blob: 74c96dce605e467a99def17607908cb651488e11
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, "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 | |
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 | |
109 | static void free_temp(AudioData *a){ |
110 | av_free(a->data); |
111 | memset(a, 0, sizeof(*a)); |
112 | } |
113 | |
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 | |
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 | |
148 | av_cold void swr_close(SwrContext *s){ |
149 | clear_context(s); |
150 | } |
151 | |
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 | s->out.ch_count = s-> user_out_ch_count; |
168 | s-> in.ch_count = s-> user_in_ch_count; |
169 | s->used_ch_count = s->user_used_ch_count; |
170 | |
171 | s-> in_ch_layout = s-> user_in_ch_layout; |
172 | s->out_ch_layout = s->user_out_ch_layout; |
173 | |
174 | s->int_sample_fmt= s->user_int_sample_fmt; |
175 | |
176 | s->dither.method = s->user_dither_method; |
177 | |
178 | if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) { |
179 | av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout); |
180 | s->in_ch_layout = 0; |
181 | } |
182 | |
183 | if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) { |
184 | av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout); |
185 | s->out_ch_layout = 0; |
186 | } |
187 | |
188 | switch(s->engine){ |
189 | #if CONFIG_LIBSOXR |
190 | case SWR_ENGINE_SOXR: s->resampler = &swri_soxr_resampler; break; |
191 | #endif |
192 | case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break; |
193 | default: |
194 | av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n"); |
195 | return AVERROR(EINVAL); |
196 | } |
197 | |
198 | if(!s->used_ch_count) |
199 | s->used_ch_count= s->in.ch_count; |
200 | |
201 | if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){ |
202 | av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n"); |
203 | s-> in_ch_layout= 0; |
204 | } |
205 | |
206 | if(!s-> in_ch_layout) |
207 | s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count); |
208 | if(!s->out_ch_layout) |
209 | s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count); |
210 | |
211 | s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 || |
212 | s->rematrix_custom; |
213 | |
214 | if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){ |
215 | if( av_get_bytes_per_sample(s-> in_sample_fmt) <= 2 |
216 | && av_get_bytes_per_sample(s->out_sample_fmt) <= 2){ |
217 | s->int_sample_fmt= AV_SAMPLE_FMT_S16P; |
218 | }else if( av_get_bytes_per_sample(s-> in_sample_fmt) <= 2 |
219 | && !s->rematrix |
220 | && s->out_sample_rate==s->in_sample_rate |
221 | && !(s->flags & SWR_FLAG_RESAMPLE)){ |
222 | s->int_sample_fmt= AV_SAMPLE_FMT_S16P; |
223 | }else if( av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P |
224 | && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P |
225 | && !s->rematrix |
226 | && s->out_sample_rate == s->in_sample_rate |
227 | && !(s->flags & SWR_FLAG_RESAMPLE) |
228 | && s->engine != SWR_ENGINE_SOXR){ |
229 | s->int_sample_fmt= AV_SAMPLE_FMT_S32P; |
230 | }else if(av_get_bytes_per_sample(s->in_sample_fmt) <= 4){ |
231 | s->int_sample_fmt= AV_SAMPLE_FMT_FLTP; |
232 | }else{ |
233 | s->int_sample_fmt= AV_SAMPLE_FMT_DBLP; |
234 | } |
235 | } |
236 | av_log(s, AV_LOG_DEBUG, "Using %s internally between filters\n", av_get_sample_fmt_name(s->int_sample_fmt)); |
237 | |
238 | if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P |
239 | &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P |
240 | &&s->int_sample_fmt != AV_SAMPLE_FMT_S64P |
241 | &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP |
242 | &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){ |
243 | av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, S16/S32/S64/FLT/DBL is supported\n", av_get_sample_fmt_name(s->int_sample_fmt)); |
244 | return AVERROR(EINVAL); |
245 | } |
246 | |
247 | set_audiodata_fmt(&s-> in, s-> in_sample_fmt); |
248 | set_audiodata_fmt(&s->out, s->out_sample_fmt); |
249 | |
250 | if (s->firstpts_in_samples != AV_NOPTS_VALUE) { |
251 | if (!s->async && s->min_compensation >= FLT_MAX/2) |
252 | s->async = 1; |
253 | s->firstpts = |
254 | s->outpts = s->firstpts_in_samples * s->out_sample_rate; |
255 | } else |
256 | s->firstpts = AV_NOPTS_VALUE; |
257 | |
258 | if (s->async) { |
259 | if (s->min_compensation >= FLT_MAX/2) |
260 | s->min_compensation = 0.001; |
261 | if (s->async > 1.0001) { |
262 | s->max_soft_compensation = s->async / (double) s->in_sample_rate; |
263 | } |
264 | } |
265 | |
266 | if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){ |
267 | 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); |
268 | if (!s->resample) { |
269 | av_log(s, AV_LOG_ERROR, "Failed to initialize resampler\n"); |
270 | return AVERROR(ENOMEM); |
271 | } |
272 | }else |
273 | s->resampler->free(&s->resample); |
274 | if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P |
275 | && s->int_sample_fmt != AV_SAMPLE_FMT_S32P |
276 | && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP |
277 | && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP |
278 | && s->resample){ |
279 | av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n"); |
280 | ret = AVERROR(EINVAL); |
281 | goto fail; |
282 | } |
283 | |
284 | #define RSC 1 //FIXME finetune |
285 | if(!s-> in.ch_count) |
286 | s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout); |
287 | if(!s->used_ch_count) |
288 | s->used_ch_count= s->in.ch_count; |
289 | if(!s->out.ch_count) |
290 | s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout); |
291 | |
292 | if(!s-> in.ch_count){ |
293 | av_assert0(!s->in_ch_layout); |
294 | av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n"); |
295 | ret = AVERROR(EINVAL); |
296 | goto fail; |
297 | } |
298 | |
299 | av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout); |
300 | av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout); |
301 | if (s->out_ch_layout && s->out.ch_count != av_get_channel_layout_nb_channels(s->out_ch_layout)) { |
302 | av_log(s, AV_LOG_ERROR, "Output channel layout %s mismatches specified channel count %d\n", l2, s->out.ch_count); |
303 | ret = AVERROR(EINVAL); |
304 | goto fail; |
305 | } |
306 | if (s->in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s->in_ch_layout)) { |
307 | av_log(s, AV_LOG_ERROR, "Input channel layout %s mismatches specified channel count %d\n", l1, s->used_ch_count); |
308 | ret = AVERROR(EINVAL); |
309 | goto fail; |
310 | } |
311 | |
312 | if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) { |
313 | av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s " |
314 | "but there is not enough information to do it\n", l1, l2); |
315 | ret = AVERROR(EINVAL); |
316 | goto fail; |
317 | } |
318 | |
319 | av_assert0(s->used_ch_count); |
320 | av_assert0(s->out.ch_count); |
321 | s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0; |
322 | |
323 | s->in_buffer= s->in; |
324 | s->silence = s->in; |
325 | s->drop_temp= s->out; |
326 | |
327 | if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0) |
328 | goto fail; |
329 | |
330 | if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){ |
331 | s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt, |
332 | s-> in_sample_fmt, s-> in.ch_count, NULL, 0); |
333 | return 0; |
334 | } |
335 | |
336 | s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt, |
337 | s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0); |
338 | s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt, |
339 | s->int_sample_fmt, s->out.ch_count, NULL, 0); |
340 | |
341 | if (!s->in_convert || !s->out_convert) { |
342 | ret = AVERROR(ENOMEM); |
343 | goto fail; |
344 | } |
345 | |
346 | s->postin= s->in; |
347 | s->preout= s->out; |
348 | s->midbuf= s->in; |
349 | |
350 | if(s->channel_map){ |
351 | s->postin.ch_count= |
352 | s->midbuf.ch_count= s->used_ch_count; |
353 | if(s->resample) |
354 | s->in_buffer.ch_count= s->used_ch_count; |
355 | } |
356 | if(!s->resample_first){ |
357 | s->midbuf.ch_count= s->out.ch_count; |
358 | if(s->resample) |
359 | s->in_buffer.ch_count = s->out.ch_count; |
360 | } |
361 | |
362 | set_audiodata_fmt(&s->postin, s->int_sample_fmt); |
363 | set_audiodata_fmt(&s->midbuf, s->int_sample_fmt); |
364 | set_audiodata_fmt(&s->preout, s->int_sample_fmt); |
365 | |
366 | if(s->resample){ |
367 | set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt); |
368 | } |
369 | |
370 | av_assert0(!s->preout.count); |
371 | s->dither.noise = s->preout; |
372 | s->dither.temp = s->preout; |
373 | if (s->dither.method > SWR_DITHER_NS) { |
374 | s->dither.noise.bps = 4; |
375 | s->dither.noise.fmt = AV_SAMPLE_FMT_FLTP; |
376 | s->dither.noise_scale = 1; |
377 | } |
378 | |
379 | if(s->rematrix || s->dither.method) { |
380 | ret = swri_rematrix_init(s); |
381 | if (ret < 0) |
382 | goto fail; |
383 | } |
384 | |
385 | return 0; |
386 | fail: |
387 | swr_close(s); |
388 | return ret; |
389 | |
390 | } |
391 | |
392 | int swri_realloc_audio(AudioData *a, int count){ |
393 | int i, countb; |
394 | AudioData old; |
395 | |
396 | if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count) |
397 | return AVERROR(EINVAL); |
398 | |
399 | if(a->count >= count) |
400 | return 0; |
401 | |
402 | count*=2; |
403 | |
404 | countb= FFALIGN(count*a->bps, ALIGN); |
405 | old= *a; |
406 | |
407 | av_assert0(a->bps); |
408 | av_assert0(a->ch_count); |
409 | |
410 | a->data= av_mallocz_array(countb, a->ch_count); |
411 | if(!a->data) |
412 | return AVERROR(ENOMEM); |
413 | for(i=0; i<a->ch_count; i++){ |
414 | a->ch[i]= a->data + i*(a->planar ? countb : a->bps); |
415 | if(a->count && a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps); |
416 | } |
417 | if(a->count && !a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps); |
418 | av_freep(&old.data); |
419 | a->count= count; |
420 | |
421 | return 1; |
422 | } |
423 | |
424 | static void copy(AudioData *out, AudioData *in, |
425 | int count){ |
426 | av_assert0(out->planar == in->planar); |
427 | av_assert0(out->bps == in->bps); |
428 | av_assert0(out->ch_count == in->ch_count); |
429 | if(out->planar){ |
430 | int ch; |
431 | for(ch=0; ch<out->ch_count; ch++) |
432 | memcpy(out->ch[ch], in->ch[ch], count*out->bps); |
433 | }else |
434 | memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps); |
435 | } |
436 | |
437 | static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){ |
438 | int i; |
439 | if(!in_arg){ |
440 | memset(out->ch, 0, sizeof(out->ch)); |
441 | }else if(out->planar){ |
442 | for(i=0; i<out->ch_count; i++) |
443 | out->ch[i]= in_arg[i]; |
444 | }else{ |
445 | for(i=0; i<out->ch_count; i++) |
446 | out->ch[i]= in_arg[0] + i*out->bps; |
447 | } |
448 | } |
449 | |
450 | static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){ |
451 | int i; |
452 | if(out->planar){ |
453 | for(i=0; i<out->ch_count; i++) |
454 | in_arg[i]= out->ch[i]; |
455 | }else{ |
456 | in_arg[0]= out->ch[0]; |
457 | } |
458 | } |
459 | |
460 | /** |
461 | * |
462 | * out may be equal in. |
463 | */ |
464 | static void buf_set(AudioData *out, AudioData *in, int count){ |
465 | int ch; |
466 | if(in->planar){ |
467 | for(ch=0; ch<out->ch_count; ch++) |
468 | out->ch[ch]= in->ch[ch] + count*out->bps; |
469 | }else{ |
470 | for(ch=out->ch_count-1; ch>=0; ch--) |
471 | out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps; |
472 | } |
473 | } |
474 | |
475 | /** |
476 | * |
477 | * @return number of samples output per channel |
478 | */ |
479 | static int resample(SwrContext *s, AudioData *out_param, int out_count, |
480 | const AudioData * in_param, int in_count){ |
481 | AudioData in, out, tmp; |
482 | int ret_sum=0; |
483 | int border=0; |
484 | int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0; |
485 | |
486 | av_assert1(s->in_buffer.ch_count == in_param->ch_count); |
487 | av_assert1(s->in_buffer.planar == in_param->planar); |
488 | av_assert1(s->in_buffer.fmt == in_param->fmt); |
489 | |
490 | tmp=out=*out_param; |
491 | in = *in_param; |
492 | |
493 | border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer, |
494 | &in, in_count, &s->in_buffer_index, &s->in_buffer_count); |
495 | if (border == INT_MAX) { |
496 | return 0; |
497 | } else if (border < 0) { |
498 | return border; |
499 | } else if (border) { |
500 | buf_set(&in, &in, border); |
501 | in_count -= border; |
502 | s->resample_in_constraint = 0; |
503 | } |
504 | |
505 | do{ |
506 | int ret, size, consumed; |
507 | if(!s->resample_in_constraint && s->in_buffer_count){ |
508 | buf_set(&tmp, &s->in_buffer, s->in_buffer_index); |
509 | ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed); |
510 | out_count -= ret; |
511 | ret_sum += ret; |
512 | buf_set(&out, &out, ret); |
513 | s->in_buffer_count -= consumed; |
514 | s->in_buffer_index += consumed; |
515 | |
516 | if(!in_count) |
517 | break; |
518 | if(s->in_buffer_count <= border){ |
519 | buf_set(&in, &in, -s->in_buffer_count); |
520 | in_count += s->in_buffer_count; |
521 | s->in_buffer_count=0; |
522 | s->in_buffer_index=0; |
523 | border = 0; |
524 | } |
525 | } |
526 | |
527 | if((s->flushed || in_count > padless) && !s->in_buffer_count){ |
528 | s->in_buffer_index=0; |
529 | ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed); |
530 | out_count -= ret; |
531 | ret_sum += ret; |
532 | buf_set(&out, &out, ret); |
533 | in_count -= consumed; |
534 | buf_set(&in, &in, consumed); |
535 | } |
536 | |
537 | //TODO is this check sane considering the advanced copy avoidance below |
538 | size= s->in_buffer_index + s->in_buffer_count + in_count; |
539 | if( size > s->in_buffer.count |
540 | && s->in_buffer_count + in_count <= s->in_buffer_index){ |
541 | buf_set(&tmp, &s->in_buffer, s->in_buffer_index); |
542 | copy(&s->in_buffer, &tmp, s->in_buffer_count); |
543 | s->in_buffer_index=0; |
544 | }else |
545 | if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0) |
546 | return ret; |
547 | |
548 | if(in_count){ |
549 | int count= in_count; |
550 | if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2; |
551 | |
552 | buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count); |
553 | copy(&tmp, &in, /*in_*/count); |
554 | s->in_buffer_count += count; |
555 | in_count -= count; |
556 | border += count; |
557 | buf_set(&in, &in, count); |
558 | s->resample_in_constraint= 0; |
559 | if(s->in_buffer_count != count || in_count) |
560 | continue; |
561 | if (padless) { |
562 | padless = 0; |
563 | continue; |
564 | } |
565 | } |
566 | break; |
567 | }while(1); |
568 | |
569 | s->resample_in_constraint= !!out_count; |
570 | |
571 | return ret_sum; |
572 | } |
573 | |
574 | static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count, |
575 | AudioData *in , int in_count){ |
576 | AudioData *postin, *midbuf, *preout; |
577 | int ret/*, in_max*/; |
578 | AudioData preout_tmp, midbuf_tmp; |
579 | |
580 | if(s->full_convert){ |
581 | av_assert0(!s->resample); |
582 | swri_audio_convert(s->full_convert, out, in, in_count); |
583 | return out_count; |
584 | } |
585 | |
586 | // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps; |
587 | // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count); |
588 | |
589 | if((ret=swri_realloc_audio(&s->postin, in_count))<0) |
590 | return ret; |
591 | if(s->resample_first){ |
592 | av_assert0(s->midbuf.ch_count == s->used_ch_count); |
593 | if((ret=swri_realloc_audio(&s->midbuf, out_count))<0) |
594 | return ret; |
595 | }else{ |
596 | av_assert0(s->midbuf.ch_count == s->out.ch_count); |
597 | if((ret=swri_realloc_audio(&s->midbuf, in_count))<0) |
598 | return ret; |
599 | } |
600 | if((ret=swri_realloc_audio(&s->preout, out_count))<0) |
601 | return ret; |
602 | |
603 | postin= &s->postin; |
604 | |
605 | midbuf_tmp= s->midbuf; |
606 | midbuf= &midbuf_tmp; |
607 | preout_tmp= s->preout; |
608 | preout= &preout_tmp; |
609 | |
610 | if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map) |
611 | postin= in; |
612 | |
613 | if(s->resample_first ? !s->resample : !s->rematrix) |
614 | midbuf= postin; |
615 | |
616 | if(s->resample_first ? !s->rematrix : !s->resample) |
617 | preout= midbuf; |
618 | |
619 | if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar |
620 | && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){ |
621 | if(preout==in){ |
622 | out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant |
623 | av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though |
624 | copy(out, in, out_count); |
625 | return out_count; |
626 | } |
627 | else if(preout==postin) preout= midbuf= postin= out; |
628 | else if(preout==midbuf) preout= midbuf= out; |
629 | else preout= out; |
630 | } |
631 | |
632 | if(in != postin){ |
633 | swri_audio_convert(s->in_convert, postin, in, in_count); |
634 | } |
635 | |
636 | if(s->resample_first){ |
637 | if(postin != midbuf) |
638 | out_count= resample(s, midbuf, out_count, postin, in_count); |
639 | if(midbuf != preout) |
640 | swri_rematrix(s, preout, midbuf, out_count, preout==out); |
641 | }else{ |
642 | if(postin != midbuf) |
643 | swri_rematrix(s, midbuf, postin, in_count, midbuf==out); |
644 | if(midbuf != preout) |
645 | out_count= resample(s, preout, out_count, midbuf, in_count); |
646 | } |
647 | |
648 | if(preout != out && out_count){ |
649 | AudioData *conv_src = preout; |
650 | if(s->dither.method){ |
651 | int ch; |
652 | int dither_count= FFMAX(out_count, 1<<16); |
653 | |
654 | if (preout == in) { |
655 | conv_src = &s->dither.temp; |
656 | if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0) |
657 | return ret; |
658 | } |
659 | |
660 | if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0) |
661 | return ret; |
662 | if(ret) |
663 | for(ch=0; ch<s->dither.noise.ch_count; ch++) |
664 | if((ret=swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, (12345678913579ULL*ch + 3141592) % 2718281828U, s->dither.noise.fmt))<0) |
665 | return ret; |
666 | av_assert0(s->dither.noise.ch_count == preout->ch_count); |
667 | |
668 | if(s->dither.noise_pos + out_count > s->dither.noise.count) |
669 | s->dither.noise_pos = 0; |
670 | |
671 | if (s->dither.method < SWR_DITHER_NS){ |
672 | if (s->mix_2_1_simd) { |
673 | int len1= out_count&~15; |
674 | int off = len1 * preout->bps; |
675 | |
676 | if(len1) |
677 | for(ch=0; ch<preout->ch_count; ch++) |
678 | 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); |
679 | if(out_count != len1) |
680 | for(ch=0; ch<preout->ch_count; ch++) |
681 | 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); |
682 | } else { |
683 | for(ch=0; ch<preout->ch_count; ch++) |
684 | 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); |
685 | } |
686 | } else { |
687 | switch(s->int_sample_fmt) { |
688 | case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break; |
689 | case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break; |
690 | case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break; |
691 | case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break; |
692 | } |
693 | } |
694 | s->dither.noise_pos += out_count; |
695 | } |
696 | //FIXME packed doesn't need more than 1 chan here! |
697 | swri_audio_convert(s->out_convert, out, conv_src, out_count); |
698 | } |
699 | return out_count; |
700 | } |
701 | |
702 | int swr_is_initialized(struct SwrContext *s) { |
703 | return !!s->in_buffer.ch_count; |
704 | } |
705 | |
706 | int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count, |
707 | const uint8_t *in_arg [SWR_CH_MAX], int in_count){ |
708 | AudioData * in= &s->in; |
709 | AudioData *out= &s->out; |
710 | int av_unused max_output; |
711 | |
712 | if (!swr_is_initialized(s)) { |
713 | av_log(s, AV_LOG_ERROR, "Context has not been initialized\n"); |
714 | return AVERROR(EINVAL); |
715 | } |
716 | #if defined(ASSERT_LEVEL) && ASSERT_LEVEL >1 |
717 | max_output = swr_get_out_samples(s, in_count); |
718 | #endif |
719 | |
720 | while(s->drop_output > 0){ |
721 | int ret; |
722 | uint8_t *tmp_arg[SWR_CH_MAX]; |
723 | #define MAX_DROP_STEP 16384 |
724 | if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0) |
725 | return ret; |
726 | |
727 | reversefill_audiodata(&s->drop_temp, tmp_arg); |
728 | s->drop_output *= -1; //FIXME find a less hackish solution |
729 | 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 |
730 | s->drop_output *= -1; |
731 | in_count = 0; |
732 | if(ret>0) { |
733 | s->drop_output -= ret; |
734 | if (!s->drop_output && !out_arg) |
735 | return 0; |
736 | continue; |
737 | } |
738 | |
739 | av_assert0(s->drop_output); |
740 | return 0; |
741 | } |
742 | |
743 | if(!in_arg){ |
744 | if(s->resample){ |
745 | if (!s->flushed) |
746 | s->resampler->flush(s); |
747 | s->resample_in_constraint = 0; |
748 | s->flushed = 1; |
749 | }else if(!s->in_buffer_count){ |
750 | return 0; |
751 | } |
752 | }else |
753 | fill_audiodata(in , (void*)in_arg); |
754 | |
755 | fill_audiodata(out, out_arg); |
756 | |
757 | if(s->resample){ |
758 | int ret = swr_convert_internal(s, out, out_count, in, in_count); |
759 | if(ret>0 && !s->drop_output) |
760 | s->outpts += ret * (int64_t)s->in_sample_rate; |
761 | |
762 | av_assert2(max_output < 0 || ret < 0 || ret <= max_output); |
763 | |
764 | return ret; |
765 | }else{ |
766 | AudioData tmp= *in; |
767 | int ret2=0; |
768 | int ret, size; |
769 | size = FFMIN(out_count, s->in_buffer_count); |
770 | if(size){ |
771 | buf_set(&tmp, &s->in_buffer, s->in_buffer_index); |
772 | ret= swr_convert_internal(s, out, size, &tmp, size); |
773 | if(ret<0) |
774 | return ret; |
775 | ret2= ret; |
776 | s->in_buffer_count -= ret; |
777 | s->in_buffer_index += ret; |
778 | buf_set(out, out, ret); |
779 | out_count -= ret; |
780 | if(!s->in_buffer_count) |
781 | s->in_buffer_index = 0; |
782 | } |
783 | |
784 | if(in_count){ |
785 | size= s->in_buffer_index + s->in_buffer_count + in_count - out_count; |
786 | |
787 | if(in_count > out_count) { //FIXME move after swr_convert_internal |
788 | if( size > s->in_buffer.count |
789 | && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){ |
790 | buf_set(&tmp, &s->in_buffer, s->in_buffer_index); |
791 | copy(&s->in_buffer, &tmp, s->in_buffer_count); |
792 | s->in_buffer_index=0; |
793 | }else |
794 | if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0) |
795 | return ret; |
796 | } |
797 | |
798 | if(out_count){ |
799 | size = FFMIN(in_count, out_count); |
800 | ret= swr_convert_internal(s, out, size, in, size); |
801 | if(ret<0) |
802 | return ret; |
803 | buf_set(in, in, ret); |
804 | in_count -= ret; |
805 | ret2 += ret; |
806 | } |
807 | if(in_count){ |
808 | buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count); |
809 | copy(&tmp, in, in_count); |
810 | s->in_buffer_count += in_count; |
811 | } |
812 | } |
813 | if(ret2>0 && !s->drop_output) |
814 | s->outpts += ret2 * (int64_t)s->in_sample_rate; |
815 | av_assert2(max_output < 0 || ret2 < 0 || ret2 <= max_output); |
816 | return ret2; |
817 | } |
818 | } |
819 | |
820 | int swr_drop_output(struct SwrContext *s, int count){ |
821 | const uint8_t *tmp_arg[SWR_CH_MAX]; |
822 | s->drop_output += count; |
823 | |
824 | if(s->drop_output <= 0) |
825 | return 0; |
826 | |
827 | av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count); |
828 | return swr_convert(s, NULL, s->drop_output, tmp_arg, 0); |
829 | } |
830 | |
831 | int swr_inject_silence(struct SwrContext *s, int count){ |
832 | int ret, i; |
833 | uint8_t *tmp_arg[SWR_CH_MAX]; |
834 | |
835 | if(count <= 0) |
836 | return 0; |
837 | |
838 | #define MAX_SILENCE_STEP 16384 |
839 | while (count > MAX_SILENCE_STEP) { |
840 | if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0) |
841 | return ret; |
842 | count -= MAX_SILENCE_STEP; |
843 | } |
844 | |
845 | if((ret=swri_realloc_audio(&s->silence, count))<0) |
846 | return ret; |
847 | |
848 | if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) { |
849 | memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps); |
850 | } else |
851 | memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count); |
852 | |
853 | reversefill_audiodata(&s->silence, tmp_arg); |
854 | av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count); |
855 | ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count); |
856 | return ret; |
857 | } |
858 | |
859 | int64_t swr_get_delay(struct SwrContext *s, int64_t base){ |
860 | if (s->resampler && s->resample){ |
861 | return s->resampler->get_delay(s, base); |
862 | }else{ |
863 | return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate; |
864 | } |
865 | } |
866 | |
867 | int swr_get_out_samples(struct SwrContext *s, int in_samples) |
868 | { |
869 | int64_t out_samples; |
870 | |
871 | if (in_samples < 0) |
872 | return AVERROR(EINVAL); |
873 | |
874 | if (s->resampler && s->resample) { |
875 | if (!s->resampler->get_out_samples) |
876 | return AVERROR(ENOSYS); |
877 | out_samples = s->resampler->get_out_samples(s, in_samples); |
878 | } else { |
879 | out_samples = s->in_buffer_count + in_samples; |
880 | av_assert0(s->out_sample_rate == s->in_sample_rate); |
881 | } |
882 | |
883 | if (out_samples > INT_MAX) |
884 | return AVERROR(EINVAL); |
885 | |
886 | return out_samples; |
887 | } |
888 | |
889 | int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){ |
890 | int ret; |
891 | |
892 | if (!s || compensation_distance < 0) |
893 | return AVERROR(EINVAL); |
894 | if (!compensation_distance && sample_delta) |
895 | return AVERROR(EINVAL); |
896 | if (!s->resample) { |
897 | s->flags |= SWR_FLAG_RESAMPLE; |
898 | ret = swr_init(s); |
899 | if (ret < 0) |
900 | return ret; |
901 | } |
902 | if (!s->resampler->set_compensation){ |
903 | return AVERROR(EINVAL); |
904 | }else{ |
905 | return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance); |
906 | } |
907 | } |
908 | |
909 | int64_t swr_next_pts(struct SwrContext *s, int64_t pts){ |
910 | if(pts == INT64_MIN) |
911 | return s->outpts; |
912 | |
913 | if (s->firstpts == AV_NOPTS_VALUE) |
914 | s->outpts = s->firstpts = pts; |
915 | |
916 | if(s->min_compensation >= FLT_MAX) { |
917 | return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate)); |
918 | } else { |
919 | 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; |
920 | double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate); |
921 | |
922 | if(fabs(fdelta) > s->min_compensation) { |
923 | if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){ |
924 | int ret; |
925 | if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate); |
926 | else ret = swr_drop_output (s, -delta / s-> in_sample_rate); |
927 | if(ret<0){ |
928 | av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta); |
929 | } |
930 | } else if(s->soft_compensation_duration && s->max_soft_compensation) { |
931 | int duration = s->out_sample_rate * s->soft_compensation_duration; |
932 | double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1); |
933 | int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ; |
934 | av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration); |
935 | swr_set_compensation(s, comp, duration); |
936 | } |
937 | } |
938 | |
939 | return s->outpts; |
940 | } |
941 | } |
942 |