]> git.sesse.net Git - ffmpeg/blob - libswresample/swresample.c
Merge commit '5e87080f2c73186066df0b9c43877b4af0beef3a'
[ffmpeg] / 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 #include "libavutil/internal.h"
27
28 #include <float.h>
29
30 #define ALIGN 32
31
32 #include "libavutil/ffversion.h"
33 const char swr_ffversion[] = "FFmpeg version " FFMPEG_VERSION;
34
35 unsigned swresample_version(void)
36 {
37     av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
38     return LIBSWRESAMPLE_VERSION_INT;
39 }
40
41 const char *swresample_configuration(void)
42 {
43     return FFMPEG_CONFIGURATION;
44 }
45
46 const char *swresample_license(void)
47 {
48 #define LICENSE_PREFIX "libswresample license: "
49     return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
50 }
51
52 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
53     if(!s || s->in_convert) // s needs to be allocated but not initialized
54         return AVERROR(EINVAL);
55     s->channel_map = channel_map;
56     return 0;
57 }
58
59 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
60                                       int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
61                                       int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
62                                       int log_offset, void *log_ctx){
63     if(!s) s= swr_alloc();
64     if(!s) return NULL;
65
66     s->log_level_offset= log_offset;
67     s->log_ctx= log_ctx;
68
69     if (av_opt_set_int(s, "ocl", out_ch_layout,   0) < 0)
70         goto fail;
71
72     if (av_opt_set_int(s, "osf", out_sample_fmt,  0) < 0)
73         goto fail;
74
75     if (av_opt_set_int(s, "osr", out_sample_rate, 0) < 0)
76         goto fail;
77
78     if (av_opt_set_int(s, "icl", in_ch_layout,    0) < 0)
79         goto fail;
80
81     if (av_opt_set_int(s, "isf", in_sample_fmt,   0) < 0)
82         goto fail;
83
84     if (av_opt_set_int(s, "isr", in_sample_rate,  0) < 0)
85         goto fail;
86
87     if (av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE,   0) < 0)
88         goto fail;
89
90     if (av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> user_in_ch_layout), 0) < 0)
91         goto fail;
92
93     if (av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->user_out_ch_layout), 0) < 0)
94         goto fail;
95
96     av_opt_set_int(s, "uch", 0, 0);
97     return s;
98 fail:
99     av_log(s, AV_LOG_ERROR, "Failed to set option\n");
100     swr_free(&s);
101     return NULL;
102 }
103
104 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
105     a->fmt   = fmt;
106     a->bps   = av_get_bytes_per_sample(fmt);
107     a->planar= av_sample_fmt_is_planar(fmt);
108     if (a->ch_count == 1)
109         a->planar = 1;
110 }
111
112 static void free_temp(AudioData *a){
113     av_free(a->data);
114     memset(a, 0, sizeof(*a));
115 }
116
117 static void clear_context(SwrContext *s){
118     s->in_buffer_index= 0;
119     s->in_buffer_count= 0;
120     s->resample_in_constraint= 0;
121     memset(s->in.ch, 0, sizeof(s->in.ch));
122     memset(s->out.ch, 0, sizeof(s->out.ch));
123     free_temp(&s->postin);
124     free_temp(&s->midbuf);
125     free_temp(&s->preout);
126     free_temp(&s->in_buffer);
127     free_temp(&s->silence);
128     free_temp(&s->drop_temp);
129     free_temp(&s->dither.noise);
130     free_temp(&s->dither.temp);
131     swri_audio_convert_free(&s-> in_convert);
132     swri_audio_convert_free(&s->out_convert);
133     swri_audio_convert_free(&s->full_convert);
134     swri_rematrix_free(s);
135
136     s->flushed = 0;
137 }
138
139 av_cold void swr_free(SwrContext **ss){
140     SwrContext *s= *ss;
141     if(s){
142         clear_context(s);
143         if (s->resampler)
144             s->resampler->free(&s->resample);
145     }
146
147     av_freep(ss);
148 }
149
150 av_cold void swr_close(SwrContext *s){
151     clear_context(s);
152 }
153
154 av_cold int swr_init(struct SwrContext *s){
155     int ret;
156     char l1[1024], l2[1024];
157
158     clear_context(s);
159
160     if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
161         av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
162         return AVERROR(EINVAL);
163     }
164     if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
165         av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
166         return AVERROR(EINVAL);
167     }
168
169     s->out.ch_count  = s-> user_out_ch_count;
170     s-> in.ch_count  = s->  user_in_ch_count;
171     s->used_ch_count = s->user_used_ch_count;
172
173     s-> in_ch_layout = s-> user_in_ch_layout;
174     s->out_ch_layout = s->user_out_ch_layout;
175
176     if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) {
177         av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
178         s->in_ch_layout = 0;
179     }
180
181     if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) {
182         av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
183         s->out_ch_layout = 0;
184     }
185
186     switch(s->engine){
187 #if CONFIG_LIBSOXR
188         case SWR_ENGINE_SOXR: s->resampler = &swri_soxr_resampler; break;
189 #endif
190         case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
191         default:
192             av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
193             return AVERROR(EINVAL);
194     }
195
196     if(!s->used_ch_count)
197         s->used_ch_count= s->in.ch_count;
198
199     if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
200         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
201         s-> in_ch_layout= 0;
202     }
203
204     if(!s-> in_ch_layout)
205         s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
206     if(!s->out_ch_layout)
207         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
208
209     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
210                  s->rematrix_custom;
211
212     if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
213         if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
214             s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
215         }else if(   av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
216                  && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P
217                  && !s->rematrix
218                  && s->engine != SWR_ENGINE_SOXR){
219             s->int_sample_fmt= AV_SAMPLE_FMT_S32P;
220         }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
221             s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
222         }else{
223             av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
224             s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
225         }
226     }
227
228     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P
229         &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
230         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
231         &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
232         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));
233         return AVERROR(EINVAL);
234     }
235
236     set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
237     set_audiodata_fmt(&s->out, s->out_sample_fmt);
238
239     if (s->firstpts_in_samples != AV_NOPTS_VALUE) {
240         if (!s->async && s->min_compensation >= FLT_MAX/2)
241             s->async = 1;
242         s->firstpts =
243         s->outpts   = s->firstpts_in_samples * s->out_sample_rate;
244     } else
245         s->firstpts = AV_NOPTS_VALUE;
246
247     if (s->async) {
248         if (s->min_compensation >= FLT_MAX/2)
249             s->min_compensation = 0.001;
250         if (s->async > 1.0001) {
251             s->max_soft_compensation = s->async / (double) s->in_sample_rate;
252         }
253     }
254
255     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
256         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);
257         if (!s->resample) {
258             av_log(s, AV_LOG_ERROR, "Failed to initialize resampler\n");
259             return AVERROR(ENOMEM);
260         }
261     }else
262         s->resampler->free(&s->resample);
263     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P
264         && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
265         && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
266         && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
267         && s->resample){
268         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
269         ret = AVERROR(EINVAL);
270         goto fail;
271     }
272
273 #define RSC 1 //FIXME finetune
274     if(!s-> in.ch_count)
275         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
276     if(!s->used_ch_count)
277         s->used_ch_count= s->in.ch_count;
278     if(!s->out.ch_count)
279         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
280
281     if(!s-> in.ch_count){
282         av_assert0(!s->in_ch_layout);
283         av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
284         ret = AVERROR(EINVAL);
285         goto fail;
286     }
287
288     av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
289     av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
290     if (s->out_ch_layout && s->out.ch_count != av_get_channel_layout_nb_channels(s->out_ch_layout)) {
291         av_log(s, AV_LOG_ERROR, "Output channel layout %s mismatches specified channel count %d\n", l2, s->out.ch_count);
292         ret = AVERROR(EINVAL);
293         goto fail;
294     }
295     if (s->in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s->in_ch_layout)) {
296         av_log(s, AV_LOG_ERROR, "Input channel layout %s mismatches specified channel count %d\n", l1, s->used_ch_count);
297         ret = AVERROR(EINVAL);
298         goto fail;
299     }
300
301     if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
302         av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
303                "but there is not enough information to do it\n", l1, l2);
304         ret = AVERROR(EINVAL);
305         goto fail;
306     }
307
308 av_assert0(s->used_ch_count);
309 av_assert0(s->out.ch_count);
310     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
311
312     s->in_buffer= s->in;
313     s->silence  = s->in;
314     s->drop_temp= s->out;
315
316     if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
317         s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
318                                                    s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
319         return 0;
320     }
321
322     s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
323                                              s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
324     s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
325                                              s->int_sample_fmt, s->out.ch_count, NULL, 0);
326
327     if (!s->in_convert || !s->out_convert) {
328         ret = AVERROR(ENOMEM);
329         goto fail;
330     }
331
332     s->postin= s->in;
333     s->preout= s->out;
334     s->midbuf= s->in;
335
336     if(s->channel_map){
337         s->postin.ch_count=
338         s->midbuf.ch_count= s->used_ch_count;
339         if(s->resample)
340             s->in_buffer.ch_count= s->used_ch_count;
341     }
342     if(!s->resample_first){
343         s->midbuf.ch_count= s->out.ch_count;
344         if(s->resample)
345             s->in_buffer.ch_count = s->out.ch_count;
346     }
347
348     set_audiodata_fmt(&s->postin, s->int_sample_fmt);
349     set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
350     set_audiodata_fmt(&s->preout, s->int_sample_fmt);
351
352     if(s->resample){
353         set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
354     }
355
356     if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
357         goto fail;
358
359     if(s->rematrix || s->dither.method) {
360         ret = swri_rematrix_init(s);
361         if (ret < 0)
362             goto fail;
363     }
364
365     return 0;
366 fail:
367     swr_close(s);
368     return ret;
369
370 }
371
372 int swri_realloc_audio(AudioData *a, int count){
373     int i, countb;
374     AudioData old;
375
376     if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
377         return AVERROR(EINVAL);
378
379     if(a->count >= count)
380         return 0;
381
382     count*=2;
383
384     countb= FFALIGN(count*a->bps, ALIGN);
385     old= *a;
386
387     av_assert0(a->bps);
388     av_assert0(a->ch_count);
389
390     a->data= av_mallocz_array(countb, a->ch_count);
391     if(!a->data)
392         return AVERROR(ENOMEM);
393     for(i=0; i<a->ch_count; i++){
394         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
395         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
396     }
397     if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
398     av_freep(&old.data);
399     a->count= count;
400
401     return 1;
402 }
403
404 static void copy(AudioData *out, AudioData *in,
405                  int count){
406     av_assert0(out->planar == in->planar);
407     av_assert0(out->bps == in->bps);
408     av_assert0(out->ch_count == in->ch_count);
409     if(out->planar){
410         int ch;
411         for(ch=0; ch<out->ch_count; ch++)
412             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
413     }else
414         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
415 }
416
417 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
418     int i;
419     if(!in_arg){
420         memset(out->ch, 0, sizeof(out->ch));
421     }else if(out->planar){
422         for(i=0; i<out->ch_count; i++)
423             out->ch[i]= in_arg[i];
424     }else{
425         for(i=0; i<out->ch_count; i++)
426             out->ch[i]= in_arg[0] + i*out->bps;
427     }
428 }
429
430 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
431     int i;
432     if(out->planar){
433         for(i=0; i<out->ch_count; i++)
434             in_arg[i]= out->ch[i];
435     }else{
436         in_arg[0]= out->ch[0];
437     }
438 }
439
440 /**
441  *
442  * out may be equal in.
443  */
444 static void buf_set(AudioData *out, AudioData *in, int count){
445     int ch;
446     if(in->planar){
447         for(ch=0; ch<out->ch_count; ch++)
448             out->ch[ch]= in->ch[ch] + count*out->bps;
449     }else{
450         for(ch=out->ch_count-1; ch>=0; ch--)
451             out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
452     }
453 }
454
455 /**
456  *
457  * @return number of samples output per channel
458  */
459 static int resample(SwrContext *s, AudioData *out_param, int out_count,
460                              const AudioData * in_param, int in_count){
461     AudioData in, out, tmp;
462     int ret_sum=0;
463     int border=0;
464     int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
465
466     av_assert1(s->in_buffer.ch_count == in_param->ch_count);
467     av_assert1(s->in_buffer.planar   == in_param->planar);
468     av_assert1(s->in_buffer.fmt      == in_param->fmt);
469
470     tmp=out=*out_param;
471     in =  *in_param;
472
473     border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
474                  &in, in_count, &s->in_buffer_index, &s->in_buffer_count);
475     if (border == INT_MAX) {
476         return 0;
477     } else if (border < 0) {
478         return border;
479     } else if (border) {
480         buf_set(&in, &in, border);
481         in_count -= border;
482         s->resample_in_constraint = 0;
483     }
484
485     do{
486         int ret, size, consumed;
487         if(!s->resample_in_constraint && s->in_buffer_count){
488             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
489             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
490             out_count -= ret;
491             ret_sum += ret;
492             buf_set(&out, &out, ret);
493             s->in_buffer_count -= consumed;
494             s->in_buffer_index += consumed;
495
496             if(!in_count)
497                 break;
498             if(s->in_buffer_count <= border){
499                 buf_set(&in, &in, -s->in_buffer_count);
500                 in_count += s->in_buffer_count;
501                 s->in_buffer_count=0;
502                 s->in_buffer_index=0;
503                 border = 0;
504             }
505         }
506
507         if((s->flushed || in_count > padless) && !s->in_buffer_count){
508             s->in_buffer_index=0;
509             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
510             out_count -= ret;
511             ret_sum += ret;
512             buf_set(&out, &out, ret);
513             in_count -= consumed;
514             buf_set(&in, &in, consumed);
515         }
516
517         //TODO is this check sane considering the advanced copy avoidance below
518         size= s->in_buffer_index + s->in_buffer_count + in_count;
519         if(   size > s->in_buffer.count
520            && s->in_buffer_count + in_count <= s->in_buffer_index){
521             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
522             copy(&s->in_buffer, &tmp, s->in_buffer_count);
523             s->in_buffer_index=0;
524         }else
525             if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
526                 return ret;
527
528         if(in_count){
529             int count= in_count;
530             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
531
532             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
533             copy(&tmp, &in, /*in_*/count);
534             s->in_buffer_count += count;
535             in_count -= count;
536             border += count;
537             buf_set(&in, &in, count);
538             s->resample_in_constraint= 0;
539             if(s->in_buffer_count != count || in_count)
540                 continue;
541             if (padless) {
542                 padless = 0;
543                 continue;
544             }
545         }
546         break;
547     }while(1);
548
549     s->resample_in_constraint= !!out_count;
550
551     return ret_sum;
552 }
553
554 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
555                                                       AudioData *in , int  in_count){
556     AudioData *postin, *midbuf, *preout;
557     int ret/*, in_max*/;
558     AudioData preout_tmp, midbuf_tmp;
559
560     if(s->full_convert){
561         av_assert0(!s->resample);
562         swri_audio_convert(s->full_convert, out, in, in_count);
563         return out_count;
564     }
565
566 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
567 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
568
569     if((ret=swri_realloc_audio(&s->postin, in_count))<0)
570         return ret;
571     if(s->resample_first){
572         av_assert0(s->midbuf.ch_count == s->used_ch_count);
573         if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
574             return ret;
575     }else{
576         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
577         if((ret=swri_realloc_audio(&s->midbuf,  in_count))<0)
578             return ret;
579     }
580     if((ret=swri_realloc_audio(&s->preout, out_count))<0)
581         return ret;
582
583     postin= &s->postin;
584
585     midbuf_tmp= s->midbuf;
586     midbuf= &midbuf_tmp;
587     preout_tmp= s->preout;
588     preout= &preout_tmp;
589
590     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
591         postin= in;
592
593     if(s->resample_first ? !s->resample : !s->rematrix)
594         midbuf= postin;
595
596     if(s->resample_first ? !s->rematrix : !s->resample)
597         preout= midbuf;
598
599     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
600        && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){
601         if(preout==in){
602             out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
603             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
604             copy(out, in, out_count);
605             return out_count;
606         }
607         else if(preout==postin) preout= midbuf= postin= out;
608         else if(preout==midbuf) preout= midbuf= out;
609         else                    preout= out;
610     }
611
612     if(in != postin){
613         swri_audio_convert(s->in_convert, postin, in, in_count);
614     }
615
616     if(s->resample_first){
617         if(postin != midbuf)
618             out_count= resample(s, midbuf, out_count, postin, in_count);
619         if(midbuf != preout)
620             swri_rematrix(s, preout, midbuf, out_count, preout==out);
621     }else{
622         if(postin != midbuf)
623             swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
624         if(midbuf != preout)
625             out_count= resample(s, preout, out_count, midbuf, in_count);
626     }
627
628     if(preout != out && out_count){
629         AudioData *conv_src = preout;
630         if(s->dither.method){
631             int ch;
632             int dither_count= FFMAX(out_count, 1<<16);
633
634             if (preout == in) {
635                 conv_src = &s->dither.temp;
636                 if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
637                     return ret;
638             }
639
640             if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
641                 return ret;
642             if(ret)
643                 for(ch=0; ch<s->dither.noise.ch_count; ch++)
644                     if((ret=swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt))<0)
645                         return ret;
646             av_assert0(s->dither.noise.ch_count == preout->ch_count);
647
648             if(s->dither.noise_pos + out_count > s->dither.noise.count)
649                 s->dither.noise_pos = 0;
650
651             if (s->dither.method < SWR_DITHER_NS){
652                 if (s->mix_2_1_simd) {
653                     int len1= out_count&~15;
654                     int off = len1 * preout->bps;
655
656                     if(len1)
657                         for(ch=0; ch<preout->ch_count; ch++)
658                             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);
659                     if(out_count != len1)
660                         for(ch=0; ch<preout->ch_count; ch++)
661                             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);
662                 } else {
663                     for(ch=0; ch<preout->ch_count; ch++)
664                         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);
665                 }
666             } else {
667                 switch(s->int_sample_fmt) {
668                 case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
669                 case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
670                 case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
671                 case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
672                 }
673             }
674             s->dither.noise_pos += out_count;
675         }
676 //FIXME packed doesn't need more than 1 chan here!
677         swri_audio_convert(s->out_convert, out, conv_src, out_count);
678     }
679     return out_count;
680 }
681
682 int swr_is_initialized(struct SwrContext *s) {
683     return !!s->in_buffer.ch_count;
684 }
685
686 int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
687                                                     const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
688     AudioData * in= &s->in;
689     AudioData *out= &s->out;
690     int av_unused max_output;
691
692     if (!swr_is_initialized(s)) {
693         av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
694         return AVERROR(EINVAL);
695     }
696 #if defined(ASSERT_LEVEL) && ASSERT_LEVEL >1
697     max_output = swr_get_out_samples(s, in_count);
698 #endif
699
700     while(s->drop_output > 0){
701         int ret;
702         uint8_t *tmp_arg[SWR_CH_MAX];
703 #define MAX_DROP_STEP 16384
704         if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)
705             return ret;
706
707         reversefill_audiodata(&s->drop_temp, tmp_arg);
708         s->drop_output *= -1; //FIXME find a less hackish solution
709         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
710         s->drop_output *= -1;
711         in_count = 0;
712         if(ret>0) {
713             s->drop_output -= ret;
714             if (!s->drop_output && !out_arg)
715                 return 0;
716             continue;
717         }
718
719         av_assert0(s->drop_output);
720         return 0;
721     }
722
723     if(!in_arg){
724         if(s->resample){
725             if (!s->flushed)
726                 s->resampler->flush(s);
727             s->resample_in_constraint = 0;
728             s->flushed = 1;
729         }else if(!s->in_buffer_count){
730             return 0;
731         }
732     }else
733         fill_audiodata(in ,  (void*)in_arg);
734
735     fill_audiodata(out, out_arg);
736
737     if(s->resample){
738         int ret = swr_convert_internal(s, out, out_count, in, in_count);
739         if(ret>0 && !s->drop_output)
740             s->outpts += ret * (int64_t)s->in_sample_rate;
741
742         av_assert2(max_output < 0 || ret < 0 || ret <= max_output);
743
744         return ret;
745     }else{
746         AudioData tmp= *in;
747         int ret2=0;
748         int ret, size;
749         size = FFMIN(out_count, s->in_buffer_count);
750         if(size){
751             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
752             ret= swr_convert_internal(s, out, size, &tmp, size);
753             if(ret<0)
754                 return ret;
755             ret2= ret;
756             s->in_buffer_count -= ret;
757             s->in_buffer_index += ret;
758             buf_set(out, out, ret);
759             out_count -= ret;
760             if(!s->in_buffer_count)
761                 s->in_buffer_index = 0;
762         }
763
764         if(in_count){
765             size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
766
767             if(in_count > out_count) { //FIXME move after swr_convert_internal
768                 if(   size > s->in_buffer.count
769                 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
770                     buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
771                     copy(&s->in_buffer, &tmp, s->in_buffer_count);
772                     s->in_buffer_index=0;
773                 }else
774                     if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
775                         return ret;
776             }
777
778             if(out_count){
779                 size = FFMIN(in_count, out_count);
780                 ret= swr_convert_internal(s, out, size, in, size);
781                 if(ret<0)
782                     return ret;
783                 buf_set(in, in, ret);
784                 in_count -= ret;
785                 ret2 += ret;
786             }
787             if(in_count){
788                 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
789                 copy(&tmp, in, in_count);
790                 s->in_buffer_count += in_count;
791             }
792         }
793         if(ret2>0 && !s->drop_output)
794             s->outpts += ret2 * (int64_t)s->in_sample_rate;
795         av_assert2(max_output < 0 || ret2 < 0 || ret2 <= max_output);
796         return ret2;
797     }
798 }
799
800 int swr_drop_output(struct SwrContext *s, int count){
801     const uint8_t *tmp_arg[SWR_CH_MAX];
802     s->drop_output += count;
803
804     if(s->drop_output <= 0)
805         return 0;
806
807     av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
808     return swr_convert(s, NULL, s->drop_output, tmp_arg, 0);
809 }
810
811 int swr_inject_silence(struct SwrContext *s, int count){
812     int ret, i;
813     uint8_t *tmp_arg[SWR_CH_MAX];
814
815     if(count <= 0)
816         return 0;
817
818 #define MAX_SILENCE_STEP 16384
819     while (count > MAX_SILENCE_STEP) {
820         if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
821             return ret;
822         count -= MAX_SILENCE_STEP;
823     }
824
825     if((ret=swri_realloc_audio(&s->silence, count))<0)
826         return ret;
827
828     if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
829         memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
830     } else
831         memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
832
833     reversefill_audiodata(&s->silence, tmp_arg);
834     av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
835     ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
836     return ret;
837 }
838
839 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
840     if (s->resampler && s->resample){
841         return s->resampler->get_delay(s, base);
842     }else{
843         return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
844     }
845 }
846
847 int swr_get_out_samples(struct SwrContext *s, int in_samples)
848 {
849     int64_t out_samples;
850
851     if (in_samples < 0)
852         return AVERROR(EINVAL);
853
854     if (s->resampler && s->resample) {
855         if (!s->resampler->get_out_samples)
856             return AVERROR(ENOSYS);
857         out_samples = s->resampler->get_out_samples(s, in_samples);
858     } else {
859         out_samples = s->in_buffer_count + in_samples;
860         av_assert0(s->out_sample_rate == s->in_sample_rate);
861     }
862
863     if (out_samples > INT_MAX)
864         return AVERROR(EINVAL);
865
866     return out_samples;
867 }
868
869 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
870     int ret;
871
872     if (!s || compensation_distance < 0)
873         return AVERROR(EINVAL);
874     if (!compensation_distance && sample_delta)
875         return AVERROR(EINVAL);
876     if (!s->resample) {
877         s->flags |= SWR_FLAG_RESAMPLE;
878         ret = swr_init(s);
879         if (ret < 0)
880             return ret;
881     }
882     if (!s->resampler->set_compensation){
883         return AVERROR(EINVAL);
884     }else{
885         return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
886     }
887 }
888
889 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
890     if(pts == INT64_MIN)
891         return s->outpts;
892
893     if (s->firstpts == AV_NOPTS_VALUE)
894         s->outpts = s->firstpts = pts;
895
896     if(s->min_compensation >= FLT_MAX) {
897         return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
898     } else {
899         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;
900         double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
901
902         if(fabs(fdelta) > s->min_compensation) {
903             if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
904                 int ret;
905                 if(delta > 0) ret = swr_inject_silence(s,  delta / s->out_sample_rate);
906                 else          ret = swr_drop_output   (s, -delta / s-> in_sample_rate);
907                 if(ret<0){
908                     av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
909                 }
910             } else if(s->soft_compensation_duration && s->max_soft_compensation) {
911                 int duration = s->out_sample_rate * s->soft_compensation_duration;
912                 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
913                 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
914                 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
915                 swr_set_compensation(s, comp, duration);
916             }
917         }
918
919         return s->outpts;
920     }
921 }