Add FFMPEG_VERSION into the binary libs
[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         extern struct Resampler const soxr_resampler;
180         case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break;
181 #endif
182         case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
183         default:
184             av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
185             return AVERROR(EINVAL);
186     }
187
188     if(!s->used_ch_count)
189         s->used_ch_count= s->in.ch_count;
190
191     if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
192         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
193         s-> in_ch_layout= 0;
194     }
195
196     if(!s-> in_ch_layout)
197         s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
198     if(!s->out_ch_layout)
199         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
200
201     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
202                  s->rematrix_custom;
203
204     if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
205         if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
206             s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
207         }else if(   av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
208                  && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P
209                  && !s->rematrix
210                  && s->engine != SWR_ENGINE_SOXR){
211             s->int_sample_fmt= AV_SAMPLE_FMT_S32P;
212         }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
213             s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
214         }else{
215             av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
216             s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
217         }
218     }
219
220     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P
221         &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
222         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
223         &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
224         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));
225         return AVERROR(EINVAL);
226     }
227
228     set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
229     set_audiodata_fmt(&s->out, s->out_sample_fmt);
230
231     if (s->firstpts_in_samples != AV_NOPTS_VALUE) {
232         if (!s->async && s->min_compensation >= FLT_MAX/2)
233             s->async = 1;
234         s->firstpts =
235         s->outpts   = s->firstpts_in_samples * s->out_sample_rate;
236     } else
237         s->firstpts = AV_NOPTS_VALUE;
238
239     if (s->async) {
240         if (s->min_compensation >= FLT_MAX/2)
241             s->min_compensation = 0.001;
242         if (s->async > 1.0001) {
243             s->max_soft_compensation = s->async / (double) s->in_sample_rate;
244         }
245     }
246
247     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
248         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);
249     }else
250         s->resampler->free(&s->resample);
251     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P
252         && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
253         && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
254         && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
255         && s->resample){
256         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
257         return -1;
258     }
259
260 #define RSC 1 //FIXME finetune
261     if(!s-> in.ch_count)
262         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
263     if(!s->used_ch_count)
264         s->used_ch_count= s->in.ch_count;
265     if(!s->out.ch_count)
266         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
267
268     if(!s-> in.ch_count){
269         av_assert0(!s->in_ch_layout);
270         av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
271         return -1;
272     }
273
274     if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
275         char l1[1024], l2[1024];
276         av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
277         av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
278         av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
279                "but there is not enough information to do it\n", l1, l2);
280         return -1;
281     }
282
283 av_assert0(s->used_ch_count);
284 av_assert0(s->out.ch_count);
285     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
286
287     s->in_buffer= s->in;
288     s->silence  = s->in;
289     s->drop_temp= s->out;
290
291     if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
292         s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
293                                                    s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
294         return 0;
295     }
296
297     s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
298                                              s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
299     s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
300                                              s->int_sample_fmt, s->out.ch_count, NULL, 0);
301
302     if (!s->in_convert || !s->out_convert)
303         return AVERROR(ENOMEM);
304
305     s->postin= s->in;
306     s->preout= s->out;
307     s->midbuf= s->in;
308
309     if(s->channel_map){
310         s->postin.ch_count=
311         s->midbuf.ch_count= s->used_ch_count;
312         if(s->resample)
313             s->in_buffer.ch_count= s->used_ch_count;
314     }
315     if(!s->resample_first){
316         s->midbuf.ch_count= s->out.ch_count;
317         if(s->resample)
318             s->in_buffer.ch_count = s->out.ch_count;
319     }
320
321     set_audiodata_fmt(&s->postin, s->int_sample_fmt);
322     set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
323     set_audiodata_fmt(&s->preout, s->int_sample_fmt);
324
325     if(s->resample){
326         set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
327     }
328
329     if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
330         return ret;
331
332     if(s->rematrix || s->dither.method)
333         return swri_rematrix_init(s);
334
335     return 0;
336 }
337
338 int swri_realloc_audio(AudioData *a, int count){
339     int i, countb;
340     AudioData old;
341
342     if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
343         return AVERROR(EINVAL);
344
345     if(a->count >= count)
346         return 0;
347
348     count*=2;
349
350     countb= FFALIGN(count*a->bps, ALIGN);
351     old= *a;
352
353     av_assert0(a->bps);
354     av_assert0(a->ch_count);
355
356     a->data= av_mallocz(countb*a->ch_count);
357     if(!a->data)
358         return AVERROR(ENOMEM);
359     for(i=0; i<a->ch_count; i++){
360         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
361         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
362     }
363     if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
364     av_freep(&old.data);
365     a->count= count;
366
367     return 1;
368 }
369
370 static void copy(AudioData *out, AudioData *in,
371                  int count){
372     av_assert0(out->planar == in->planar);
373     av_assert0(out->bps == in->bps);
374     av_assert0(out->ch_count == in->ch_count);
375     if(out->planar){
376         int ch;
377         for(ch=0; ch<out->ch_count; ch++)
378             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
379     }else
380         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
381 }
382
383 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
384     int i;
385     if(!in_arg){
386         memset(out->ch, 0, sizeof(out->ch));
387     }else if(out->planar){
388         for(i=0; i<out->ch_count; i++)
389             out->ch[i]= in_arg[i];
390     }else{
391         for(i=0; i<out->ch_count; i++)
392             out->ch[i]= in_arg[0] + i*out->bps;
393     }
394 }
395
396 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
397     int i;
398     if(out->planar){
399         for(i=0; i<out->ch_count; i++)
400             in_arg[i]= out->ch[i];
401     }else{
402         in_arg[0]= out->ch[0];
403     }
404 }
405
406 /**
407  *
408  * out may be equal in.
409  */
410 static void buf_set(AudioData *out, AudioData *in, int count){
411     int ch;
412     if(in->planar){
413         for(ch=0; ch<out->ch_count; ch++)
414             out->ch[ch]= in->ch[ch] + count*out->bps;
415     }else{
416         for(ch=out->ch_count-1; ch>=0; ch--)
417             out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
418     }
419 }
420
421 /**
422  *
423  * @return number of samples output per channel
424  */
425 static int resample(SwrContext *s, AudioData *out_param, int out_count,
426                              const AudioData * in_param, int in_count){
427     AudioData in, out, tmp;
428     int ret_sum=0;
429     int border=0;
430     int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
431
432     av_assert1(s->in_buffer.ch_count == in_param->ch_count);
433     av_assert1(s->in_buffer.planar   == in_param->planar);
434     av_assert1(s->in_buffer.fmt      == in_param->fmt);
435
436     tmp=out=*out_param;
437     in =  *in_param;
438
439     border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
440                  &in, in_count, &s->in_buffer_index, &s->in_buffer_count);
441     if (border == INT_MAX) {
442         return 0;
443     } else if (border < 0) {
444         return border;
445     } else if (border) {
446         buf_set(&in, &in, border);
447         in_count -= border;
448         s->resample_in_constraint = 0;
449     }
450
451     do{
452         int ret, size, consumed;
453         if(!s->resample_in_constraint && s->in_buffer_count){
454             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
455             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
456             out_count -= ret;
457             ret_sum += ret;
458             buf_set(&out, &out, ret);
459             s->in_buffer_count -= consumed;
460             s->in_buffer_index += consumed;
461
462             if(!in_count)
463                 break;
464             if(s->in_buffer_count <= border){
465                 buf_set(&in, &in, -s->in_buffer_count);
466                 in_count += s->in_buffer_count;
467                 s->in_buffer_count=0;
468                 s->in_buffer_index=0;
469                 border = 0;
470             }
471         }
472
473         if((s->flushed || in_count > padless) && !s->in_buffer_count){
474             s->in_buffer_index=0;
475             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
476             out_count -= ret;
477             ret_sum += ret;
478             buf_set(&out, &out, ret);
479             in_count -= consumed;
480             buf_set(&in, &in, consumed);
481         }
482
483         //TODO is this check sane considering the advanced copy avoidance below
484         size= s->in_buffer_index + s->in_buffer_count + in_count;
485         if(   size > s->in_buffer.count
486            && s->in_buffer_count + in_count <= s->in_buffer_index){
487             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
488             copy(&s->in_buffer, &tmp, s->in_buffer_count);
489             s->in_buffer_index=0;
490         }else
491             if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
492                 return ret;
493
494         if(in_count){
495             int count= in_count;
496             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
497
498             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
499             copy(&tmp, &in, /*in_*/count);
500             s->in_buffer_count += count;
501             in_count -= count;
502             border += count;
503             buf_set(&in, &in, count);
504             s->resample_in_constraint= 0;
505             if(s->in_buffer_count != count || in_count)
506                 continue;
507             if (padless) {
508                 padless = 0;
509                 continue;
510             }
511         }
512         break;
513     }while(1);
514
515     s->resample_in_constraint= !!out_count;
516
517     return ret_sum;
518 }
519
520 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
521                                                       AudioData *in , int  in_count){
522     AudioData *postin, *midbuf, *preout;
523     int ret/*, in_max*/;
524     AudioData preout_tmp, midbuf_tmp;
525
526     if(s->full_convert){
527         av_assert0(!s->resample);
528         swri_audio_convert(s->full_convert, out, in, in_count);
529         return out_count;
530     }
531
532 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
533 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
534
535     if((ret=swri_realloc_audio(&s->postin, in_count))<0)
536         return ret;
537     if(s->resample_first){
538         av_assert0(s->midbuf.ch_count == s->used_ch_count);
539         if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
540             return ret;
541     }else{
542         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
543         if((ret=swri_realloc_audio(&s->midbuf,  in_count))<0)
544             return ret;
545     }
546     if((ret=swri_realloc_audio(&s->preout, out_count))<0)
547         return ret;
548
549     postin= &s->postin;
550
551     midbuf_tmp= s->midbuf;
552     midbuf= &midbuf_tmp;
553     preout_tmp= s->preout;
554     preout= &preout_tmp;
555
556     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
557         postin= in;
558
559     if(s->resample_first ? !s->resample : !s->rematrix)
560         midbuf= postin;
561
562     if(s->resample_first ? !s->rematrix : !s->resample)
563         preout= midbuf;
564
565     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
566        && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){
567         if(preout==in){
568             out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
569             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
570             copy(out, in, out_count);
571             return out_count;
572         }
573         else if(preout==postin) preout= midbuf= postin= out;
574         else if(preout==midbuf) preout= midbuf= out;
575         else                    preout= out;
576     }
577
578     if(in != postin){
579         swri_audio_convert(s->in_convert, postin, in, in_count);
580     }
581
582     if(s->resample_first){
583         if(postin != midbuf)
584             out_count= resample(s, midbuf, out_count, postin, in_count);
585         if(midbuf != preout)
586             swri_rematrix(s, preout, midbuf, out_count, preout==out);
587     }else{
588         if(postin != midbuf)
589             swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
590         if(midbuf != preout)
591             out_count= resample(s, preout, out_count, midbuf, in_count);
592     }
593
594     if(preout != out && out_count){
595         AudioData *conv_src = preout;
596         if(s->dither.method){
597             int ch;
598             int dither_count= FFMAX(out_count, 1<<16);
599
600             if (preout == in) {
601                 conv_src = &s->dither.temp;
602                 if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
603                     return ret;
604             }
605
606             if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
607                 return ret;
608             if(ret)
609                 for(ch=0; ch<s->dither.noise.ch_count; ch++)
610                     swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);
611             av_assert0(s->dither.noise.ch_count == preout->ch_count);
612
613             if(s->dither.noise_pos + out_count > s->dither.noise.count)
614                 s->dither.noise_pos = 0;
615
616             if (s->dither.method < SWR_DITHER_NS){
617                 if (s->mix_2_1_simd) {
618                     int len1= out_count&~15;
619                     int off = len1 * preout->bps;
620
621                     if(len1)
622                         for(ch=0; ch<preout->ch_count; ch++)
623                             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);
624                     if(out_count != len1)
625                         for(ch=0; ch<preout->ch_count; ch++)
626                             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);
627                 } else {
628                     for(ch=0; ch<preout->ch_count; ch++)
629                         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);
630                 }
631             } else {
632                 switch(s->int_sample_fmt) {
633                 case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
634                 case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
635                 case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
636                 case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
637                 }
638             }
639             s->dither.noise_pos += out_count;
640         }
641 //FIXME packed doesn't need more than 1 chan here!
642         swri_audio_convert(s->out_convert, out, conv_src, out_count);
643     }
644     return out_count;
645 }
646
647 int swr_is_initialized(struct SwrContext *s) {
648     return !!s->in_buffer.ch_count;
649 }
650
651 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
652                                 const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
653     AudioData * in= &s->in;
654     AudioData *out= &s->out;
655
656     if (!swr_is_initialized(s)) {
657         av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
658         return AVERROR(EINVAL);
659     }
660
661     while(s->drop_output > 0){
662         int ret;
663         uint8_t *tmp_arg[SWR_CH_MAX];
664 #define MAX_DROP_STEP 16384
665         if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)
666             return ret;
667
668         reversefill_audiodata(&s->drop_temp, tmp_arg);
669         s->drop_output *= -1; //FIXME find a less hackish solution
670         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
671         s->drop_output *= -1;
672         in_count = 0;
673         if(ret>0) {
674             s->drop_output -= ret;
675             if (!s->drop_output && !out_arg)
676                 return 0;
677             continue;
678         }
679
680         av_assert0(s->drop_output);
681         return 0;
682     }
683
684     if(!in_arg){
685         if(s->resample){
686             if (!s->flushed)
687                 s->resampler->flush(s);
688             s->resample_in_constraint = 0;
689             s->flushed = 1;
690         }else if(!s->in_buffer_count){
691             return 0;
692         }
693     }else
694         fill_audiodata(in ,  (void*)in_arg);
695
696     fill_audiodata(out, out_arg);
697
698     if(s->resample){
699         int ret = swr_convert_internal(s, out, out_count, in, in_count);
700         if(ret>0 && !s->drop_output)
701             s->outpts += ret * (int64_t)s->in_sample_rate;
702         return ret;
703     }else{
704         AudioData tmp= *in;
705         int ret2=0;
706         int ret, size;
707         size = FFMIN(out_count, s->in_buffer_count);
708         if(size){
709             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
710             ret= swr_convert_internal(s, out, size, &tmp, size);
711             if(ret<0)
712                 return ret;
713             ret2= ret;
714             s->in_buffer_count -= ret;
715             s->in_buffer_index += ret;
716             buf_set(out, out, ret);
717             out_count -= ret;
718             if(!s->in_buffer_count)
719                 s->in_buffer_index = 0;
720         }
721
722         if(in_count){
723             size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
724
725             if(in_count > out_count) { //FIXME move after swr_convert_internal
726                 if(   size > s->in_buffer.count
727                 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
728                     buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
729                     copy(&s->in_buffer, &tmp, s->in_buffer_count);
730                     s->in_buffer_index=0;
731                 }else
732                     if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
733                         return ret;
734             }
735
736             if(out_count){
737                 size = FFMIN(in_count, out_count);
738                 ret= swr_convert_internal(s, out, size, in, size);
739                 if(ret<0)
740                     return ret;
741                 buf_set(in, in, ret);
742                 in_count -= ret;
743                 ret2 += ret;
744             }
745             if(in_count){
746                 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
747                 copy(&tmp, in, in_count);
748                 s->in_buffer_count += in_count;
749             }
750         }
751         if(ret2>0 && !s->drop_output)
752             s->outpts += ret2 * (int64_t)s->in_sample_rate;
753         return ret2;
754     }
755 }
756
757 int swr_drop_output(struct SwrContext *s, int count){
758     const uint8_t *tmp_arg[SWR_CH_MAX];
759     s->drop_output += count;
760
761     if(s->drop_output <= 0)
762         return 0;
763
764     av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
765     return swr_convert(s, NULL, s->drop_output, tmp_arg, 0);
766 }
767
768 int swr_inject_silence(struct SwrContext *s, int count){
769     int ret, i;
770     uint8_t *tmp_arg[SWR_CH_MAX];
771
772     if(count <= 0)
773         return 0;
774
775 #define MAX_SILENCE_STEP 16384
776     while (count > MAX_SILENCE_STEP) {
777         if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
778             return ret;
779         count -= MAX_SILENCE_STEP;
780     }
781
782     if((ret=swri_realloc_audio(&s->silence, count))<0)
783         return ret;
784
785     if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
786         memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
787     } else
788         memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
789
790     reversefill_audiodata(&s->silence, tmp_arg);
791     av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
792     ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
793     return ret;
794 }
795
796 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
797     if (s->resampler && s->resample){
798         return s->resampler->get_delay(s, base);
799     }else{
800         return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
801     }
802 }
803
804 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
805     int ret;
806
807     if (!s || compensation_distance < 0)
808         return AVERROR(EINVAL);
809     if (!compensation_distance && sample_delta)
810         return AVERROR(EINVAL);
811     if (!s->resample) {
812         s->flags |= SWR_FLAG_RESAMPLE;
813         ret = swr_init(s);
814         if (ret < 0)
815             return ret;
816     }
817     if (!s->resampler->set_compensation){
818         return AVERROR(EINVAL);
819     }else{
820         return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
821     }
822 }
823
824 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
825     if(pts == INT64_MIN)
826         return s->outpts;
827
828     if (s->firstpts == AV_NOPTS_VALUE)
829         s->outpts = s->firstpts = pts;
830
831     if(s->min_compensation >= FLT_MAX) {
832         return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
833     } else {
834         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;
835         double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
836
837         if(fabs(fdelta) > s->min_compensation) {
838             if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
839                 int ret;
840                 if(delta > 0) ret = swr_inject_silence(s,  delta / s->out_sample_rate);
841                 else          ret = swr_drop_output   (s, -delta / s-> in_sample_rate);
842                 if(ret<0){
843                     av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
844                 }
845             } else if(s->soft_compensation_duration && s->max_soft_compensation) {
846                 int duration = s->out_sample_rate * s->soft_compensation_duration;
847                 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
848                 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
849                 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
850                 swr_set_compensation(s, comp, duration);
851             }
852         }
853
854         return s->outpts;
855     }
856 }