(s->resample)
317             s->in_buffer.ch_count = s->out.ch_count;
318     }
319
320     set_audiodata_fmt(&s->postin, s->int_sample_fmt);
321     set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
322     set_audiodata_fmt(&s->preout, s->int_sample_fmt);
323
324     if(s->resample){
325         set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
326     }
327
328     if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
329         return ret;
330
331     if(s->rematrix || s->dither.method)
332         return swri_rematrix_init(s);
333
334     return 0;
335 }
336
337 int swri_realloc_audio(AudioData *a, int count){
338     int i, countb;
339     AudioData old;
340
341     if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
342         return AVERROR(EINVAL);
343
344     if(a->count >= count)
345         return 0;
346
347     count*=2;
348
349     countb= FFALIGN(count*a->bps, ALIGN);
350     old= *a;
351
352     av_assert0(a->bps);
353     av_assert0(a->ch_count);
354
355     a->data= av_mallocz_array(countb, a->ch_count);
356     if(!a->data)
357         return AVERROR(ENOMEM);
358     for(i=0; i<a->ch_count; i++){
359         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
360         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
361     }
362     if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
363     av_freep(&old.data);
364     a->count= count;
365
366     return 1;
367 }
368
369 static void copy(AudioData *out, AudioData *in,
370                  int count){
371     av_assert0(out->planar == in->planar);
372     av_assert0(out->bps == in->bps);
373     av_assert0(out->ch_count == in->ch_count);
374     if(out->planar){
375         int ch;
376         for(ch=0; ch<out->ch_count; ch++)
377             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
378     }else
379         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
380 }
381
382 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
383     int i;
384     if(!in_arg){
385         memset(out->ch, 0, sizeof(out->ch));
386     }else if(out->planar){
387         for(i=0; i<out->ch_count; i++)
388             out->ch[i]= in_arg[i];
389     }else{
390         for(i=0; i<out->ch_count; i++)
391             out->ch[i]= in_arg[0] + i*out->bps;
392     }
393 }
394
395 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
396     int i;
397     if(out->planar){
398         for(i=0; i<out->ch_count; i++)
399             in_arg[i]= out->ch[i];
400     }else{
401         in_arg[0]= out->ch[0];
402     }
403 }
404
405 /**
406  *
407  * out may be equal in.
408  */
409 static void buf_set(AudioData *out, AudioData *in, int count){
410     int ch;
411     if(in->planar){
412         for(ch=0; ch<out->ch_count; ch++)
413             out->ch[ch]= in->ch[ch] + count*out->bps;
414     }else{
415         for(ch=out->ch_count-1; ch>=0; ch--)
416             out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
417     }
418 }
419
420 /**
421  *
422  * @return number of samples output per channel
423  */
424 static int resample(SwrContext *s, AudioData *out_param, int out_count,
425                              const AudioData * in_param, int in_count){
426     AudioData in, out, tmp;
427     int ret_sum=0;
428     int border=0;
429     int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
430
431     av_assert1(s->in_buffer.ch_count == in_param->ch_count);
432     av_assert1(s->in_buffer.planar   == in_param->planar);
433     av_assert1(s->in_buffer.fmt      == in_param->fmt);
434
435     tmp=out=*out_param;
436     in =  *in_param;
437
438     border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
439                  &in, in_count, &s->in_buffer_index, &s->in_buffer_count);
440     if (border == INT_MAX) {
441         return 0;
442     } else if (border < 0) {
443         return border;
444     } else if (border) {
445         buf_set(&in, &in, border);
446         in_count -= border;
447         s->resample_in_constraint = 0;
448     }
449
450     do{
451         int ret, size, consumed;
452         if(!s->resample_in_constraint && s->in_buffer_count){
453             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
454             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
455             out_count -= ret;
456             ret_sum += ret;
457             buf_set(&out, &out, ret);
458             s->in_buffer_count -= consumed;
459             s->in_buffer_index += consumed;
460
461             if(!in_count)
462                 break;
463             if(s->in_buffer_count <= border){
464                 buf_set(&in, &in, -s->in_buffer_count);
465                 in_count += s->in_buffer_count;
466                 s->in_buffer_count=0;
467                 s->in_buffer_index=0;
468                 border = 0;
469             }
470         }
471
472         if((s->flushed || in_count > padless) && !s->in_buffer_count){
473             s->in_buffer_index=0;
474             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
475             out_count -= ret;
476             ret_sum += ret;
477             buf_set(&out, &out, ret);
478             in_count -= consumed;
479             buf_set(&in, &in, consumed);
480         }
481
482         //TODO is this check sane considering the advanced copy avoidance below
483         size= s->in_buffer_index + s->in_buffer_count + in_count;
484         if(   size > s->in_buffer.count
485            && s->in_buffer_count + in_count <= s->in_buffer_index){
486             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
487             copy(&s->in_buffer, &tmp, s->in_buffer_count);
488             s->in_buffer_index=0;
489         }else
490             if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
491                 return ret;
492
493         if(in_count){
494             int count= in_count;
495             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
496
497             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
498             copy(&tmp, &in, /*in_*/count);
499             s->in_buffer_count += count;
500             in_count -= count;
501             border += count;
502             buf_set(&in, &in, count);
503             s->resample_in_constraint= 0;
504             if(s->in_buffer_count != count || in_count)
505                 continue;
506             if (padless) {
507                 padless = 0;
508                 continue;
509             }
510         }
511         break;
512     }while(1);
513
514     s->resample_in_constraint= !!out_count;
515
516     return ret_sum;
517 }
518
519 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
520                                                       AudioData *in , int  in_count){
521     AudioData *postin, *midbuf, *preout;
522     int ret/*, in_max*/;
523     AudioData preout_tmp, midbuf_tmp;
524
525     if(s->full_convert){
526         av_assert0(!s->resample);
527         swri_audio_convert(s->full_convert, out, in, in_count);
528         return out_count;
529     }
530
531 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
532 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
533
534     if((ret=swri_realloc_audio(&s->postin, in_count))<0)
535         return ret;
536     if(s->resample_first){
537         av_assert0(s->midbuf.ch_count == s->used_ch_count);
538         if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
539             return ret;
540     }else{
541         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
542         if((ret=swri_realloc_audio(&s->midbuf,  in_count))<0)
543             return ret;
544     }
545     if((ret=swri_realloc_audio(&s->preout, out_count))<0)
546         return ret;
547
548     postin= &s->postin;
549
550     midbuf_tmp= s->midbuf;
551     midbuf= &midbuf_tmp;
552     preout_tmp= s->preout;
553     preout= &preout_tmp;
554
555     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
556         postin= in;
557
558     if(s->resample_first ? !s->resample : !s->rematrix)
559         midbuf= postin;
560
561     if(s->resample_first ? !s->rematrix : !s->resample)
562         preout= midbuf;
563
564     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
565        && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){
566         if(preout==in){
567             out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
568             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
569             copy(out, in, out_count);
570             return out_count;
571         }
572         else if(preout==postin) preout= midbuf= postin= out;
573         else if(preout==midbuf) preout= midbuf= out;
574         else                    preout= out;
575     }
576
577     if(in != postin){
578         swri_audio_convert(s->in_convert, postin, in, in_count);
579     }
580
581     if(s->resample_first){
582         if(postin != midbuf)
583             out_count= resample(s, midbuf, out_count, postin, in_count);
584         if(midbuf != preout)
585             swri_rematrix(s, preout, midbuf, out_count, preout==out);
586     }else{
587         if(postin != midbuf)
588             swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
589         if(midbuf != preout)
590             out_count= resample(s, preout, out_count, midbuf, in_count);
591     }
592
593     if(preout != out && out_count){
594         AudioData *conv_src = preout;
595         if(s->dither.method){
596             int ch;
597             int dither_count= FFMAX(out_count, 1<<16);
598
599             if (preout == in) {
600                 conv_src = &s->dither.temp;
601                 if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
602                     return ret;
603             }
604
605             if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
606                 return ret;
607             if(ret)
608                 for(ch=0; ch<s->dither.noise.ch_count; ch++)
609                     swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);
610             av_assert0(s->dither.noise.ch_count == preout->ch_count);
611
612             if(s->dither.noise_pos + out_count > s->dither.noise.count)
613                 s->dither.noise_pos = 0;
614
615             if (s->dither.method < SWR_DITHER_NS){
616                 if (s->mix_2_1_simd) {
617                     int len1= out_count&~15;
618                     int off = len1 * preout->bps;
619
620                     if(len1)
621                         for(ch=0; ch<preout->ch_count; ch++)
622                             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);
623                     if(out_count != len1)
624                         for(ch=0; ch<preout->ch_count; ch++)
625                             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);
626                 } else {
627                     for(ch=0; ch<preout->ch_count; ch++)
628                         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);
629                 }
630             } else {
631                 switch(s->int_sample_fmt) {
632                 case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
633                 case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
634                 case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
635                 case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
636                 }
637             }
638             s->dither.noise_pos += out_count;
639         }
640 //FIXME packed doesn't need more than 1 chan here!
641         swri_audio_convert(s->out_convert, out, conv_src, out_count);
642     }
643     return out_count;
644 }
645
646 int swr_is_initialized(struct SwrContext *s) {
647     return !!s->in_buffer.ch_count;
648 }
649
650 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
651                                 const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
652     AudioData * in= &s->in;
653     AudioData *out= &s->out;
654
655     if (!swr_is_initialized(s)) {
656         av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
657         return AVERROR(EINVAL);
658     }
659
660     while(s->drop_output > 0){
661         int ret;
662         uint8_t *tmp_arg[SWR_CH_MAX];
663 #define MAX_DROP_STEP 16384
664         if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)
665             return ret;
666
667         reversefill_audiodata(&s->drop_temp, tmp_arg);
668         s->drop_output *= -1; //FIXME find a less hackish solution
669         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
670         s->drop_output *= -1;
671         in_count = 0;
672         if(ret>0) {
673             s->drop_output -= ret;
674             if (!s->drop_output && !out_arg)
675                 return 0;
676             continue;
677         }
678
679         av_assert0(s->drop_output);
680         return 0;
681     }
682
683     if(!in_arg){
684         if(s->resample){
685             if (!s->flushed)
686                 s->resampler->flush(s);
687             s->resample_in_constraint = 0;
688             s->flushed = 1;
689         }else if(!s->in_buffer_count){
690             return 0;
691         }
692     }else
693         fill_audiodata(in ,  (void*)in_arg);
694
695     fill_audiodata(out, out_arg);
696
697     if(s->resample){
698         int ret = swr_convert_internal(s, out, out_count, in, in_count);
699         if(ret>0 && !s->drop_output)
700             s->outpts += ret * (int64_t)s->in_sample_rate;
701         return ret;
702     }else{
703         AudioData tmp= *in;
704         int ret2=0;
705         int ret, size;
706         size = FFMIN(out_count, s->in_buffer_count);
707         if(size){
708             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
709             ret= swr_convert_internal(s, out, size, &tmp, size);
710             if(ret<0)
711                 return ret;
712             ret2= ret;
713             s->in_buffer_count -= ret;
714             s->in_buffer_index += ret;
715             buf_set(out, out, ret);
716             out_count -= ret;
717             if(!s->in_buffer_count)
718                 s->in_buffer_index = 0;
719         }
720
721         if(in_count){
722             size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
723
724             if(in_count > out_count) { //FIXME move after swr_convert_internal
725                 if(   size > s->in_buffer.count
726                 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
727                     buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
728                     copy(&s->in_buffer, &tmp, s->in_buffer_count);
729                     s->in_buffer_index=0;
730                 }else
731                     if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
732                         return ret;
733             }
734
735             if(out_count){
736                 size = FFMIN(in_count, out_count);
737                 ret= swr_convert_internal(s, out, size, in, size);
738                 if(ret<0)
739                     return ret;
740                 buf_set(in, in, ret);
741                 in_count -= ret;
742                 ret2 += ret;
743             }
744             if(in_count){
745                 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
746                 copy(&tmp, in, in_count);
747                 s->in_buffer_count += in_count;
748             }
749         }
750         if(ret2>0 && !s->drop_output)
751             s->outpts += ret2 * (int64_t)s->in_sample_rate;
752         return ret2;
753     }
754 }
755
756 int swr_drop_output(struct SwrContext *s, int count){
757     const uint8_t *tmp_arg[SWR_CH_MAX];
758     s->drop_output += count;
759
760     if(s->drop_output <= 0)
761         return 0;
762
763     av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
764     return swr_convert(s, NULL, s->drop_output, tmp_arg, 0);
765 }
766
767 int swr_inject_silence(struct SwrContext *s, int count){
768     int ret, i;
769     uint8_t *tmp_arg[SWR_CH_MAX];
770
771     if(count <= 0)
772         return 0;
773
774 #define MAX_SILENCE_STEP 16384
775     while (count > MAX_SILENCE_STEP) {
776         if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
777             return ret;
778         count -= MAX_SILENCE_STEP;
779     }
780
781     if((ret=swri_realloc_audio(&s->silence, count))<0)
782         return ret;
783
784     if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
785         memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
786     } else
787         memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
788
789     reversefill_audiodata(&s->silence, tmp_arg);
790     av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
791     ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
792     return ret;
793 }
794
795 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
796     if (s->resampler && s->resample){
797         return s->resampler->get_delay(s, base);
798     }else{
799         return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
800     }
801 }
802
803 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
804     int ret;
805
806     if (!s || compensation_distance < 0)
807         return AVERROR(EINVAL);
808     if (!compensation_distance && sample_delta)
809         return AVERROR(EINVAL);
810     if (!s->resample) {
811         s->flags |= SWR_FLAG_RESAMPLE;
812         ret = swr_init(s);
813         if (ret < 0)
814             return ret;
815     }
816     if (!s->resampler->set_compensation){
817         return AVERROR(EINVAL);
818     }else{
819         return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
820     }
821 }
822
823 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
824     if(pts == INT64_MIN)
825         return s->outpts;
826
827     if (s->firstpts == AV_NOPTS_VALUE)
828         s->outpts = s->firstpts = pts;
829
830     if(s->min_compensation >= FLT_MAX) {
831         return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
832     } else {
833         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;
834         double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
835
836         if(fabs(fdelta) > s->min_compensation) {
837             if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
838                 int ret;
839                 if(delta > 0) ret = swr_inject_silence(s,  delta / s->out_sample_rate);
840                 else          ret = swr_drop_output   (s, -delta / s-> in_sample_rate);
841                 if(ret<0){
842                     av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
843                 }
844             } else if(s->soft_compensation_duration && s->max_soft_compensation) {
845                 int duration = s->out_sample_rate * s->soft_compensation_duration;
846                 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
847                 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
848                 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
849                 swr_set_compensation(s, comp, duration);
850             }
851         }
852
853         return s->outpts;
854     }
855 }