| 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) { |
| 439 | return 0; |
| 440 | } else if (border < 0) { |
| 441 | return border; |
| 442 | } else if (border) { |
| 443 | buf_set(&in, &in, border); |
| 444 | in_count -= border; |
| 445 | s->resample_in_constraint = 0; |
| 446 | } |
| 447 | |
| 448 | do{ |
| 449 | int ret, size, consumed; |
| 450 | if(!s->resample_in_constraint && s->in_buffer_count){ |
| 451 | buf_set(&tmp, &s->in_buffer, s->in_buffer_index); |
| 452 | ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed); |
| 453 | out_count -= ret; |
| 454 | ret_sum += ret; |
| 455 | buf_set(&out, &out, ret); |
| 456 | s->in_buffer_count -= consumed; |
| 457 | s->in_buffer_index += consumed; |
| 458 | |
| 459 | if(!in_count) |
| 460 | break; |
| 461 | if(s->in_buffer_count <= border){ |
| 462 | buf_set(&in, &in, -s->in_buffer_count); |
| 463 | in_count += s->in_buffer_count; |
| 464 | s->in_buffer_count=0; |
| 465 | s->in_buffer_index=0; |
| 466 | border = 0; |
| 467 | } |
| 468 | } |
| 469 | |
| 470 | if((s->flushed || in_count > padless) && !s->in_buffer_count){ |
| 471 | s->in_buffer_index=0; |
| 472 | ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed); |
| 473 | out_count -= ret; |
| 474 | ret_sum += ret; |
| 475 | buf_set(&out, &out, ret); |
| 476 | in_count -= consumed; |
| 477 | buf_set(&in, &in, consumed); |
| 478 | } |
| 479 | |
| 480 | //TODO is this check sane considering the advanced copy avoidance below |
| 481 | size= s->in_buffer_index + s->in_buffer_count + in_count; |
| 482 | if( size > s->in_buffer.count |
| 483 | && s->in_buffer_count + in_count <= s->in_buffer_index){ |
| 484 | buf_set(&tmp, &s->in_buffer, s->in_buffer_index); |
| 485 | copy(&s->in_buffer, &tmp, s->in_buffer_count); |
| 486 | s->in_buffer_index=0; |
| 487 | }else |
| 488 | if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0) |
| 489 | return ret; |
| 490 | |
| 491 | if(in_count){ |
| 492 | int count= in_count; |
| 493 | if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2; |
| 494 | |
| 495 | buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count); |
| 496 | copy(&tmp, &in, /*in_*/count); |
| 497 | s->in_buffer_count += count; |
| 498 | in_count -= count; |
| 499 | border += count; |
| 500 | buf_set(&in, &in, count); |
| 501 | s->resample_in_constraint= 0; |
| 502 | if(s->in_buffer_count != count || in_count) |
| 503 | continue; |
| 504 | if (padless) { |
| 505 | padless = 0; |
| 506 | continue; |
| 507 | } |
| 508 | } |
| 509 | break; |
| 510 | }while(1); |
| 511 | |
| 512 | s->resample_in_constraint= !!out_count; |
| 513 | |
| 514 | return ret_sum; |
| 515 | } |
| 516 | |
| 517 | static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count, |
| 518 | AudioData *in , int in_count){ |
| 519 | AudioData *postin, *midbuf, *preout; |
| 520 | int ret/*, in_max*/; |
| 521 | AudioData preout_tmp, midbuf_tmp; |
| 522 | |
| 523 | if(s->full_convert){ |
| 524 | av_assert0(!s->resample); |
| 525 | swri_audio_convert(s->full_convert, out, in, in_count); |
| 526 | return out_count; |
| 527 | } |
| 528 | |
| 529 | // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps; |
| 530 | // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count); |
| 531 | |
| 532 | if((ret=swri_realloc_audio(&s->postin, in_count))<0) |
| 533 | return ret; |
| 534 | if(s->resample_first){ |
| 535 | av_assert0(s->midbuf.ch_count == s->used_ch_count); |
| 536 | if((ret=swri_realloc_audio(&s->midbuf, out_count))<0) |
| 537 | return ret; |
| 538 | }else{ |
| 539 | av_assert0(s->midbuf.ch_count == s->out.ch_count); |
| 540 | if((ret=swri_realloc_audio(&s->midbuf, in_count))<0) |
| 541 | return ret; |
| 542 | } |
| 543 | if((ret=swri_realloc_audio(&s->preout, out_count))<0) |
| 544 | return ret; |
| 545 | |
| 546 | postin= &s->postin; |
| 547 | |
| 548 | midbuf_tmp= s->midbuf; |
| 549 | midbuf= &midbuf_tmp; |
| 550 | preout_tmp= s->preout; |
| 551 | preout= &preout_tmp; |
| 552 | |
| 553 | if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map) |
| 554 | postin= in; |
| 555 | |
| 556 | if(s->resample_first ? !s->resample : !s->rematrix) |
| 557 | midbuf= postin; |
| 558 | |
| 559 | if(s->resample_first ? !s->rematrix : !s->resample) |
| 560 | preout= midbuf; |
| 561 | |
| 562 | if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar |
| 563 | && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){ |
| 564 | if(preout==in){ |
| 565 | out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant |
| 566 | av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though |
| 567 | copy(out, in, out_count); |
| 568 | return out_count; |
| 569 | } |
| 570 | else if(preout==postin) preout= midbuf= postin= out; |
| 571 | else if(preout==midbuf) preout= midbuf= out; |
| 572 | else preout= out; |
| 573 | } |
| 574 | |
| 575 | if(in != postin){ |
| 576 | swri_audio_convert(s->in_convert, postin, in, in_count); |
| 577 | } |
| 578 | |
| 579 | if(s->resample_first){ |
| 580 | if(postin != midbuf) |
| 581 | out_count= resample(s, midbuf, out_count, postin, in_count); |
| 582 | if(midbuf != preout) |
| 583 | swri_rematrix(s, preout, midbuf, out_count, preout==out); |
| 584 | }else{ |
| 585 | if(postin != midbuf) |
| 586 | swri_rematrix(s, midbuf, postin, in_count, midbuf==out); |
| 587 | if(midbuf != preout) |
| 588 | out_count= resample(s, preout, out_count, midbuf, in_count); |
| 589 | } |
| 590 | |
| 591 | if(preout != out && out_count){ |
| 592 | AudioData *conv_src = preout; |
| 593 | if(s->dither.method){ |
| 594 | int ch; |
| 595 | int dither_count= FFMAX(out_count, 1<<16); |
| 596 | |
| 597 | if (preout == in) { |
| 598 | conv_src = &s->dither.temp; |
| 599 | if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0) |
| 600 | return ret; |
| 601 | } |
| 602 | |
| 603 | if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0) |
| 604 | return ret; |
| 605 | if(ret) |
| 606 | for(ch=0; ch<s->dither.noise.ch_count; ch++) |
| 607 | swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt); |
| 608 | av_assert0(s->dither.noise.ch_count == preout->ch_count); |
| 609 | |
| 610 | if(s->dither.noise_pos + out_count > s->dither.noise.count) |
| 611 | s->dither.noise_pos = 0; |
| 612 | |
| 613 | if (s->dither.method < SWR_DITHER_NS){ |
| 614 | if (s->mix_2_1_simd) { |
| 615 | int len1= out_count&~15; |
| 616 | int off = len1 * preout->bps; |
| 617 | |
| 618 | if(len1) |
| 619 | for(ch=0; ch<preout->ch_count; ch++) |
| 620 | 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); |
| 621 | if(out_count != len1) |
| 622 | for(ch=0; ch<preout->ch_count; ch++) |
| 623 | 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); |
| 624 | } else { |
| 625 | for(ch=0; ch<preout->ch_count; ch++) |
| 626 | 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); |
| 627 | } |
| 628 | } else { |
| 629 | switch(s->int_sample_fmt) { |
| 630 | case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break; |
| 631 | case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break; |
| 632 | case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break; |
| 633 | case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break; |
| 634 | } |
| 635 | } |
| 636 | s->dither.noise_pos += out_count; |
| 637 | } |
| 638 | //FIXME packed doesn't need more than 1 chan here! |
| 639 | swri_audio_convert(s->out_convert, out, conv_src, out_count); |
| 640 | } |
| 641 | return out_count; |
| 642 | } |
| 643 | |
| 644 | int swr_is_initialized(struct SwrContext *s) { |
| 645 | return !!s->in_buffer.ch_count; |
| 646 | } |
| 647 | |
| 648 | int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count, |
| 649 | const uint8_t *in_arg [SWR_CH_MAX], int in_count){ |
| 650 | AudioData * in= &s->in; |
| 651 | AudioData *out= &s->out; |
| 652 | |
| 653 | if (!swr_is_initialized(s)) { |
| 654 | av_log(s, AV_LOG_ERROR, "Context has not been initialized\n"); |
| 655 | return AVERROR(EINVAL); |
| 656 | } |
| 657 | |
| 658 | while(s->drop_output > 0){ |
| 659 | int ret; |
| 660 | uint8_t *tmp_arg[SWR_CH_MAX]; |
| 661 | #define MAX_DROP_STEP 16384 |
| 662 | if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0) |
| 663 | return ret; |
| 664 | |
| 665 | reversefill_audiodata(&s->drop_temp, tmp_arg); |
| 666 | s->drop_output *= -1; //FIXME find a less hackish solution |
| 667 | 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 |
| 668 | s->drop_output *= -1; |
| 669 | in_count = 0; |
| 670 | if(ret>0) { |
| 671 | s->drop_output -= ret; |
| 672 | if (!s->drop_output && !out_arg) |
| 673 | return 0; |
| 674 | continue; |
| 675 | } |
| 676 | |
| 677 | av_assert0(s->drop_output); |
| 678 | return 0; |
| 679 | } |
| 680 | |
| 681 | if(!in_arg){ |
| 682 | if(s->resample){ |
| 683 | if (!s->flushed) |
| 684 | s->resampler->flush(s); |
| 685 | s->resample_in_constraint = 0; |
| 686 | s->flushed = 1; |
| 687 | }else if(!s->in_buffer_count){ |
| 688 | return 0; |
| 689 | } |
| 690 | }else |
| 691 | fill_audiodata(in , (void*)in_arg); |
| 692 | |
| 693 | fill_audiodata(out, out_arg); |
| 694 | |
| 695 | if(s->resample){ |
| 696 | int ret = swr_convert_internal(s, out, out_count, in, in_count); |
| 697 | if(ret>0 && !s->drop_output) |
| 698 | s->outpts += ret * (int64_t)s->in_sample_rate; |
| 699 | return ret; |
| 700 | }else{ |
| 701 | AudioData tmp= *in; |
| 702 | int ret2=0; |
| 703 | int ret, size; |
| 704 | size = FFMIN(out_count, s->in_buffer_count); |
| 705 | if(size){ |
| 706 | buf_set(&tmp, &s->in_buffer, s->in_buffer_index); |
| 707 | ret= swr_convert_internal(s, out, size, &tmp, size); |
| 708 | if(ret<0) |
| 709 | return ret; |
| 710 | ret2= ret; |
| 711 | s->in_buffer_count -= ret; |
| 712 | s->in_buffer_index += ret; |
| 713 | buf_set(out, out, ret); |
| 714 | out_count -= ret; |
| 715 | if(!s->in_buffer_count) |
| 716 | s->in_buffer_index = 0; |
| 717 | } |
| 718 | |
| 719 | if(in_count){ |
| 720 | size= s->in_buffer_index + s->in_buffer_count + in_count - out_count; |
| 721 | |
| 722 | if(in_count > out_count) { //FIXME move after swr_convert_internal |
| 723 | if( size > s->in_buffer.count |
| 724 | && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){ |
| 725 | buf_set(&tmp, &s->in_buffer, s->in_buffer_index); |
| 726 | copy(&s->in_buffer, &tmp, s->in_buffer_count); |
| 727 | s->in_buffer_index=0; |
| 728 | }else |
| 729 | if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0) |
| 730 | return ret; |
| 731 | } |
| 732 | |
| 733 | if(out_count){ |
| 734 | size = FFMIN(in_count, out_count); |
| 735 | ret= swr_convert_internal(s, out, size, in, size); |
| 736 | if(ret<0) |
| 737 | return ret; |
| 738 | buf_set(in, in, ret); |
| 739 | in_count -= ret; |
| 740 | ret2 += ret; |
| 741 | } |
| 742 | if(in_count){ |
| 743 | buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count); |
| 744 | copy(&tmp, in, in_count); |
| 745 | s->in_buffer_count += in_count; |
| 746 | } |
| 747 | } |
| 748 | if(ret2>0 && !s->drop_output) |
| 749 | s->outpts += ret2 * (int64_t)s->in_sample_rate; |
| 750 | return ret2; |
| 751 | } |
| 752 | } |
| 753 | |
| 754 | int swr_drop_output(struct SwrContext *s, int count){ |
| 755 | const uint8_t *tmp_arg[SWR_CH_MAX]; |
| 756 | s->drop_output += count; |
| 757 | |
| 758 | if(s->drop_output <= 0) |
| 759 | return 0; |
| 760 | |
| 761 | av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count); |
| 762 | return swr_convert(s, NULL, s->drop_output, tmp_arg, 0); |
| 763 | } |
| 764 | |
| 765 | int swr_inject_silence(struct SwrContext *s, int count){ |
| 766 | int ret, i; |
| 767 | uint8_t *tmp_arg[SWR_CH_MAX]; |
| 768 | |
| 769 | if(count <= 0) |
| 770 | return 0; |
| 771 | |
| 772 | #define MAX_SILENCE_STEP 16384 |
| 773 | while (count > MAX_SILENCE_STEP) { |
| 774 | if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0) |
| 775 | return ret; |
| 776 | count -= MAX_SILENCE_STEP; |
| 777 | } |
| 778 | |
| 779 | if((ret=swri_realloc_audio(&s->silence, count))<0) |
| 780 | return ret; |
| 781 | |
| 782 | if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) { |
| 783 | memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps); |
| 784 | } else |
| 785 | memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count); |
| 786 | |
| 787 | reversefill_audiodata(&s->silence, tmp_arg); |
| 788 | av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count); |
| 789 | ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count); |
| 790 | return ret; |
| 791 | } |
| 792 | |
| 793 | int64_t swr_get_delay(struct SwrContext *s, int64_t base){ |
| 794 | if (s->resampler && s->resample){ |
| 795 | return s->resampler->get_delay(s, base); |
| 796 | }else{ |
| 797 | return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate; |
| 798 | } |
| 799 | } |
| 800 | |
| 801 | int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){ |
| 802 | int ret; |
| 803 | |
| 804 | if (!s || compensation_distance < 0) |
| 805 | return AVERROR(EINVAL); |
| 806 | if (!compensation_distance && sample_delta) |
| 807 | return AVERROR(EINVAL); |
| 808 | if (!s->resample) { |
| 809 | s->flags |= SWR_FLAG_RESAMPLE; |
| 810 | ret = swr_init(s); |
| 811 | if (ret < 0) |
| 812 | return ret; |
| 813 | } |
| 814 | if (!s->resampler->set_compensation){ |
| 815 | return AVERROR(EINVAL); |
| 816 | }else{ |
| 817 | return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance); |
| 818 | } |
| 819 | } |
| 820 | |
| 821 | int64_t swr_next_pts(struct SwrContext *s, int64_t pts){ |
| 822 | if(pts == INT64_MIN) |
| 823 | return s->outpts; |
| 824 | |
| 825 | if (s->firstpts == AV_NOPTS_VALUE) |
| 826 | s->outpts = s->firstpts = pts; |
| 827 | |
| 828 | if(s->min_compensation >= FLT_MAX) { |
| 829 | return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate)); |
| 830 | } else { |
| 831 | 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; |
| 832 | double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate); |
| 833 | |
| 834 | if(fabs(fdelta) > s->min_compensation) { |
| 835 | if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){ |
| 836 | int ret; |
| 837 | if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate); |
| 838 | else ret = swr_drop_output (s, -delta / s-> in_sample_rate); |
| 839 | if(ret<0){ |
| 840 | av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta); |
| 841 | } |
| 842 | } else if(s->soft_compensation_duration && s->max_soft_compensation) { |
| 843 | int duration = s->out_sample_rate * s->soft_compensation_duration; |
| 844 | double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1); |
| 845 | int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ; |
| 846 | av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration); |
| 847 | swr_set_compensation(s, comp, duration); |
| 848 | } |
| 849 | } |
| 850 | |
| 851 | return s->outpts; |
| 852 | } |
| 853 | } |