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1 | /* |
2 | * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at) | |
3 | * | |
4 | * This file is part of libswresample | |
5 | * | |
6 | * libswresample is free software; you can redistribute it and/or | |
7 | * modify it under the terms of the GNU Lesser General Public | |
8 | * License as published by the Free Software Foundation; either | |
9 | * version 2.1 of the License, or (at your option) any later version. | |
10 | * | |
11 | * libswresample is distributed in the hope that it will be useful, | |
12 | * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
13 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
14 | * Lesser General Public License for more details. | |
15 | * | |
16 | * You should have received a copy of the GNU Lesser General Public | |
17 | * License along with libswresample; if not, write to the Free Software | |
18 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | |
19 | */ | |
20 | ||
21 | #include "libavutil/opt.h" | |
22 | #include "swresample_internal.h" | |
23 | #include "audioconvert.h" | |
24 | #include "libavutil/avassert.h" | |
25 | #include "libavutil/channel_layout.h" | |
26 | ||
27 | #include <float.h> | |
28 | ||
29 | #define ALIGN 32 | |
30 | ||
31 | unsigned swresample_version(void) | |
32 | { | |
33 | av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100); | |
34 | return LIBSWRESAMPLE_VERSION_INT; | |
35 | } | |
36 | ||
37 | const char *swresample_configuration(void) | |
38 | { | |
39 | return FFMPEG_CONFIGURATION; | |
40 | } | |
41 | ||
42 | const char *swresample_license(void) | |
43 | { | |
44 | #define LICENSE_PREFIX "libswresample license: " | |
45 | return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1; | |
46 | } | |
47 | ||
48 | int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){ | |
49 | if(!s || s->in_convert) // s needs to be allocated but not initialized | |
50 | return AVERROR(EINVAL); | |
51 | s->channel_map = channel_map; | |
52 | return 0; | |
53 | } | |
54 | ||
55 | struct SwrContext *swr_alloc_set_opts(struct SwrContext *s, | |
56 | int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate, | |
57 | int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate, | |
58 | int log_offset, void *log_ctx){ | |
59 | if(!s) s= swr_alloc(); | |
60 | if(!s) return NULL; | |
61 | ||
62 | s->log_level_offset= log_offset; | |
63 | s->log_ctx= log_ctx; | |
64 | ||
65 | if (av_opt_set_int(s, "ocl", out_ch_layout, 0) < 0) | |
66 | goto fail; | |
67 | ||
68 | if (av_opt_set_int(s, "osf", out_sample_fmt, 0) < 0) | |
69 | goto fail; | |
70 | ||
71 | if (av_opt_set_int(s, "osr", out_sample_rate, 0) < 0) | |
72 | goto fail; | |
73 | ||
74 | if (av_opt_set_int(s, "icl", in_ch_layout, 0) < 0) | |
75 | goto fail; | |
76 | ||
77 | if (av_opt_set_int(s, "isf", in_sample_fmt, 0) < 0) | |
78 | goto fail; | |
79 | ||
80 | if (av_opt_set_int(s, "isr", in_sample_rate, 0) < 0) | |
81 | goto fail; | |
82 | ||
83 | if (av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE, 0) < 0) | |
84 | goto fail; | |
85 | ||
86 | if (av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0) < 0) | |
87 | goto fail; | |
88 | ||
89 | if (av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0) < 0) | |
90 | goto fail; | |
91 | ||
92 | av_opt_set_int(s, "uch", 0, 0); | |
93 | return s; | |
94 | fail: | |
95 | av_log(s, AV_LOG_ERROR, "Failed to set option\n"); | |
96 | swr_free(&s); | |
97 | return NULL; | |
98 | } | |
99 | ||
100 | static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){ | |
101 | a->fmt = fmt; | |
102 | a->bps = av_get_bytes_per_sample(fmt); | |
103 | a->planar= av_sample_fmt_is_planar(fmt); | |
104 | if (a->ch_count == 1) | |
105 | a->planar = 1; | |
106 | } | |
107 | ||
108 | static void free_temp(AudioData *a){ | |
109 | av_free(a->data); | |
110 | memset(a, 0, sizeof(*a)); | |
111 | } | |
112 | ||
113 | static void clear_context(SwrContext *s){ | |
114 | s->in_buffer_index= 0; | |
115 | s->in_buffer_count= 0; | |
116 | s->resample_in_constraint= 0; | |
117 | memset(s->in.ch, 0, sizeof(s->in.ch)); | |
118 | memset(s->out.ch, 0, sizeof(s->out.ch)); | |
119 | free_temp(&s->postin); | |
120 | free_temp(&s->midbuf); | |
121 | free_temp(&s->preout); | |
122 | free_temp(&s->in_buffer); | |
123 | free_temp(&s->silence); | |
124 | free_temp(&s->drop_temp); | |
125 | free_temp(&s->dither.noise); | |
126 | free_temp(&s->dither.temp); | |
127 | swri_audio_convert_free(&s-> in_convert); | |
128 | swri_audio_convert_free(&s->out_convert); | |
129 | swri_audio_convert_free(&s->full_convert); | |
130 | swri_rematrix_free(s); | |
131 | ||
132 | s->flushed = 0; | |
133 | } | |
134 | ||
135 | av_cold void swr_free(SwrContext **ss){ | |
136 | SwrContext *s= *ss; | |
137 | if(s){ | |
138 | clear_context(s); | |
139 | if (s->resampler) | |
140 | s->resampler->free(&s->resample); | |
141 | } | |
142 | ||
143 | av_freep(ss); | |
144 | } | |
145 | ||
146 | av_cold void swr_close(SwrContext *s){ | |
147 | clear_context(s); | |
148 | } | |
149 | ||
150 | av_cold int swr_init(struct SwrContext *s){ | |
151 | int ret; | |
152 | ||
153 | clear_context(s); | |
154 | ||
155 | if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){ | |
156 | av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt); | |
157 | return AVERROR(EINVAL); | |
158 | } | |
159 | if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){ | |
160 | av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt); | |
161 | return AVERROR(EINVAL); | |
162 | } | |
163 | ||
164 | if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) { | |
165 | av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout); | |
166 | s->in_ch_layout = 0; | |
167 | } | |
168 | ||
169 | if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) { | |
170 | av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout); | |
171 | s->out_ch_layout = 0; | |
172 | } | |
173 | ||
174 | switch(s->engine){ | |
175 | #if CONFIG_LIBSOXR | |
176 | extern struct Resampler const soxr_resampler; | |
177 | case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break; | |
178 | #endif | |
179 | case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break; | |
180 | default: | |
181 | av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n"); | |
182 | return AVERROR(EINVAL); | |
183 | } | |
184 | ||
185 | if(!s->used_ch_count) | |
186 | s->used_ch_count= s->in.ch_count; | |
187 | ||
188 | if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){ | |
189 | av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n"); | |
190 | s-> in_ch_layout= 0; | |
191 | } | |
192 | ||
193 | if(!s-> in_ch_layout) | |
194 | s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count); | |
195 | if(!s->out_ch_layout) | |
196 | s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count); | |
197 | ||
198 | s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 || | |
199 | s->rematrix_custom; | |
200 | ||
201 | if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){ | |
202 | if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){ | |
203 | s->int_sample_fmt= AV_SAMPLE_FMT_S16P; | |
204 | }else if( av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P | |
205 | && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P | |
206 | && !s->rematrix | |
207 | && s->engine != SWR_ENGINE_SOXR){ | |
208 | s->int_sample_fmt= AV_SAMPLE_FMT_S32P; | |
209 | }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){ | |
210 | s->int_sample_fmt= AV_SAMPLE_FMT_FLTP; | |
211 | }else{ | |
212 | av_log(s, AV_LOG_DEBUG, "Using double precision mode\n"); | |
213 | s->int_sample_fmt= AV_SAMPLE_FMT_DBLP; | |
214 | } | |
215 | } | |
216 | ||
217 | if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P | |
218 | &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P | |
219 | &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP | |
220 | &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){ | |
221 | 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)); | |
222 | return AVERROR(EINVAL); | |
223 | } | |
224 | ||
225 | set_audiodata_fmt(&s-> in, s-> in_sample_fmt); | |
226 | set_audiodata_fmt(&s->out, s->out_sample_fmt); | |
227 | ||
228 | if (s->firstpts_in_samples != AV_NOPTS_VALUE) { | |
229 | if (!s->async && s->min_compensation >= FLT_MAX/2) | |
230 | s->async = 1; | |
231 | s->firstpts = | |
232 | s->outpts = s->firstpts_in_samples * s->out_sample_rate; | |
233 | } else | |
234 | s->firstpts = AV_NOPTS_VALUE; | |
235 | ||
236 | if (s->async) { | |
237 | if (s->min_compensation >= FLT_MAX/2) | |
238 | s->min_compensation = 0.001; | |
239 | if (s->async > 1.0001) { | |
240 | s->max_soft_compensation = s->async / (double) s->in_sample_rate; | |
241 | } | |
242 | } | |
243 | ||
244 | if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){ | |
245 | 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); | |
246 | }else | |
247 | s->resampler->free(&s->resample); | |
248 | if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P | |
249 | && s->int_sample_fmt != AV_SAMPLE_FMT_S32P | |
250 | && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP | |
251 | && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP | |
252 | && s->resample){ | |
253 | av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n"); | |
254 | return -1; | |
255 | } | |
256 | ||
257 | #define RSC 1 //FIXME finetune | |
258 | if(!s-> in.ch_count) | |
259 | s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout); | |
260 | if(!s->used_ch_count) | |
261 | s->used_ch_count= s->in.ch_count; | |
262 | if(!s->out.ch_count) | |
263 | s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout); | |
264 | ||
265 | if(!s-> in.ch_count){ | |
266 | av_assert0(!s->in_ch_layout); | |
267 | av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n"); | |
268 | return -1; | |
269 | } | |
270 | ||
271 | if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) { | |
272 | char l1[1024], l2[1024]; | |
273 | av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout); | |
274 | av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout); | |
275 | av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s " | |
276 | "but there is not enough information to do it\n", l1, l2); | |
277 | return -1; | |
278 | } | |
279 | ||
280 | av_assert0(s->used_ch_count); | |
281 | av_assert0(s->out.ch_count); | |
282 | s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0; | |
283 | ||
284 | s->in_buffer= s->in; | |
285 | s->silence = s->in; | |
286 | s->drop_temp= s->out; | |
287 | ||
288 | if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){ | |
289 | s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt, | |
290 | s-> in_sample_fmt, s-> in.ch_count, NULL, 0); | |
291 | return 0; | |
292 | } | |
293 | ||
294 | s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt, | |
295 | s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0); | |
296 | s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt, | |
297 | s->int_sample_fmt, s->out.ch_count, NULL, 0); | |
298 | ||
299 | if (!s->in_convert || !s->out_convert) | |
300 | return AVERROR(ENOMEM); | |
301 | ||
302 | s->postin= s->in; | |
303 | s->preout= s->out; | |
304 | s->midbuf= s->in; | |
305 | ||
306 | if(s->channel_map){ | |
307 | s->postin.ch_count= | |
308 | s->midbuf.ch_count= s->used_ch_count; | |
309 | if(s->resample) | |
310 | s->in_buffer.ch_count= s->used_ch_count; | |
311 | } | |
312 | if(!s->resample_first){ | |
313 | s->midbuf.ch_count= s->out.ch_count; | |
314 | if(s->resample) | |
315 | s->in_buffer.ch_count = s->out.ch_count; | |
316 | } | |
317 | ||
318 | set_audiodata_fmt(&s->postin, s->int_sample_fmt); | |
319 | set_audiodata_fmt(&s->midbuf, s->int_sample_fmt); | |
320 | set_audiodata_fmt(&s->preout, s->int_sample_fmt); | |
321 | ||
322 | if(s->resample){ | |
323 | set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt); | |
324 | } | |
325 | ||
326 | if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0) | |
327 | return ret; | |
328 | ||
329 | if(s->rematrix || s->dither.method) | |
330 | return swri_rematrix_init(s); | |
331 | ||
332 | return 0; | |
333 | } | |
334 | ||
335 | int swri_realloc_audio(AudioData *a, int count){ | |
336 | int i, countb; | |
337 | AudioData old; | |
338 | ||
339 | if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count) | |
340 | return AVERROR(EINVAL); | |
341 | ||
342 | if(a->count >= count) | |
343 | return 0; | |
344 | ||
345 | count*=2; | |
346 | ||
347 | countb= FFALIGN(count*a->bps, ALIGN); | |
348 | old= *a; | |
349 | ||
350 | av_assert0(a->bps); | |
351 | av_assert0(a->ch_count); | |
352 | ||
353 | a->data= av_mallocz(countb*a->ch_count); | |
354 | if(!a->data) | |
355 | return AVERROR(ENOMEM); | |
356 | for(i=0; i<a->ch_count; i++){ | |
357 | a->ch[i]= a->data + i*(a->planar ? countb : a->bps); | |
358 | if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps); | |
359 | } | |
360 | if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps); | |
361 | av_freep(&old.data); | |
362 | a->count= count; | |
363 | ||
364 | return 1; | |
365 | } | |
366 | ||
367 | static void copy(AudioData *out, AudioData *in, | |
368 | int count){ | |
369 | av_assert0(out->planar == in->planar); | |
370 | av_assert0(out->bps == in->bps); | |
371 | av_assert0(out->ch_count == in->ch_count); | |
372 | if(out->planar){ | |
373 | int ch; | |
374 | for(ch=0; ch<out->ch_count; ch++) | |
375 | memcpy(out->ch[ch], in->ch[ch], count*out->bps); | |
376 | }else | |
377 | memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps); | |
378 | } | |
379 | ||
380 | static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){ | |
381 | int i; | |
382 | if(!in_arg){ | |
383 | memset(out->ch, 0, sizeof(out->ch)); | |
384 | }else if(out->planar){ | |
385 | for(i=0; i<out->ch_count; i++) | |
386 | out->ch[i]= in_arg[i]; | |
387 | }else{ | |
388 | for(i=0; i<out->ch_count; i++) | |
389 | out->ch[i]= in_arg[0] + i*out->bps; | |
390 | } | |
391 | } | |
392 | ||
393 | static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){ | |
394 | int i; | |
395 | if(out->planar){ | |
396 | for(i=0; i<out->ch_count; i++) | |
397 | in_arg[i]= out->ch[i]; | |
398 | }else{ | |
399 | in_arg[0]= out->ch[0]; | |
400 | } | |
401 | } | |
402 | ||
403 | /** | |
404 | * | |
405 | * out may be equal in. | |
406 | */ | |
407 | static void buf_set(AudioData *out, AudioData *in, int count){ | |
408 | int ch; | |
409 | if(in->planar){ | |
410 | for(ch=0; ch<out->ch_count; ch++) | |
411 | out->ch[ch]= in->ch[ch] + count*out->bps; | |
412 | }else{ | |
413 | for(ch=out->ch_count-1; ch>=0; ch--) | |
414 | out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps; | |
415 | } | |
416 | } | |
417 | ||
418 | /** | |
419 | * | |
420 | * @return number of samples output per channel | |
421 | */ | |
422 | static int resample(SwrContext *s, AudioData *out_param, int out_count, | |
423 | const AudioData * in_param, int in_count){ | |
424 | AudioData in, out, tmp; | |
425 | int ret_sum=0; | |
426 | int border=0; | |
427 | int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0; | |
428 | ||
429 | av_assert1(s->in_buffer.ch_count == in_param->ch_count); | |
430 | av_assert1(s->in_buffer.planar == in_param->planar); | |
431 | av_assert1(s->in_buffer.fmt == in_param->fmt); | |
432 | ||
433 | tmp=out=*out_param; | |
434 | in = *in_param; | |
435 | ||
436 | border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer, | |
437 | &in, in_count, &s->in_buffer_index, &s->in_buffer_count); | |
438 | if (border == INT_MAX) return 0; | |
439 | else if (border < 0) return border; | |
440 | else if (border) { buf_set(&in, &in, border); in_count -= border; s->resample_in_constraint = 0; } | |
441 | ||
442 | do{ | |
443 | int ret, size, consumed; | |
444 | if(!s->resample_in_constraint && s->in_buffer_count){ | |
445 | buf_set(&tmp, &s->in_buffer, s->in_buffer_index); | |
446 | ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed); | |
447 | out_count -= ret; | |
448 | ret_sum += ret; | |
449 | buf_set(&out, &out, ret); | |
450 | s->in_buffer_count -= consumed; | |
451 | s->in_buffer_index += consumed; | |
452 | ||
453 | if(!in_count) | |
454 | break; | |
455 | if(s->in_buffer_count <= border){ | |
456 | buf_set(&in, &in, -s->in_buffer_count); | |
457 | in_count += s->in_buffer_count; | |
458 | s->in_buffer_count=0; | |
459 | s->in_buffer_index=0; | |
460 | border = 0; | |
461 | } | |
462 | } | |
463 | ||
464 | if((s->flushed || in_count > padless) && !s->in_buffer_count){ | |
465 | s->in_buffer_index=0; | |
466 | ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed); | |
467 | out_count -= ret; | |
468 | ret_sum += ret; | |
469 | buf_set(&out, &out, ret); | |
470 | in_count -= consumed; | |
471 | buf_set(&in, &in, consumed); | |
472 | } | |
473 | ||
474 | //TODO is this check sane considering the advanced copy avoidance below | |
475 | size= s->in_buffer_index + s->in_buffer_count + in_count; | |
476 | if( size > s->in_buffer.count | |
477 | && s->in_buffer_count + in_count <= s->in_buffer_index){ | |
478 | buf_set(&tmp, &s->in_buffer, s->in_buffer_index); | |
479 | copy(&s->in_buffer, &tmp, s->in_buffer_count); | |
480 | s->in_buffer_index=0; | |
481 | }else | |
482 | if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0) | |
483 | return ret; | |
484 | ||
485 | if(in_count){ | |
486 | int count= in_count; | |
487 | if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2; | |
488 | ||
489 | buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count); | |
490 | copy(&tmp, &in, /*in_*/count); | |
491 | s->in_buffer_count += count; | |
492 | in_count -= count; | |
493 | border += count; | |
494 | buf_set(&in, &in, count); | |
495 | s->resample_in_constraint= 0; | |
496 | if(s->in_buffer_count != count || in_count) | |
497 | continue; | |
498 | if (padless) { | |
499 | padless = 0; | |
500 | continue; | |
501 | } | |
502 | } | |
503 | break; | |
504 | }while(1); | |
505 | ||
506 | s->resample_in_constraint= !!out_count; | |
507 | ||
508 | return ret_sum; | |
509 | } | |
510 | ||
511 | static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count, | |
512 | AudioData *in , int in_count){ | |
513 | AudioData *postin, *midbuf, *preout; | |
514 | int ret/*, in_max*/; | |
515 | AudioData preout_tmp, midbuf_tmp; | |
516 | ||
517 | if(s->full_convert){ | |
518 | av_assert0(!s->resample); | |
519 | swri_audio_convert(s->full_convert, out, in, in_count); | |
520 | return out_count; | |
521 | } | |
522 | ||
523 | // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps; | |
524 | // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count); | |
525 | ||
526 | if((ret=swri_realloc_audio(&s->postin, in_count))<0) | |
527 | return ret; | |
528 | if(s->resample_first){ | |
529 | av_assert0(s->midbuf.ch_count == s->used_ch_count); | |
530 | if((ret=swri_realloc_audio(&s->midbuf, out_count))<0) | |
531 | return ret; | |
532 | }else{ | |
533 | av_assert0(s->midbuf.ch_count == s->out.ch_count); | |
534 | if((ret=swri_realloc_audio(&s->midbuf, in_count))<0) | |
535 | return ret; | |
536 | } | |
537 | if((ret=swri_realloc_audio(&s->preout, out_count))<0) | |
538 | return ret; | |
539 | ||
540 | postin= &s->postin; | |
541 | ||
542 | midbuf_tmp= s->midbuf; | |
543 | midbuf= &midbuf_tmp; | |
544 | preout_tmp= s->preout; | |
545 | preout= &preout_tmp; | |
546 | ||
547 | if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map) | |
548 | postin= in; | |
549 | ||
550 | if(s->resample_first ? !s->resample : !s->rematrix) | |
551 | midbuf= postin; | |
552 | ||
553 | if(s->resample_first ? !s->rematrix : !s->resample) | |
554 | preout= midbuf; | |
555 | ||
556 | if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar | |
557 | && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){ | |
558 | if(preout==in){ | |
559 | out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant | |
560 | av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though | |
561 | copy(out, in, out_count); | |
562 | return out_count; | |
563 | } | |
564 | else if(preout==postin) preout= midbuf= postin= out; | |
565 | else if(preout==midbuf) preout= midbuf= out; | |
566 | else preout= out; | |
567 | } | |
568 | ||
569 | if(in != postin){ | |
570 | swri_audio_convert(s->in_convert, postin, in, in_count); | |
571 | } | |
572 | ||
573 | if(s->resample_first){ | |
574 | if(postin != midbuf) | |
575 | out_count= resample(s, midbuf, out_count, postin, in_count); | |
576 | if(midbuf != preout) | |
577 | swri_rematrix(s, preout, midbuf, out_count, preout==out); | |
578 | }else{ | |
579 | if(postin != midbuf) | |
580 | swri_rematrix(s, midbuf, postin, in_count, midbuf==out); | |
581 | if(midbuf != preout) | |
582 | out_count= resample(s, preout, out_count, midbuf, in_count); | |
583 | } | |
584 | ||
585 | if(preout != out && out_count){ | |
586 | AudioData *conv_src = preout; | |
587 | if(s->dither.method){ | |
588 | int ch; | |
589 | int dither_count= FFMAX(out_count, 1<<16); | |
590 | ||
591 | if (preout == in) { | |
592 | conv_src = &s->dither.temp; | |
593 | if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0) | |
594 | return ret; | |
595 | } | |
596 | ||
597 | if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0) | |
598 | return ret; | |
599 | if(ret) | |
600 | for(ch=0; ch<s->dither.noise.ch_count; ch++) | |
601 | swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt); | |
602 | av_assert0(s->dither.noise.ch_count == preout->ch_count); | |
603 | ||
604 | if(s->dither.noise_pos + out_count > s->dither.noise.count) | |
605 | s->dither.noise_pos = 0; | |
606 | ||
607 | if (s->dither.method < SWR_DITHER_NS){ | |
608 | if (s->mix_2_1_simd) { | |
609 | int len1= out_count&~15; | |
610 | int off = len1 * preout->bps; | |
611 | ||
612 | if(len1) | |
613 | for(ch=0; ch<preout->ch_count; ch++) | |
614 | 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); | |
615 | if(out_count != len1) | |
616 | for(ch=0; ch<preout->ch_count; ch++) | |
617 | 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); | |
618 | } else { | |
619 | for(ch=0; ch<preout->ch_count; ch++) | |
620 | 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); | |
621 | } | |
622 | } else { | |
623 | switch(s->int_sample_fmt) { | |
624 | case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break; | |
625 | case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break; | |
626 | case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break; | |
627 | case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break; | |
628 | } | |
629 | } | |
630 | s->dither.noise_pos += out_count; | |
631 | } | |
632 | //FIXME packed doesn't need more than 1 chan here! | |
633 | swri_audio_convert(s->out_convert, out, conv_src, out_count); | |
634 | } | |
635 | return out_count; | |
636 | } | |
637 | ||
638 | int swr_is_initialized(struct SwrContext *s) { | |
639 | return !!s->in_buffer.ch_count; | |
640 | } | |
641 | ||
642 | int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count, | |
643 | const uint8_t *in_arg [SWR_CH_MAX], int in_count){ | |
644 | AudioData * in= &s->in; | |
645 | AudioData *out= &s->out; | |
646 | ||
647 | if (!swr_is_initialized(s)) { | |
648 | av_log(s, AV_LOG_ERROR, "Context has not been initialized\n"); | |
649 | return AVERROR(EINVAL); | |
650 | } | |
651 | ||
652 | while(s->drop_output > 0){ | |
653 | int ret; | |
654 | uint8_t *tmp_arg[SWR_CH_MAX]; | |
655 | #define MAX_DROP_STEP 16384 | |
656 | if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0) | |
657 | return ret; | |
658 | ||
659 | reversefill_audiodata(&s->drop_temp, tmp_arg); | |
660 | s->drop_output *= -1; //FIXME find a less hackish solution | |
661 | 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 | |
662 | s->drop_output *= -1; | |
663 | in_count = 0; | |
664 | if(ret>0) { | |
665 | s->drop_output -= ret; | |
666 | if (!s->drop_output && !out_arg) | |
667 | return 0; | |
668 | continue; | |
669 | } | |
670 | ||
671 | if(s->drop_output || !out_arg) | |
672 | return 0; | |
673 | } | |
674 | ||
675 | if(!in_arg){ | |
676 | if(s->resample){ | |
677 | if (!s->flushed) | |
678 | s->resampler->flush(s); | |
679 | s->resample_in_constraint = 0; | |
680 | s->flushed = 1; | |
681 | }else if(!s->in_buffer_count){ | |
682 | return 0; | |
683 | } | |
684 | }else | |
685 | fill_audiodata(in , (void*)in_arg); | |
686 | ||
687 | fill_audiodata(out, out_arg); | |
688 | ||
689 | if(s->resample){ | |
690 | int ret = swr_convert_internal(s, out, out_count, in, in_count); | |
691 | if(ret>0 && !s->drop_output) | |
692 | s->outpts += ret * (int64_t)s->in_sample_rate; | |
693 | return ret; | |
694 | }else{ | |
695 | AudioData tmp= *in; | |
696 | int ret2=0; | |
697 | int ret, size; | |
698 | size = FFMIN(out_count, s->in_buffer_count); | |
699 | if(size){ | |
700 | buf_set(&tmp, &s->in_buffer, s->in_buffer_index); | |
701 | ret= swr_convert_internal(s, out, size, &tmp, size); | |
702 | if(ret<0) | |
703 | return ret; | |
704 | ret2= ret; | |
705 | s->in_buffer_count -= ret; | |
706 | s->in_buffer_index += ret; | |
707 | buf_set(out, out, ret); | |
708 | out_count -= ret; | |
709 | if(!s->in_buffer_count) | |
710 | s->in_buffer_index = 0; | |
711 | } | |
712 | ||
713 | if(in_count){ | |
714 | size= s->in_buffer_index + s->in_buffer_count + in_count - out_count; | |
715 | ||
716 | if(in_count > out_count) { //FIXME move after swr_convert_internal | |
717 | if( size > s->in_buffer.count | |
718 | && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){ | |
719 | buf_set(&tmp, &s->in_buffer, s->in_buffer_index); | |
720 | copy(&s->in_buffer, &tmp, s->in_buffer_count); | |
721 | s->in_buffer_index=0; | |
722 | }else | |
723 | if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0) | |
724 | return ret; | |
725 | } | |
726 | ||
727 | if(out_count){ | |
728 | size = FFMIN(in_count, out_count); | |
729 | ret= swr_convert_internal(s, out, size, in, size); | |
730 | if(ret<0) | |
731 | return ret; | |
732 | buf_set(in, in, ret); | |
733 | in_count -= ret; | |
734 | ret2 += ret; | |
735 | } | |
736 | if(in_count){ | |
737 | buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count); | |
738 | copy(&tmp, in, in_count); | |
739 | s->in_buffer_count += in_count; | |
740 | } | |
741 | } | |
742 | if(ret2>0 && !s->drop_output) | |
743 | s->outpts += ret2 * (int64_t)s->in_sample_rate; | |
744 | return ret2; | |
745 | } | |
746 | } | |
747 | ||
748 | int swr_drop_output(struct SwrContext *s, int count){ | |
749 | s->drop_output += count; | |
750 | ||
751 | if(s->drop_output <= 0) | |
752 | return 0; | |
753 | ||
754 | av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count); | |
755 | return swr_convert(s, NULL, s->drop_output, NULL, 0); | |
756 | } | |
757 | ||
758 | int swr_inject_silence(struct SwrContext *s, int count){ | |
759 | int ret, i; | |
760 | uint8_t *tmp_arg[SWR_CH_MAX]; | |
761 | ||
762 | if(count <= 0) | |
763 | return 0; | |
764 | ||
765 | #define MAX_SILENCE_STEP 16384 | |
766 | while (count > MAX_SILENCE_STEP) { | |
767 | if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0) | |
768 | return ret; | |
769 | count -= MAX_SILENCE_STEP; | |
770 | } | |
771 | ||
772 | if((ret=swri_realloc_audio(&s->silence, count))<0) | |
773 | return ret; | |
774 | ||
775 | if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) { | |
776 | memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps); | |
777 | } else | |
778 | memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count); | |
779 | ||
780 | reversefill_audiodata(&s->silence, tmp_arg); | |
781 | av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count); | |
782 | ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count); | |
783 | return ret; | |
784 | } | |
785 | ||
786 | int64_t swr_get_delay(struct SwrContext *s, int64_t base){ | |
787 | if (s->resampler && s->resample){ | |
788 | return s->resampler->get_delay(s, base); | |
789 | }else{ | |
790 | return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate; | |
791 | } | |
792 | } | |
793 | ||
794 | int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){ | |
795 | int ret; | |
796 | ||
797 | if (!s || compensation_distance < 0) | |
798 | return AVERROR(EINVAL); | |
799 | if (!compensation_distance && sample_delta) | |
800 | return AVERROR(EINVAL); | |
801 | if (!s->resample) { | |
802 | s->flags |= SWR_FLAG_RESAMPLE; | |
803 | ret = swr_init(s); | |
804 | if (ret < 0) | |
805 | return ret; | |
806 | } | |
807 | if (!s->resampler->set_compensation){ | |
808 | return AVERROR(EINVAL); | |
809 | }else{ | |
810 | return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance); | |
811 | } | |
812 | } | |
813 | ||
814 | int64_t swr_next_pts(struct SwrContext *s, int64_t pts){ | |
815 | if(pts == INT64_MIN) | |
816 | return s->outpts; | |
817 | ||
818 | if (s->firstpts == AV_NOPTS_VALUE) | |
819 | s->outpts = s->firstpts = pts; | |
820 | ||
821 | if(s->min_compensation >= FLT_MAX) { | |
822 | return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate)); | |
823 | } else { | |
824 | 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; | |
825 | double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate); | |
826 | ||
827 | if(fabs(fdelta) > s->min_compensation) { | |
828 | if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){ | |
829 | int ret; | |
830 | if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate); | |
831 | else ret = swr_drop_output (s, -delta / s-> in_sample_rate); | |
832 | if(ret<0){ | |
833 | av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta); | |
834 | } | |
835 | } else if(s->soft_compensation_duration && s->max_soft_compensation) { | |
836 | int duration = s->out_sample_rate * s->soft_compensation_duration; | |
837 | double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1); | |
838 | int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ; | |
839 | av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration); | |
840 | swr_set_compensation(s, comp, duration); | |
841 | } | |
842 | } | |
843 | ||
844 | return s->outpts; | |
845 | } | |
846 | } |