1/* 2 * MLP decoder 3 * Copyright (c) 2007-2008 Ian Caulfield 4 * 5 * This file is part of FFmpeg. 6 * 7 * FFmpeg is free software; you can redistribute it and/or 8 * modify it under the terms of the GNU Lesser General Public 9 * License as published by the Free Software Foundation; either 10 * version 2.1 of the License, or (at your option) any later version. 11 * 12 * FFmpeg is distributed in the hope that it will be useful, 13 * but WITHOUT ANY WARRANTY; without even the implied warranty of 14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 15 * Lesser General Public License for more details. 16 * 17 * You should have received a copy of the GNU Lesser General Public 18 * License along with FFmpeg; if not, write to the Free Software 19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 20 */ 21 22/** 23 * @file 24 * MLP decoder 25 */ 26 27#include <stdint.h> 28 29#include "avcodec.h" 30#include "dsputil.h" 31#include "libavutil/intreadwrite.h" 32#include "get_bits.h" 33#include "libavutil/crc.h" 34#include "parser.h" 35#include "mlp_parser.h" 36#include "mlp.h" 37 38/** number of bits used for VLC lookup - longest Huffman code is 9 */ 39#define VLC_BITS 9 40 41 42static const char* sample_message = 43 "Please file a bug report following the instructions at " 44 "http://ffmpeg.org/bugreports.html and include " 45 "a sample of this file."; 46 47typedef struct SubStream { 48 //! Set if a valid restart header has been read. Otherwise the substream cannot be decoded. 49 uint8_t restart_seen; 50 51 //@{ 52 /** restart header data */ 53 //! The type of noise to be used in the rematrix stage. 54 uint16_t noise_type; 55 56 //! The index of the first channel coded in this substream. 57 uint8_t min_channel; 58 //! The index of the last channel coded in this substream. 59 uint8_t max_channel; 60 //! The number of channels input into the rematrix stage. 61 uint8_t max_matrix_channel; 62 //! For each channel output by the matrix, the output channel to map it to 63 uint8_t ch_assign[MAX_CHANNELS]; 64 65 //! Channel coding parameters for channels in the substream 66 ChannelParams channel_params[MAX_CHANNELS]; 67 68 //! The left shift applied to random noise in 0x31ea substreams. 69 uint8_t noise_shift; 70 //! The current seed value for the pseudorandom noise generator(s). 71 uint32_t noisegen_seed; 72 73 //! Set if the substream contains extra info to check the size of VLC blocks. 74 uint8_t data_check_present; 75 76 //! Bitmask of which parameter sets are conveyed in a decoding parameter block. 77 uint8_t param_presence_flags; 78#define PARAM_BLOCKSIZE (1 << 7) 79#define PARAM_MATRIX (1 << 6) 80#define PARAM_OUTSHIFT (1 << 5) 81#define PARAM_QUANTSTEP (1 << 4) 82#define PARAM_FIR (1 << 3) 83#define PARAM_IIR (1 << 2) 84#define PARAM_HUFFOFFSET (1 << 1) 85#define PARAM_PRESENCE (1 << 0) 86 //@} 87 88 //@{ 89 /** matrix data */ 90 91 //! Number of matrices to be applied. 92 uint8_t num_primitive_matrices; 93 94 //! matrix output channel 95 uint8_t matrix_out_ch[MAX_MATRICES]; 96 97 //! Whether the LSBs of the matrix output are encoded in the bitstream. 98 uint8_t lsb_bypass[MAX_MATRICES]; 99 //! Matrix coefficients, stored as 2.14 fixed point. 100 int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS]; 101 //! Left shift to apply to noise values in 0x31eb substreams. 102 uint8_t matrix_noise_shift[MAX_MATRICES]; 103 //@} 104 105 //! Left shift to apply to Huffman-decoded residuals. 106 uint8_t quant_step_size[MAX_CHANNELS]; 107 108 //! number of PCM samples in current audio block 109 uint16_t blocksize; 110 //! Number of PCM samples decoded so far in this frame. 111 uint16_t blockpos; 112 113 //! Left shift to apply to decoded PCM values to get final 24-bit output. 114 int8_t output_shift[MAX_CHANNELS]; 115 116 //! Running XOR of all output samples. 117 int32_t lossless_check_data; 118 119} SubStream; 120 121typedef struct MLPDecodeContext { 122 AVCodecContext *avctx; 123 124 //! Current access unit being read has a major sync. 125 int is_major_sync_unit; 126 127 //! Set if a valid major sync block has been read. Otherwise no decoding is possible. 128 uint8_t params_valid; 129 130 //! Number of substreams contained within this stream. 131 uint8_t num_substreams; 132 133 //! Index of the last substream to decode - further substreams are skipped. 134 uint8_t max_decoded_substream; 135 136 //! number of PCM samples contained in each frame 137 int access_unit_size; 138 //! next power of two above the number of samples in each frame 139 int access_unit_size_pow2; 140 141 SubStream substream[MAX_SUBSTREAMS]; 142 143 int matrix_changed; 144 int filter_changed[MAX_CHANNELS][NUM_FILTERS]; 145 146 int8_t noise_buffer[MAX_BLOCKSIZE_POW2]; 147 int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS]; 148 int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS]; 149 150 DSPContext dsp; 151} MLPDecodeContext; 152 153static VLC huff_vlc[3]; 154 155/** Initialize static data, constant between all invocations of the codec. */ 156 157static av_cold void init_static(void) 158{ 159 if (!huff_vlc[0].bits) { 160 INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18, 161 &ff_mlp_huffman_tables[0][0][1], 2, 1, 162 &ff_mlp_huffman_tables[0][0][0], 2, 1, 512); 163 INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16, 164 &ff_mlp_huffman_tables[1][0][1], 2, 1, 165 &ff_mlp_huffman_tables[1][0][0], 2, 1, 512); 166 INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15, 167 &ff_mlp_huffman_tables[2][0][1], 2, 1, 168 &ff_mlp_huffman_tables[2][0][0], 2, 1, 512); 169 } 170 171 ff_mlp_init_crc(); 172} 173 174static inline int32_t calculate_sign_huff(MLPDecodeContext *m, 175 unsigned int substr, unsigned int ch) 176{ 177 SubStream *s = &m->substream[substr]; 178 ChannelParams *cp = &s->channel_params[ch]; 179 int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch]; 180 int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1); 181 int32_t sign_huff_offset = cp->huff_offset; 182 183 if (cp->codebook > 0) 184 sign_huff_offset -= 7 << lsb_bits; 185 186 if (sign_shift >= 0) 187 sign_huff_offset -= 1 << sign_shift; 188 189 return sign_huff_offset; 190} 191 192/** Read a sample, consisting of either, both or neither of entropy-coded MSBs 193 * and plain LSBs. */ 194 195static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp, 196 unsigned int substr, unsigned int pos) 197{ 198 SubStream *s = &m->substream[substr]; 199 unsigned int mat, channel; 200 201 for (mat = 0; mat < s->num_primitive_matrices; mat++) 202 if (s->lsb_bypass[mat]) 203 m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp); 204 205 for (channel = s->min_channel; channel <= s->max_channel; channel++) { 206 ChannelParams *cp = &s->channel_params[channel]; 207 int codebook = cp->codebook; 208 int quant_step_size = s->quant_step_size[channel]; 209 int lsb_bits = cp->huff_lsbs - quant_step_size; 210 int result = 0; 211 212 if (codebook > 0) 213 result = get_vlc2(gbp, huff_vlc[codebook-1].table, 214 VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS); 215 216 if (result < 0) 217 return -1; 218 219 if (lsb_bits > 0) 220 result = (result << lsb_bits) + get_bits(gbp, lsb_bits); 221 222 result += cp->sign_huff_offset; 223 result <<= quant_step_size; 224 225 m->sample_buffer[pos + s->blockpos][channel] = result; 226 } 227 228 return 0; 229} 230 231static av_cold int mlp_decode_init(AVCodecContext *avctx) 232{ 233 MLPDecodeContext *m = avctx->priv_data; 234 int substr; 235 236 init_static(); 237 m->avctx = avctx; 238 for (substr = 0; substr < MAX_SUBSTREAMS; substr++) 239 m->substream[substr].lossless_check_data = 0xffffffff; 240 dsputil_init(&m->dsp, avctx); 241 242 return 0; 243} 244 245/** Read a major sync info header - contains high level information about 246 * the stream - sample rate, channel arrangement etc. Most of this 247 * information is not actually necessary for decoding, only for playback. 248 */ 249 250static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb) 251{ 252 MLPHeaderInfo mh; 253 int substr; 254 255 if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0) 256 return -1; 257 258 if (mh.group1_bits == 0) { 259 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n"); 260 return -1; 261 } 262 if (mh.group2_bits > mh.group1_bits) { 263 av_log(m->avctx, AV_LOG_ERROR, 264 "Channel group 2 cannot have more bits per sample than group 1.\n"); 265 return -1; 266 } 267 268 if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) { 269 av_log(m->avctx, AV_LOG_ERROR, 270 "Channel groups with differing sample rates are not currently supported.\n"); 271 return -1; 272 } 273 274 if (mh.group1_samplerate == 0) { 275 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n"); 276 return -1; 277 } 278 if (mh.group1_samplerate > MAX_SAMPLERATE) { 279 av_log(m->avctx, AV_LOG_ERROR, 280 "Sampling rate %d is greater than the supported maximum (%d).\n", 281 mh.group1_samplerate, MAX_SAMPLERATE); 282 return -1; 283 } 284 if (mh.access_unit_size > MAX_BLOCKSIZE) { 285 av_log(m->avctx, AV_LOG_ERROR, 286 "Block size %d is greater than the supported maximum (%d).\n", 287 mh.access_unit_size, MAX_BLOCKSIZE); 288 return -1; 289 } 290 if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) { 291 av_log(m->avctx, AV_LOG_ERROR, 292 "Block size pow2 %d is greater than the supported maximum (%d).\n", 293 mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2); 294 return -1; 295 } 296 297 if (mh.num_substreams == 0) 298 return -1; 299 if (m->avctx->codec_id == CODEC_ID_MLP && mh.num_substreams > 2) { 300 av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n"); 301 return -1; 302 } 303 if (mh.num_substreams > MAX_SUBSTREAMS) { 304 av_log(m->avctx, AV_LOG_ERROR, 305 "Number of substreams %d is larger than the maximum supported " 306 "by the decoder. %s\n", mh.num_substreams, sample_message); 307 return -1; 308 } 309 310 m->access_unit_size = mh.access_unit_size; 311 m->access_unit_size_pow2 = mh.access_unit_size_pow2; 312 313 m->num_substreams = mh.num_substreams; 314 m->max_decoded_substream = m->num_substreams - 1; 315 316 m->avctx->sample_rate = mh.group1_samplerate; 317 m->avctx->frame_size = mh.access_unit_size; 318 319 m->avctx->bits_per_raw_sample = mh.group1_bits; 320 if (mh.group1_bits > 16) 321 m->avctx->sample_fmt = SAMPLE_FMT_S32; 322 else 323 m->avctx->sample_fmt = SAMPLE_FMT_S16; 324 325 m->params_valid = 1; 326 for (substr = 0; substr < MAX_SUBSTREAMS; substr++) 327 m->substream[substr].restart_seen = 0; 328 329 return 0; 330} 331 332/** Read a restart header from a block in a substream. This contains parameters 333 * required to decode the audio that do not change very often. Generally 334 * (always) present only in blocks following a major sync. */ 335 336static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp, 337 const uint8_t *buf, unsigned int substr) 338{ 339 SubStream *s = &m->substream[substr]; 340 unsigned int ch; 341 int sync_word, tmp; 342 uint8_t checksum; 343 uint8_t lossless_check; 344 int start_count = get_bits_count(gbp); 345 const int max_matrix_channel = m->avctx->codec_id == CODEC_ID_MLP 346 ? MAX_MATRIX_CHANNEL_MLP 347 : MAX_MATRIX_CHANNEL_TRUEHD; 348 349 sync_word = get_bits(gbp, 13); 350 351 if (sync_word != 0x31ea >> 1) { 352 av_log(m->avctx, AV_LOG_ERROR, 353 "restart header sync incorrect (got 0x%04x)\n", sync_word); 354 return -1; 355 } 356 357 s->noise_type = get_bits1(gbp); 358 359 if (m->avctx->codec_id == CODEC_ID_MLP && s->noise_type) { 360 av_log(m->avctx, AV_LOG_ERROR, "MLP must have 0x31ea sync word.\n"); 361 return -1; 362 } 363 364 skip_bits(gbp, 16); /* Output timestamp */ 365 366 s->min_channel = get_bits(gbp, 4); 367 s->max_channel = get_bits(gbp, 4); 368 s->max_matrix_channel = get_bits(gbp, 4); 369 370 if (s->max_matrix_channel > max_matrix_channel) { 371 av_log(m->avctx, AV_LOG_ERROR, 372 "Max matrix channel cannot be greater than %d.\n", 373 max_matrix_channel); 374 return -1; 375 } 376 377 if (s->max_channel != s->max_matrix_channel) { 378 av_log(m->avctx, AV_LOG_ERROR, 379 "Max channel must be equal max matrix channel.\n"); 380 return -1; 381 } 382 383 /* This should happen for TrueHD streams with >6 channels and MLP's noise 384 * type. It is not yet known if this is allowed. */ 385 if (s->max_channel > MAX_MATRIX_CHANNEL_MLP && !s->noise_type) { 386 av_log(m->avctx, AV_LOG_ERROR, 387 "Number of channels %d is larger than the maximum supported " 388 "by the decoder. %s\n", s->max_channel+2, sample_message); 389 return -1; 390 } 391 392 if (s->min_channel > s->max_channel) { 393 av_log(m->avctx, AV_LOG_ERROR, 394 "Substream min channel cannot be greater than max channel.\n"); 395 return -1; 396 } 397 398 if (m->avctx->request_channels > 0 399 && s->max_channel + 1 >= m->avctx->request_channels 400 && substr < m->max_decoded_substream) { 401 av_log(m->avctx, AV_LOG_DEBUG, 402 "Extracting %d channel downmix from substream %d. " 403 "Further substreams will be skipped.\n", 404 s->max_channel + 1, substr); 405 m->max_decoded_substream = substr; 406 } 407 408 s->noise_shift = get_bits(gbp, 4); 409 s->noisegen_seed = get_bits(gbp, 23); 410 411 skip_bits(gbp, 19); 412 413 s->data_check_present = get_bits1(gbp); 414 lossless_check = get_bits(gbp, 8); 415 if (substr == m->max_decoded_substream 416 && s->lossless_check_data != 0xffffffff) { 417 tmp = xor_32_to_8(s->lossless_check_data); 418 if (tmp != lossless_check) 419 av_log(m->avctx, AV_LOG_WARNING, 420 "Lossless check failed - expected %02x, calculated %02x.\n", 421 lossless_check, tmp); 422 } 423 424 skip_bits(gbp, 16); 425 426 memset(s->ch_assign, 0, sizeof(s->ch_assign)); 427 428 for (ch = 0; ch <= s->max_matrix_channel; ch++) { 429 int ch_assign = get_bits(gbp, 6); 430 if (ch_assign > s->max_matrix_channel) { 431 av_log(m->avctx, AV_LOG_ERROR, 432 "Assignment of matrix channel %d to invalid output channel %d. %s\n", 433 ch, ch_assign, sample_message); 434 return -1; 435 } 436 s->ch_assign[ch_assign] = ch; 437 } 438 439 checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count); 440 441 if (checksum != get_bits(gbp, 8)) 442 av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n"); 443 444 /* Set default decoding parameters. */ 445 s->param_presence_flags = 0xff; 446 s->num_primitive_matrices = 0; 447 s->blocksize = 8; 448 s->lossless_check_data = 0; 449 450 memset(s->output_shift , 0, sizeof(s->output_shift )); 451 memset(s->quant_step_size, 0, sizeof(s->quant_step_size)); 452 453 for (ch = s->min_channel; ch <= s->max_channel; ch++) { 454 ChannelParams *cp = &s->channel_params[ch]; 455 cp->filter_params[FIR].order = 0; 456 cp->filter_params[IIR].order = 0; 457 cp->filter_params[FIR].shift = 0; 458 cp->filter_params[IIR].shift = 0; 459 460 /* Default audio coding is 24-bit raw PCM. */ 461 cp->huff_offset = 0; 462 cp->sign_huff_offset = (-1) << 23; 463 cp->codebook = 0; 464 cp->huff_lsbs = 24; 465 } 466 467 if (substr == m->max_decoded_substream) 468 m->avctx->channels = s->max_matrix_channel + 1; 469 470 return 0; 471} 472 473/** Read parameters for one of the prediction filters. */ 474 475static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp, 476 unsigned int substr, unsigned int channel, 477 unsigned int filter) 478{ 479 SubStream *s = &m->substream[substr]; 480 FilterParams *fp = &s->channel_params[channel].filter_params[filter]; 481 const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER; 482 const char fchar = filter ? 'I' : 'F'; 483 int i, order; 484 485 // Filter is 0 for FIR, 1 for IIR. 486 assert(filter < 2); 487 488 if (m->filter_changed[channel][filter]++ > 1) { 489 av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n"); 490 return -1; 491 } 492 493 order = get_bits(gbp, 4); 494 if (order > max_order) { 495 av_log(m->avctx, AV_LOG_ERROR, 496 "%cIR filter order %d is greater than maximum %d.\n", 497 fchar, order, max_order); 498 return -1; 499 } 500 fp->order = order; 501 502 if (order > 0) { 503 int32_t *fcoeff = s->channel_params[channel].coeff[filter]; 504 int coeff_bits, coeff_shift; 505 506 fp->shift = get_bits(gbp, 4); 507 508 coeff_bits = get_bits(gbp, 5); 509 coeff_shift = get_bits(gbp, 3); 510 if (coeff_bits < 1 || coeff_bits > 16) { 511 av_log(m->avctx, AV_LOG_ERROR, 512 "%cIR filter coeff_bits must be between 1 and 16.\n", 513 fchar); 514 return -1; 515 } 516 if (coeff_bits + coeff_shift > 16) { 517 av_log(m->avctx, AV_LOG_ERROR, 518 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n", 519 fchar); 520 return -1; 521 } 522 523 for (i = 0; i < order; i++) 524 fcoeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift; 525 526 if (get_bits1(gbp)) { 527 int state_bits, state_shift; 528 529 if (filter == FIR) { 530 av_log(m->avctx, AV_LOG_ERROR, 531 "FIR filter has state data specified.\n"); 532 return -1; 533 } 534 535 state_bits = get_bits(gbp, 4); 536 state_shift = get_bits(gbp, 4); 537 538 /* TODO: Check validity of state data. */ 539 540 for (i = 0; i < order; i++) 541 fp->state[i] = get_sbits(gbp, state_bits) << state_shift; 542 } 543 } 544 545 return 0; 546} 547 548/** Read parameters for primitive matrices. */ 549 550static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp) 551{ 552 SubStream *s = &m->substream[substr]; 553 unsigned int mat, ch; 554 const int max_primitive_matrices = m->avctx->codec_id == CODEC_ID_MLP 555 ? MAX_MATRICES_MLP 556 : MAX_MATRICES_TRUEHD; 557 558 if (m->matrix_changed++ > 1) { 559 av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n"); 560 return -1; 561 } 562 563 s->num_primitive_matrices = get_bits(gbp, 4); 564 565 if (s->num_primitive_matrices > max_primitive_matrices) { 566 av_log(m->avctx, AV_LOG_ERROR, 567 "Number of primitive matrices cannot be greater than %d.\n", 568 max_primitive_matrices); 569 return -1; 570 } 571 572 for (mat = 0; mat < s->num_primitive_matrices; mat++) { 573 int frac_bits, max_chan; 574 s->matrix_out_ch[mat] = get_bits(gbp, 4); 575 frac_bits = get_bits(gbp, 4); 576 s->lsb_bypass [mat] = get_bits1(gbp); 577 578 if (s->matrix_out_ch[mat] > s->max_matrix_channel) { 579 av_log(m->avctx, AV_LOG_ERROR, 580 "Invalid channel %d specified as output from matrix.\n", 581 s->matrix_out_ch[mat]); 582 return -1; 583 } 584 if (frac_bits > 14) { 585 av_log(m->avctx, AV_LOG_ERROR, 586 "Too many fractional bits specified.\n"); 587 return -1; 588 } 589 590 max_chan = s->max_matrix_channel; 591 if (!s->noise_type) 592 max_chan+=2; 593 594 for (ch = 0; ch <= max_chan; ch++) { 595 int coeff_val = 0; 596 if (get_bits1(gbp)) 597 coeff_val = get_sbits(gbp, frac_bits + 2); 598 599 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits); 600 } 601 602 if (s->noise_type) 603 s->matrix_noise_shift[mat] = get_bits(gbp, 4); 604 else 605 s->matrix_noise_shift[mat] = 0; 606 } 607 608 return 0; 609} 610 611/** Read channel parameters. */ 612 613static int read_channel_params(MLPDecodeContext *m, unsigned int substr, 614 GetBitContext *gbp, unsigned int ch) 615{ 616 SubStream *s = &m->substream[substr]; 617 ChannelParams *cp = &s->channel_params[ch]; 618 FilterParams *fir = &cp->filter_params[FIR]; 619 FilterParams *iir = &cp->filter_params[IIR]; 620 621 if (s->param_presence_flags & PARAM_FIR) 622 if (get_bits1(gbp)) 623 if (read_filter_params(m, gbp, substr, ch, FIR) < 0) 624 return -1; 625 626 if (s->param_presence_flags & PARAM_IIR) 627 if (get_bits1(gbp)) 628 if (read_filter_params(m, gbp, substr, ch, IIR) < 0) 629 return -1; 630 631 if (fir->order + iir->order > 8) { 632 av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n"); 633 return -1; 634 } 635 636 if (fir->order && iir->order && 637 fir->shift != iir->shift) { 638 av_log(m->avctx, AV_LOG_ERROR, 639 "FIR and IIR filters must use the same precision.\n"); 640 return -1; 641 } 642 /* The FIR and IIR filters must have the same precision. 643 * To simplify the filtering code, only the precision of the 644 * FIR filter is considered. If only the IIR filter is employed, 645 * the FIR filter precision is set to that of the IIR filter, so 646 * that the filtering code can use it. */ 647 if (!fir->order && iir->order) 648 fir->shift = iir->shift; 649 650 if (s->param_presence_flags & PARAM_HUFFOFFSET) 651 if (get_bits1(gbp)) 652 cp->huff_offset = get_sbits(gbp, 15); 653 654 cp->codebook = get_bits(gbp, 2); 655 cp->huff_lsbs = get_bits(gbp, 5); 656 657 if (cp->huff_lsbs > 24) { 658 av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n"); 659 return -1; 660 } 661 662 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch); 663 664 return 0; 665} 666 667/** Read decoding parameters that change more often than those in the restart 668 * header. */ 669 670static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp, 671 unsigned int substr) 672{ 673 SubStream *s = &m->substream[substr]; 674 unsigned int ch; 675 676 if (s->param_presence_flags & PARAM_PRESENCE) 677 if (get_bits1(gbp)) 678 s->param_presence_flags = get_bits(gbp, 8); 679 680 if (s->param_presence_flags & PARAM_BLOCKSIZE) 681 if (get_bits1(gbp)) { 682 s->blocksize = get_bits(gbp, 9); 683 if (s->blocksize < 8 || s->blocksize > m->access_unit_size) { 684 av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize."); 685 s->blocksize = 0; 686 return -1; 687 } 688 } 689 690 if (s->param_presence_flags & PARAM_MATRIX) 691 if (get_bits1(gbp)) 692 if (read_matrix_params(m, substr, gbp) < 0) 693 return -1; 694 695 if (s->param_presence_flags & PARAM_OUTSHIFT) 696 if (get_bits1(gbp)) 697 for (ch = 0; ch <= s->max_matrix_channel; ch++) 698 s->output_shift[ch] = get_sbits(gbp, 4); 699 700 if (s->param_presence_flags & PARAM_QUANTSTEP) 701 if (get_bits1(gbp)) 702 for (ch = 0; ch <= s->max_channel; ch++) { 703 ChannelParams *cp = &s->channel_params[ch]; 704 705 s->quant_step_size[ch] = get_bits(gbp, 4); 706 707 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch); 708 } 709 710 for (ch = s->min_channel; ch <= s->max_channel; ch++) 711 if (get_bits1(gbp)) 712 if (read_channel_params(m, substr, gbp, ch) < 0) 713 return -1; 714 715 return 0; 716} 717 718#define MSB_MASK(bits) (-1u << bits) 719 720/** Generate PCM samples using the prediction filters and residual values 721 * read from the data stream, and update the filter state. */ 722 723static void filter_channel(MLPDecodeContext *m, unsigned int substr, 724 unsigned int channel) 725{ 726 SubStream *s = &m->substream[substr]; 727 const int32_t *fircoeff = s->channel_params[channel].coeff[FIR]; 728 int32_t state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FIR_ORDER]; 729 int32_t *firbuf = state_buffer[FIR] + MAX_BLOCKSIZE; 730 int32_t *iirbuf = state_buffer[IIR] + MAX_BLOCKSIZE; 731 FilterParams *fir = &s->channel_params[channel].filter_params[FIR]; 732 FilterParams *iir = &s->channel_params[channel].filter_params[IIR]; 733 unsigned int filter_shift = fir->shift; 734 int32_t mask = MSB_MASK(s->quant_step_size[channel]); 735 736 memcpy(firbuf, fir->state, MAX_FIR_ORDER * sizeof(int32_t)); 737 memcpy(iirbuf, iir->state, MAX_IIR_ORDER * sizeof(int32_t)); 738 739 m->dsp.mlp_filter_channel(firbuf, fircoeff, 740 fir->order, iir->order, 741 filter_shift, mask, s->blocksize, 742 &m->sample_buffer[s->blockpos][channel]); 743 744 memcpy(fir->state, firbuf - s->blocksize, MAX_FIR_ORDER * sizeof(int32_t)); 745 memcpy(iir->state, iirbuf - s->blocksize, MAX_IIR_ORDER * sizeof(int32_t)); 746} 747 748/** Read a block of PCM residual data (or actual if no filtering active). */ 749 750static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp, 751 unsigned int substr) 752{ 753 SubStream *s = &m->substream[substr]; 754 unsigned int i, ch, expected_stream_pos = 0; 755 756 if (s->data_check_present) { 757 expected_stream_pos = get_bits_count(gbp); 758 expected_stream_pos += get_bits(gbp, 16); 759 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features " 760 "we have not tested yet. %s\n", sample_message); 761 } 762 763 if (s->blockpos + s->blocksize > m->access_unit_size) { 764 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n"); 765 return -1; 766 } 767 768 memset(&m->bypassed_lsbs[s->blockpos][0], 0, 769 s->blocksize * sizeof(m->bypassed_lsbs[0])); 770 771 for (i = 0; i < s->blocksize; i++) 772 if (read_huff_channels(m, gbp, substr, i) < 0) 773 return -1; 774 775 for (ch = s->min_channel; ch <= s->max_channel; ch++) 776 filter_channel(m, substr, ch); 777 778 s->blockpos += s->blocksize; 779 780 if (s->data_check_present) { 781 if (get_bits_count(gbp) != expected_stream_pos) 782 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n"); 783 skip_bits(gbp, 8); 784 } 785 786 return 0; 787} 788 789/** Data table used for TrueHD noise generation function. */ 790 791static const int8_t noise_table[256] = { 792 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2, 793 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62, 794 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5, 795 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40, 796 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34, 797 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30, 798 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36, 799 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69, 800 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24, 801 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20, 802 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23, 803 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8, 804 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40, 805 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37, 806 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52, 807 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70, 808}; 809 810/** Noise generation functions. 811 * I'm not sure what these are for - they seem to be some kind of pseudorandom 812 * sequence generators, used to generate noise data which is used when the 813 * channels are rematrixed. I'm not sure if they provide a practical benefit 814 * to compression, or just obfuscate the decoder. Are they for some kind of 815 * dithering? */ 816 817/** Generate two channels of noise, used in the matrix when 818 * restart sync word == 0x31ea. */ 819 820static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr) 821{ 822 SubStream *s = &m->substream[substr]; 823 unsigned int i; 824 uint32_t seed = s->noisegen_seed; 825 unsigned int maxchan = s->max_matrix_channel; 826 827 for (i = 0; i < s->blockpos; i++) { 828 uint16_t seed_shr7 = seed >> 7; 829 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift; 830 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift; 831 832 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5); 833 } 834 835 s->noisegen_seed = seed; 836} 837 838/** Generate a block of noise, used when restart sync word == 0x31eb. */ 839 840static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr) 841{ 842 SubStream *s = &m->substream[substr]; 843 unsigned int i; 844 uint32_t seed = s->noisegen_seed; 845 846 for (i = 0; i < m->access_unit_size_pow2; i++) { 847 uint8_t seed_shr15 = seed >> 15; 848 m->noise_buffer[i] = noise_table[seed_shr15]; 849 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5); 850 } 851 852 s->noisegen_seed = seed; 853} 854 855 856/** Apply the channel matrices in turn to reconstruct the original audio 857 * samples. */ 858 859static void rematrix_channels(MLPDecodeContext *m, unsigned int substr) 860{ 861 SubStream *s = &m->substream[substr]; 862 unsigned int mat, src_ch, i; 863 unsigned int maxchan; 864 865 maxchan = s->max_matrix_channel; 866 if (!s->noise_type) { 867 generate_2_noise_channels(m, substr); 868 maxchan += 2; 869 } else { 870 fill_noise_buffer(m, substr); 871 } 872 873 for (mat = 0; mat < s->num_primitive_matrices; mat++) { 874 int matrix_noise_shift = s->matrix_noise_shift[mat]; 875 unsigned int dest_ch = s->matrix_out_ch[mat]; 876 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]); 877 int32_t *coeffs = s->matrix_coeff[mat]; 878 int index = s->num_primitive_matrices - mat; 879 int index2 = 2 * index + 1; 880 881 /* TODO: DSPContext? */ 882 883 for (i = 0; i < s->blockpos; i++) { 884 int32_t bypassed_lsb = m->bypassed_lsbs[i][mat]; 885 int32_t *samples = m->sample_buffer[i]; 886 int64_t accum = 0; 887 888 for (src_ch = 0; src_ch <= maxchan; src_ch++) 889 accum += (int64_t) samples[src_ch] * coeffs[src_ch]; 890 891 if (matrix_noise_shift) { 892 index &= m->access_unit_size_pow2 - 1; 893 accum += m->noise_buffer[index] << (matrix_noise_shift + 7); 894 index += index2; 895 } 896 897 samples[dest_ch] = ((accum >> 14) & mask) + bypassed_lsb; 898 } 899 } 900} 901 902/** Write the audio data into the output buffer. */ 903 904static int output_data_internal(MLPDecodeContext *m, unsigned int substr, 905 uint8_t *data, unsigned int *data_size, int is32) 906{ 907 SubStream *s = &m->substream[substr]; 908 unsigned int i, out_ch = 0; 909 int32_t *data_32 = (int32_t*) data; 910 int16_t *data_16 = (int16_t*) data; 911 912 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2)) 913 return -1; 914 915 for (i = 0; i < s->blockpos; i++) { 916 for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) { 917 int mat_ch = s->ch_assign[out_ch]; 918 int32_t sample = m->sample_buffer[i][mat_ch] 919 << s->output_shift[mat_ch]; 920 s->lossless_check_data ^= (sample & 0xffffff) << mat_ch; 921 if (is32) *data_32++ = sample << 8; 922 else *data_16++ = sample >> 8; 923 } 924 } 925 926 *data_size = i * out_ch * (is32 ? 4 : 2); 927 928 return 0; 929} 930 931static int output_data(MLPDecodeContext *m, unsigned int substr, 932 uint8_t *data, unsigned int *data_size) 933{ 934 if (m->avctx->sample_fmt == SAMPLE_FMT_S32) 935 return output_data_internal(m, substr, data, data_size, 1); 936 else 937 return output_data_internal(m, substr, data, data_size, 0); 938} 939 940 941/** Read an access unit from the stream. 942 * Returns < 0 on error, 0 if not enough data is present in the input stream 943 * otherwise returns the number of bytes consumed. */ 944 945static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size, 946 AVPacket *avpkt) 947{ 948 const uint8_t *buf = avpkt->data; 949 int buf_size = avpkt->size; 950 MLPDecodeContext *m = avctx->priv_data; 951 GetBitContext gb; 952 unsigned int length, substr; 953 unsigned int substream_start; 954 unsigned int header_size = 4; 955 unsigned int substr_header_size = 0; 956 uint8_t substream_parity_present[MAX_SUBSTREAMS]; 957 uint16_t substream_data_len[MAX_SUBSTREAMS]; 958 uint8_t parity_bits; 959 960 if (buf_size < 4) 961 return 0; 962 963 length = (AV_RB16(buf) & 0xfff) * 2; 964 965 if (length < 4 || length > buf_size) 966 return -1; 967 968 init_get_bits(&gb, (buf + 4), (length - 4) * 8); 969 970 m->is_major_sync_unit = 0; 971 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) { 972 if (read_major_sync(m, &gb) < 0) 973 goto error; 974 m->is_major_sync_unit = 1; 975 header_size += 28; 976 } 977 978 if (!m->params_valid) { 979 av_log(m->avctx, AV_LOG_WARNING, 980 "Stream parameters not seen; skipping frame.\n"); 981 *data_size = 0; 982 return length; 983 } 984 985 substream_start = 0; 986 987 for (substr = 0; substr < m->num_substreams; substr++) { 988 int extraword_present, checkdata_present, end, nonrestart_substr; 989 990 extraword_present = get_bits1(&gb); 991 nonrestart_substr = get_bits1(&gb); 992 checkdata_present = get_bits1(&gb); 993 skip_bits1(&gb); 994 995 end = get_bits(&gb, 12) * 2; 996 997 substr_header_size += 2; 998 999 if (extraword_present) { 1000 if (m->avctx->codec_id == CODEC_ID_MLP) { 1001 av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n"); 1002 goto error; 1003 } 1004 skip_bits(&gb, 16); 1005 substr_header_size += 2; 1006 } 1007 1008 if (!(nonrestart_substr ^ m->is_major_sync_unit)) { 1009 av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n"); 1010 goto error; 1011 } 1012 1013 if (end + header_size + substr_header_size > length) { 1014 av_log(m->avctx, AV_LOG_ERROR, 1015 "Indicated length of substream %d data goes off end of " 1016 "packet.\n", substr); 1017 end = length - header_size - substr_header_size; 1018 } 1019 1020 if (end < substream_start) { 1021 av_log(avctx, AV_LOG_ERROR, 1022 "Indicated end offset of substream %d data " 1023 "is smaller than calculated start offset.\n", 1024 substr); 1025 goto error; 1026 } 1027 1028 if (substr > m->max_decoded_substream) 1029 continue; 1030 1031 substream_parity_present[substr] = checkdata_present; 1032 substream_data_len[substr] = end - substream_start; 1033 substream_start = end; 1034 } 1035 1036 parity_bits = ff_mlp_calculate_parity(buf, 4); 1037 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size); 1038 1039 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) { 1040 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n"); 1041 goto error; 1042 } 1043 1044 buf += header_size + substr_header_size; 1045 1046 for (substr = 0; substr <= m->max_decoded_substream; substr++) { 1047 SubStream *s = &m->substream[substr]; 1048 init_get_bits(&gb, buf, substream_data_len[substr] * 8); 1049 1050 m->matrix_changed = 0; 1051 memset(m->filter_changed, 0, sizeof(m->filter_changed)); 1052 1053 s->blockpos = 0; 1054 do { 1055 if (get_bits1(&gb)) { 1056 if (get_bits1(&gb)) { 1057 /* A restart header should be present. */ 1058 if (read_restart_header(m, &gb, buf, substr) < 0) 1059 goto next_substr; 1060 s->restart_seen = 1; 1061 } 1062 1063 if (!s->restart_seen) 1064 goto next_substr; 1065 if (read_decoding_params(m, &gb, substr) < 0) 1066 goto next_substr; 1067 } 1068 1069 if (!s->restart_seen) 1070 goto next_substr; 1071 1072 if (read_block_data(m, &gb, substr) < 0) 1073 return -1; 1074 1075 if (get_bits_count(&gb) >= substream_data_len[substr] * 8) 1076 goto substream_length_mismatch; 1077 1078 } while (!get_bits1(&gb)); 1079 1080 skip_bits(&gb, (-get_bits_count(&gb)) & 15); 1081 1082 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) { 1083 int shorten_by; 1084 1085 if (get_bits(&gb, 16) != 0xD234) 1086 return -1; 1087 1088 shorten_by = get_bits(&gb, 16); 1089 if (m->avctx->codec_id == CODEC_ID_TRUEHD && shorten_by & 0x2000) 1090 s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos); 1091 else if (m->avctx->codec_id == CODEC_ID_MLP && shorten_by != 0xD234) 1092 return -1; 1093 1094 if (substr == m->max_decoded_substream) 1095 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n"); 1096 } 1097 1098 if (substream_parity_present[substr]) { 1099 uint8_t parity, checksum; 1100 1101 if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16) 1102 goto substream_length_mismatch; 1103 1104 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2); 1105 checksum = ff_mlp_checksum8 (buf, substream_data_len[substr] - 2); 1106 1107 if ((get_bits(&gb, 8) ^ parity) != 0xa9 ) 1108 av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr); 1109 if ( get_bits(&gb, 8) != checksum) 1110 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n" , substr); 1111 } 1112 1113 if (substream_data_len[substr] * 8 != get_bits_count(&gb)) 1114 goto substream_length_mismatch; 1115 1116next_substr: 1117 if (!s->restart_seen) 1118 av_log(m->avctx, AV_LOG_ERROR, 1119 "No restart header present in substream %d.\n", substr); 1120 1121 buf += substream_data_len[substr]; 1122 } 1123 1124 rematrix_channels(m, m->max_decoded_substream); 1125 1126 if (output_data(m, m->max_decoded_substream, data, data_size) < 0) 1127 return -1; 1128 1129 return length; 1130 1131substream_length_mismatch: 1132 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr); 1133 return -1; 1134 1135error: 1136 m->params_valid = 0; 1137 return -1; 1138} 1139 1140AVCodec mlp_decoder = { 1141 "mlp", 1142 AVMEDIA_TYPE_AUDIO, 1143 CODEC_ID_MLP, 1144 sizeof(MLPDecodeContext), 1145 mlp_decode_init, 1146 NULL, 1147 NULL, 1148 read_access_unit, 1149 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"), 1150}; 1151 1152#if CONFIG_TRUEHD_DECODER 1153AVCodec truehd_decoder = { 1154 "truehd", 1155 AVMEDIA_TYPE_AUDIO, 1156 CODEC_ID_TRUEHD, 1157 sizeof(MLPDecodeContext), 1158 mlp_decode_init, 1159 NULL, 1160 NULL, 1161 read_access_unit, 1162 .long_name = NULL_IF_CONFIG_SMALL("TrueHD"), 1163}; 1164#endif /* CONFIG_TRUEHD_DECODER */ 1165