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
swresample_version(void)35 unsigned swresample_version(void)
36 {
37 av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
38 return LIBSWRESAMPLE_VERSION_INT;
39 }
40
swresample_configuration(void)41 const char *swresample_configuration(void)
42 {
43 return FFMPEG_CONFIGURATION;
44 }
45
swresample_license(void)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
swr_set_channel_mapping(struct SwrContext * s,const int * channel_map)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
swr_alloc_set_opts(struct SwrContext * s,int64_t out_ch_layout,enum AVSampleFormat out_sample_fmt,int out_sample_rate,int64_t in_ch_layout,enum AVSampleFormat in_sample_fmt,int in_sample_rate,int log_offset,void * log_ctx)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, "ich", av_get_channel_layout_nb_channels(s-> user_in_ch_layout), 0) < 0)
88 goto fail;
89
90 if (av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->user_out_ch_layout), 0) < 0)
91 goto fail;
92
93 av_opt_set_int(s, "uch", 0, 0);
94 return s;
95 fail:
96 av_log(s, AV_LOG_ERROR, "Failed to set option\n");
97 swr_free(&s);
98 return NULL;
99 }
100
set_audiodata_fmt(AudioData * a,enum AVSampleFormat fmt)101 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
102 a->fmt = fmt;
103 a->bps = av_get_bytes_per_sample(fmt);
104 a->planar= av_sample_fmt_is_planar(fmt);
105 if (a->ch_count == 1)
106 a->planar = 1;
107 }
108
free_temp(AudioData * a)109 static void free_temp(AudioData *a){
110 av_free(a->data);
111 memset(a, 0, sizeof(*a));
112 }
113
clear_context(SwrContext * s)114 static void clear_context(SwrContext *s){
115 s->in_buffer_index= 0;
116 s->in_buffer_count= 0;
117 s->resample_in_constraint= 0;
118 memset(s->in.ch, 0, sizeof(s->in.ch));
119 memset(s->out.ch, 0, sizeof(s->out.ch));
120 free_temp(&s->postin);
121 free_temp(&s->midbuf);
122 free_temp(&s->preout);
123 free_temp(&s->in_buffer);
124 free_temp(&s->silence);
125 free_temp(&s->drop_temp);
126 free_temp(&s->dither.noise);
127 free_temp(&s->dither.temp);
128 swri_audio_convert_free(&s-> in_convert);
129 swri_audio_convert_free(&s->out_convert);
130 swri_audio_convert_free(&s->full_convert);
131 swri_rematrix_free(s);
132
133 s->delayed_samples_fixup = 0;
134 s->flushed = 0;
135 }
136
swr_free(SwrContext ** ss)137 av_cold void swr_free(SwrContext **ss){
138 SwrContext *s= *ss;
139 if(s){
140 clear_context(s);
141 if (s->resampler)
142 s->resampler->free(&s->resample);
143 }
144
145 av_freep(ss);
146 }
147
swr_close(SwrContext * s)148 av_cold void swr_close(SwrContext *s){
149 clear_context(s);
150 }
151
swr_init(struct SwrContext * s)152 av_cold int swr_init(struct SwrContext *s){
153 int ret;
154 char l1[1024], l2[1024];
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(s-> in_sample_rate <= 0){
168 av_log(s, AV_LOG_ERROR, "Requested input sample rate %d is invalid\n", s->in_sample_rate);
169 return AVERROR(EINVAL);
170 }
171 if(s->out_sample_rate <= 0){
172 av_log(s, AV_LOG_ERROR, "Requested output sample rate %d is invalid\n", s->out_sample_rate);
173 return AVERROR(EINVAL);
174 }
175 s->out.ch_count = s-> user_out_ch_count;
176 s-> in.ch_count = s-> user_in_ch_count;
177 s->used_ch_count = s->user_used_ch_count;
178
179 s-> in_ch_layout = s-> user_in_ch_layout;
180 s->out_ch_layout = s->user_out_ch_layout;
181
182 s->int_sample_fmt= s->user_int_sample_fmt;
183
184 s->dither.method = s->user_dither_method;
185
186 if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) {
187 av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
188 s->in_ch_layout = 0;
189 }
190
191 if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) {
192 av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
193 s->out_ch_layout = 0;
194 }
195
196 switch(s->engine){
197 #if CONFIG_LIBSOXR
198 case SWR_ENGINE_SOXR: s->resampler = &swri_soxr_resampler; break;
199 #endif
200 case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
201 default:
202 av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
203 return AVERROR(EINVAL);
204 }
205
206 if(!s->used_ch_count)
207 s->used_ch_count= s->in.ch_count;
208
209 if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
210 av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
211 s-> in_ch_layout= 0;
212 }
213
214 if(!s-> in_ch_layout)
215 s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
216 if(!s->out_ch_layout)
217 s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
218
219 s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
220 s->rematrix_custom;
221
222 if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
223 if( av_get_bytes_per_sample(s-> in_sample_fmt) <= 2
224 && av_get_bytes_per_sample(s->out_sample_fmt) <= 2){
225 s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
226 }else if( av_get_bytes_per_sample(s-> in_sample_fmt) <= 2
227 && !s->rematrix
228 && s->out_sample_rate==s->in_sample_rate
229 && !(s->flags & SWR_FLAG_RESAMPLE)){
230 s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
231 }else if( av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
232 && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P
233 && !s->rematrix
234 && s->out_sample_rate == s->in_sample_rate
235 && !(s->flags & SWR_FLAG_RESAMPLE)
236 && s->engine != SWR_ENGINE_SOXR){
237 s->int_sample_fmt= AV_SAMPLE_FMT_S32P;
238 }else if(av_get_bytes_per_sample(s->in_sample_fmt) <= 4){
239 s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
240 }else{
241 s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
242 }
243 }
244 av_log(s, AV_LOG_DEBUG, "Using %s internally between filters\n", av_get_sample_fmt_name(s->int_sample_fmt));
245
246 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
247 &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
248 &&s->int_sample_fmt != AV_SAMPLE_FMT_S64P
249 &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
250 &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
251 av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, s16p/s32p/s64p/fltp/dblp are supported\n", av_get_sample_fmt_name(s->int_sample_fmt));
252 return AVERROR(EINVAL);
253 }
254
255 set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
256 set_audiodata_fmt(&s->out, s->out_sample_fmt);
257
258 if (s->firstpts_in_samples != AV_NOPTS_VALUE) {
259 if (!s->async && s->min_compensation >= FLT_MAX/2)
260 s->async = 1;
261 s->firstpts =
262 s->outpts = s->firstpts_in_samples * s->out_sample_rate;
263 } else
264 s->firstpts = AV_NOPTS_VALUE;
265
266 if (s->async) {
267 if (s->min_compensation >= FLT_MAX/2)
268 s->min_compensation = 0.001;
269 if (s->async > 1.0001) {
270 s->max_soft_compensation = s->async / (double) s->in_sample_rate;
271 }
272 }
273
274 if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
275 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, s->exact_rational);
276 if (!s->resample) {
277 av_log(s, AV_LOG_ERROR, "Failed to initialize resampler\n");
278 return AVERROR(ENOMEM);
279 }
280 }else
281 s->resampler->free(&s->resample);
282 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
283 && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
284 && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
285 && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
286 && s->resample){
287 av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16p/s32p/fltp/dblp\n");
288 ret = AVERROR(EINVAL);
289 goto fail;
290 }
291
292 #define RSC 1 //FIXME finetune
293 if(!s-> in.ch_count)
294 s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
295 if(!s->used_ch_count)
296 s->used_ch_count= s->in.ch_count;
297 if(!s->out.ch_count)
298 s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
299
300 if(!s-> in.ch_count){
301 av_assert0(!s->in_ch_layout);
302 av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
303 ret = AVERROR(EINVAL);
304 goto fail;
305 }
306
307 av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
308 av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
309 if (s->out_ch_layout && s->out.ch_count != av_get_channel_layout_nb_channels(s->out_ch_layout)) {
310 av_log(s, AV_LOG_ERROR, "Output channel layout %s mismatches specified channel count %d\n", l2, s->out.ch_count);
311 ret = AVERROR(EINVAL);
312 goto fail;
313 }
314 if (s->in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s->in_ch_layout)) {
315 av_log(s, AV_LOG_ERROR, "Input channel layout %s mismatches specified channel count %d\n", l1, s->used_ch_count);
316 ret = AVERROR(EINVAL);
317 goto fail;
318 }
319
320 if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
321 av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
322 "but there is not enough information to do it\n", l1, l2);
323 ret = AVERROR(EINVAL);
324 goto fail;
325 }
326
327 av_assert0(s->used_ch_count);
328 av_assert0(s->out.ch_count);
329 s->resample_first= RSC*s->out.ch_count/s->used_ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
330
331 s->in_buffer= s->in;
332 s->silence = s->in;
333 s->drop_temp= s->out;
334
335 if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
336 goto fail;
337
338 if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
339 s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
340 s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
341 return 0;
342 }
343
344 s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
345 s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
346 s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
347 s->int_sample_fmt, s->out.ch_count, NULL, 0);
348
349 if (!s->in_convert || !s->out_convert) {
350 ret = AVERROR(ENOMEM);
351 goto fail;
352 }
353
354 s->postin= s->in;
355 s->preout= s->out;
356 s->midbuf= s->in;
357
358 if(s->channel_map){
359 s->postin.ch_count=
360 s->midbuf.ch_count= s->used_ch_count;
361 if(s->resample)
362 s->in_buffer.ch_count= s->used_ch_count;
363 }
364 if(!s->resample_first){
365 s->midbuf.ch_count= s->out.ch_count;
366 if(s->resample)
367 s->in_buffer.ch_count = s->out.ch_count;
368 }
369
370 set_audiodata_fmt(&s->postin, s->int_sample_fmt);
371 set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
372 set_audiodata_fmt(&s->preout, s->int_sample_fmt);
373
374 if(s->resample){
375 set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
376 }
377
378 av_assert0(!s->preout.count);
379 s->dither.noise = s->preout;
380 s->dither.temp = s->preout;
381 if (s->dither.method > SWR_DITHER_NS) {
382 s->dither.noise.bps = 4;
383 s->dither.noise.fmt = AV_SAMPLE_FMT_FLTP;
384 s->dither.noise_scale = 1;
385 }
386
387 if(s->rematrix || s->dither.method) {
388 ret = swri_rematrix_init(s);
389 if (ret < 0)
390 goto fail;
391 }
392
393 return 0;
394 fail:
395 swr_close(s);
396 return ret;
397
398 }
399
swri_realloc_audio(AudioData * a,int count)400 int swri_realloc_audio(AudioData *a, int count){
401 int i, countb;
402 AudioData old;
403
404 if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
405 return AVERROR(EINVAL);
406
407 if(a->count >= count)
408 return 0;
409
410 count*=2;
411
412 countb= FFALIGN(count*a->bps, ALIGN);
413 old= *a;
414
415 av_assert0(a->bps);
416 av_assert0(a->ch_count);
417
418 a->data= av_mallocz_array(countb, a->ch_count);
419 if(!a->data)
420 return AVERROR(ENOMEM);
421 for(i=0; i<a->ch_count; i++){
422 a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
423 if(a->count && a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
424 }
425 if(a->count && !a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
426 av_freep(&old.data);
427 a->count= count;
428
429 return 1;
430 }
431
copy(AudioData * out,AudioData * in,int count)432 static void copy(AudioData *out, AudioData *in,
433 int count){
434 av_assert0(out->planar == in->planar);
435 av_assert0(out->bps == in->bps);
436 av_assert0(out->ch_count == in->ch_count);
437 if(out->planar){
438 int ch;
439 for(ch=0; ch<out->ch_count; ch++)
440 memcpy(out->ch[ch], in->ch[ch], count*out->bps);
441 }else
442 memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
443 }
444
fill_audiodata(AudioData * out,uint8_t * in_arg[SWR_CH_MAX])445 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
446 int i;
447 if(!in_arg){
448 memset(out->ch, 0, sizeof(out->ch));
449 }else if(out->planar){
450 for(i=0; i<out->ch_count; i++)
451 out->ch[i]= in_arg[i];
452 }else{
453 for(i=0; i<out->ch_count; i++)
454 out->ch[i]= in_arg[0] + i*out->bps;
455 }
456 }
457
reversefill_audiodata(AudioData * out,uint8_t * in_arg[SWR_CH_MAX])458 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
459 int i;
460 if(out->planar){
461 for(i=0; i<out->ch_count; i++)
462 in_arg[i]= out->ch[i];
463 }else{
464 in_arg[0]= out->ch[0];
465 }
466 }
467
468 /**
469 *
470 * out may be equal in.
471 */
buf_set(AudioData * out,AudioData * in,int count)472 static void buf_set(AudioData *out, AudioData *in, int count){
473 int ch;
474 if(in->planar){
475 for(ch=0; ch<out->ch_count; ch++)
476 out->ch[ch]= in->ch[ch] + count*out->bps;
477 }else{
478 for(ch=out->ch_count-1; ch>=0; ch--)
479 out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
480 }
481 }
482
483 /**
484 *
485 * @return number of samples output per channel
486 */
resample(SwrContext * s,AudioData * out_param,int out_count,const AudioData * in_param,int in_count)487 static int resample(SwrContext *s, AudioData *out_param, int out_count,
488 const AudioData * in_param, int in_count){
489 AudioData in, out, tmp;
490 int ret_sum=0;
491 int border=0;
492 int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
493
494 av_assert1(s->in_buffer.ch_count == in_param->ch_count);
495 av_assert1(s->in_buffer.planar == in_param->planar);
496 av_assert1(s->in_buffer.fmt == in_param->fmt);
497
498 tmp=out=*out_param;
499 in = *in_param;
500
501 border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
502 &in, in_count, &s->in_buffer_index, &s->in_buffer_count);
503 if (border == INT_MAX) {
504 return 0;
505 } else if (border < 0) {
506 return border;
507 } else if (border) {
508 buf_set(&in, &in, border);
509 in_count -= border;
510 s->resample_in_constraint = 0;
511 }
512
513 do{
514 int ret, size, consumed;
515 if(!s->resample_in_constraint && s->in_buffer_count){
516 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
517 ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
518 out_count -= ret;
519 ret_sum += ret;
520 buf_set(&out, &out, ret);
521 s->in_buffer_count -= consumed;
522 s->in_buffer_index += consumed;
523
524 if(!in_count)
525 break;
526 if(s->in_buffer_count <= border){
527 buf_set(&in, &in, -s->in_buffer_count);
528 in_count += s->in_buffer_count;
529 s->in_buffer_count=0;
530 s->in_buffer_index=0;
531 border = 0;
532 }
533 }
534
535 if((s->flushed || in_count > padless) && !s->in_buffer_count){
536 s->in_buffer_index=0;
537 ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
538 out_count -= ret;
539 ret_sum += ret;
540 buf_set(&out, &out, ret);
541 in_count -= consumed;
542 buf_set(&in, &in, consumed);
543 }
544
545 //TODO is this check sane considering the advanced copy avoidance below
546 size= s->in_buffer_index + s->in_buffer_count + in_count;
547 if( size > s->in_buffer.count
548 && s->in_buffer_count + in_count <= s->in_buffer_index){
549 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
550 copy(&s->in_buffer, &tmp, s->in_buffer_count);
551 s->in_buffer_index=0;
552 }else
553 if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
554 return ret;
555
556 if(in_count){
557 int count= in_count;
558 if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
559
560 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
561 copy(&tmp, &in, /*in_*/count);
562 s->in_buffer_count += count;
563 in_count -= count;
564 border += count;
565 buf_set(&in, &in, count);
566 s->resample_in_constraint= 0;
567 if(s->in_buffer_count != count || in_count)
568 continue;
569 if (padless) {
570 padless = 0;
571 continue;
572 }
573 }
574 break;
575 }while(1);
576
577 s->resample_in_constraint= !!out_count;
578
579 return ret_sum;
580 }
581
swr_convert_internal(struct SwrContext * s,AudioData * out,int out_count,AudioData * in,int in_count)582 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
583 AudioData *in , int in_count){
584 AudioData *postin, *midbuf, *preout;
585 int ret/*, in_max*/;
586 AudioData preout_tmp, midbuf_tmp;
587
588 if(s->full_convert){
589 av_assert0(!s->resample);
590 swri_audio_convert(s->full_convert, out, in, in_count);
591 return out_count;
592 }
593
594 // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
595 // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
596
597 if((ret=swri_realloc_audio(&s->postin, in_count))<0)
598 return ret;
599 if(s->resample_first){
600 av_assert0(s->midbuf.ch_count == s->used_ch_count);
601 if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
602 return ret;
603 }else{
604 av_assert0(s->midbuf.ch_count == s->out.ch_count);
605 if((ret=swri_realloc_audio(&s->midbuf, in_count))<0)
606 return ret;
607 }
608 if((ret=swri_realloc_audio(&s->preout, out_count))<0)
609 return ret;
610
611 postin= &s->postin;
612
613 midbuf_tmp= s->midbuf;
614 midbuf= &midbuf_tmp;
615 preout_tmp= s->preout;
616 preout= &preout_tmp;
617
618 if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
619 postin= in;
620
621 if(s->resample_first ? !s->resample : !s->rematrix)
622 midbuf= postin;
623
624 if(s->resample_first ? !s->rematrix : !s->resample)
625 preout= midbuf;
626
627 if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
628 && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){
629 if(preout==in){
630 out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
631 av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
632 copy(out, in, out_count);
633 return out_count;
634 }
635 else if(preout==postin) preout= midbuf= postin= out;
636 else if(preout==midbuf) preout= midbuf= out;
637 else preout= out;
638 }
639
640 if(in != postin){
641 swri_audio_convert(s->in_convert, postin, in, in_count);
642 }
643
644 if(s->resample_first){
645 if(postin != midbuf)
646 out_count= resample(s, midbuf, out_count, postin, in_count);
647 if(midbuf != preout)
648 swri_rematrix(s, preout, midbuf, out_count, preout==out);
649 }else{
650 if(postin != midbuf)
651 swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
652 if(midbuf != preout)
653 out_count= resample(s, preout, out_count, midbuf, in_count);
654 }
655
656 if(preout != out && out_count){
657 AudioData *conv_src = preout;
658 if(s->dither.method){
659 int ch;
660 int dither_count= FFMAX(out_count, 1<<16);
661
662 if (preout == in) {
663 conv_src = &s->dither.temp;
664 if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
665 return ret;
666 }
667
668 if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
669 return ret;
670 if(ret)
671 for(ch=0; ch<s->dither.noise.ch_count; ch++)
672 if((ret=swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, (12345678913579ULL*ch + 3141592) % 2718281828U, s->dither.noise.fmt))<0)
673 return ret;
674 av_assert0(s->dither.noise.ch_count == preout->ch_count);
675
676 if(s->dither.noise_pos + out_count > s->dither.noise.count)
677 s->dither.noise_pos = 0;
678
679 if (s->dither.method < SWR_DITHER_NS){
680 if (s->mix_2_1_simd) {
681 int len1= out_count&~15;
682 int off = len1 * preout->bps;
683
684 if(len1)
685 for(ch=0; ch<preout->ch_count; ch++)
686 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);
687 if(out_count != len1)
688 for(ch=0; ch<preout->ch_count; ch++)
689 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, s->native_one, 0, 0, out_count - len1);
690 } else {
691 for(ch=0; ch<preout->ch_count; ch++)
692 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);
693 }
694 } else {
695 switch(s->int_sample_fmt) {
696 case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
697 case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
698 case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
699 case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
700 }
701 }
702 s->dither.noise_pos += out_count;
703 }
704 //FIXME packed doesn't need more than 1 chan here!
705 swri_audio_convert(s->out_convert, out, conv_src, out_count);
706 }
707 return out_count;
708 }
709
swr_is_initialized(struct SwrContext * s)710 int swr_is_initialized(struct SwrContext *s) {
711 return !!s->in_buffer.ch_count;
712 }
713
swr_convert(struct SwrContext * s,uint8_t * out_arg[SWR_CH_MAX],int out_count,const uint8_t * in_arg[SWR_CH_MAX],int in_count)714 int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
715 const uint8_t *in_arg [SWR_CH_MAX], int in_count){
716 AudioData * in= &s->in;
717 AudioData *out= &s->out;
718 int av_unused max_output;
719
720 if (!swr_is_initialized(s)) {
721 av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
722 return AVERROR(EINVAL);
723 }
724 #if defined(ASSERT_LEVEL) && ASSERT_LEVEL >1
725 max_output = swr_get_out_samples(s, in_count);
726 #endif
727
728 while(s->drop_output > 0){
729 int ret;
730 uint8_t *tmp_arg[SWR_CH_MAX];
731 #define MAX_DROP_STEP 16384
732 if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)
733 return ret;
734
735 reversefill_audiodata(&s->drop_temp, tmp_arg);
736 s->drop_output *= -1; //FIXME find a less hackish solution
737 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
738 s->drop_output *= -1;
739 in_count = 0;
740 if(ret>0) {
741 s->drop_output -= ret;
742 if (!s->drop_output && !out_arg)
743 return 0;
744 continue;
745 }
746
747 av_assert0(s->drop_output);
748 return 0;
749 }
750
751 if(!in_arg){
752 if(s->resample){
753 if (!s->flushed)
754 s->resampler->flush(s);
755 s->resample_in_constraint = 0;
756 s->flushed = 1;
757 }else if(!s->in_buffer_count){
758 return 0;
759 }
760 }else
761 fill_audiodata(in , (void*)in_arg);
762
763 fill_audiodata(out, out_arg);
764
765 if(s->resample){
766 int ret = swr_convert_internal(s, out, out_count, in, in_count);
767 if(ret>0 && !s->drop_output)
768 s->outpts += ret * (int64_t)s->in_sample_rate;
769
770 av_assert2(max_output < 0 || ret < 0 || ret <= max_output);
771
772 return ret;
773 }else{
774 AudioData tmp= *in;
775 int ret2=0;
776 int ret, size;
777 size = FFMIN(out_count, s->in_buffer_count);
778 if(size){
779 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
780 ret= swr_convert_internal(s, out, size, &tmp, size);
781 if(ret<0)
782 return ret;
783 ret2= ret;
784 s->in_buffer_count -= ret;
785 s->in_buffer_index += ret;
786 buf_set(out, out, ret);
787 out_count -= ret;
788 if(!s->in_buffer_count)
789 s->in_buffer_index = 0;
790 }
791
792 if(in_count){
793 size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
794
795 if(in_count > out_count) { //FIXME move after swr_convert_internal
796 if( size > s->in_buffer.count
797 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
798 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
799 copy(&s->in_buffer, &tmp, s->in_buffer_count);
800 s->in_buffer_index=0;
801 }else
802 if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
803 return ret;
804 }
805
806 if(out_count){
807 size = FFMIN(in_count, out_count);
808 ret= swr_convert_internal(s, out, size, in, size);
809 if(ret<0)
810 return ret;
811 buf_set(in, in, ret);
812 in_count -= ret;
813 ret2 += ret;
814 }
815 if(in_count){
816 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
817 copy(&tmp, in, in_count);
818 s->in_buffer_count += in_count;
819 }
820 }
821 if(ret2>0 && !s->drop_output)
822 s->outpts += ret2 * (int64_t)s->in_sample_rate;
823 av_assert2(max_output < 0 || ret2 < 0 || ret2 <= max_output);
824 return ret2;
825 }
826 }
827
swr_drop_output(struct SwrContext * s,int count)828 int swr_drop_output(struct SwrContext *s, int count){
829 const uint8_t *tmp_arg[SWR_CH_MAX];
830 s->drop_output += count;
831
832 if(s->drop_output <= 0)
833 return 0;
834
835 av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
836 return swr_convert(s, NULL, s->drop_output, tmp_arg, 0);
837 }
838
swr_inject_silence(struct SwrContext * s,int count)839 int swr_inject_silence(struct SwrContext *s, int count){
840 int ret, i;
841 uint8_t *tmp_arg[SWR_CH_MAX];
842
843 if(count <= 0)
844 return 0;
845
846 #define MAX_SILENCE_STEP 16384
847 while (count > MAX_SILENCE_STEP) {
848 if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
849 return ret;
850 count -= MAX_SILENCE_STEP;
851 }
852
853 if((ret=swri_realloc_audio(&s->silence, count))<0)
854 return ret;
855
856 if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
857 memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
858 } else
859 memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
860
861 reversefill_audiodata(&s->silence, tmp_arg);
862 av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
863 ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
864 return ret;
865 }
866
swr_get_delay(struct SwrContext * s,int64_t base)867 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
868 if (s->resampler && s->resample){
869 return s->resampler->get_delay(s, base);
870 }else{
871 return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
872 }
873 }
874
swr_get_out_samples(struct SwrContext * s,int in_samples)875 int swr_get_out_samples(struct SwrContext *s, int in_samples)
876 {
877 int64_t out_samples;
878
879 if (in_samples < 0)
880 return AVERROR(EINVAL);
881
882 if (s->resampler && s->resample) {
883 if (!s->resampler->get_out_samples)
884 return AVERROR(ENOSYS);
885 out_samples = s->resampler->get_out_samples(s, in_samples);
886 } else {
887 out_samples = s->in_buffer_count + in_samples;
888 av_assert0(s->out_sample_rate == s->in_sample_rate);
889 }
890
891 if (out_samples > INT_MAX)
892 return AVERROR(EINVAL);
893
894 return out_samples;
895 }
896
swr_set_compensation(struct SwrContext * s,int sample_delta,int compensation_distance)897 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
898 int ret;
899
900 if (!s || compensation_distance < 0)
901 return AVERROR(EINVAL);
902 if (!compensation_distance && sample_delta)
903 return AVERROR(EINVAL);
904 if (!s->resample) {
905 s->flags |= SWR_FLAG_RESAMPLE;
906 ret = swr_init(s);
907 if (ret < 0)
908 return ret;
909 }
910 if (!s->resampler->set_compensation){
911 return AVERROR(EINVAL);
912 }else{
913 return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
914 }
915 }
916
swr_next_pts(struct SwrContext * s,int64_t pts)917 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
918 if(pts == INT64_MIN)
919 return s->outpts;
920
921 if (s->firstpts == AV_NOPTS_VALUE)
922 s->outpts = s->firstpts = pts;
923
924 if(s->min_compensation >= FLT_MAX) {
925 return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
926 } else {
927 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;
928 double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
929
930 if(fabs(fdelta) > s->min_compensation) {
931 if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
932 int ret;
933 if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
934 else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
935 if(ret<0){
936 av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
937 }
938 } else if(s->soft_compensation_duration && s->max_soft_compensation) {
939 int duration = s->out_sample_rate * s->soft_compensation_duration;
940 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
941 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
942 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
943 swr_set_compensation(s, comp, duration);
944 }
945 }
946
947 return s->outpts;
948 }
949 }
950