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