1/* 2 * QDM2 compatible decoder 3 * Copyright (c) 2003 Ewald Snel 4 * Copyright (c) 2005 Benjamin Larsson 5 * Copyright (c) 2005 Alex Beregszaszi 6 * Copyright (c) 2005 Roberto Togni 7 * 8 * This file is part of FFmpeg. 9 * 10 * FFmpeg is free software; you can redistribute it and/or 11 * modify it under the terms of the GNU Lesser General Public 12 * License as published by the Free Software Foundation; either 13 * version 2.1 of the License, or (at your option) any later version. 14 * 15 * FFmpeg is distributed in the hope that it will be useful, 16 * but WITHOUT ANY WARRANTY; without even the implied warranty of 17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 18 * Lesser General Public License for more details. 19 * 20 * You should have received a copy of the GNU Lesser General Public 21 * License along with FFmpeg; if not, write to the Free Software 22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 23 */ 24 25/** 26 * @file 27 * QDM2 decoder 28 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni 29 * 30 * The decoder is not perfect yet, there are still some distortions 31 * especially on files encoded with 16 or 8 subbands. 32 */ 33 34#include <math.h> 35#include <stddef.h> 36#include <stdio.h> 37 38#define BITSTREAM_READER_LE 39#include "libavutil/channel_layout.h" 40#include "avcodec.h" 41#include "get_bits.h" 42#include "internal.h" 43#include "rdft.h" 44#include "mpegaudiodsp.h" 45#include "mpegaudio.h" 46 47#include "qdm2data.h" 48#include "qdm2_tablegen.h" 49 50#undef NDEBUG 51#include <assert.h> 52 53 54#define QDM2_LIST_ADD(list, size, packet) \ 55do { \ 56 if (size > 0) { \ 57 list[size - 1].next = &list[size]; \ 58 } \ 59 list[size].packet = packet; \ 60 list[size].next = NULL; \ 61 size++; \ 62} while(0) 63 64// Result is 8, 16 or 30 65#define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling)) 66 67#define FIX_NOISE_IDX(noise_idx) \ 68 if ((noise_idx) >= 3840) \ 69 (noise_idx) -= 3840; \ 70 71#define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)]) 72 73#define SAMPLES_NEEDED \ 74 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n"); 75 76#define SAMPLES_NEEDED_2(why) \ 77 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why); 78 79#define QDM2_MAX_FRAME_SIZE 512 80 81typedef int8_t sb_int8_array[2][30][64]; 82 83/** 84 * Subpacket 85 */ 86typedef struct { 87 int type; ///< subpacket type 88 unsigned int size; ///< subpacket size 89 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy) 90} QDM2SubPacket; 91 92/** 93 * A node in the subpacket list 94 */ 95typedef struct QDM2SubPNode { 96 QDM2SubPacket *packet; ///< packet 97 struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node 98} QDM2SubPNode; 99 100typedef struct { 101 float re; 102 float im; 103} QDM2Complex; 104 105typedef struct { 106 float level; 107 QDM2Complex *complex; 108 const float *table; 109 int phase; 110 int phase_shift; 111 int duration; 112 short time_index; 113 short cutoff; 114} FFTTone; 115 116typedef struct { 117 int16_t sub_packet; 118 uint8_t channel; 119 int16_t offset; 120 int16_t exp; 121 uint8_t phase; 122} FFTCoefficient; 123 124typedef struct { 125 DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256]; 126} QDM2FFT; 127 128/** 129 * QDM2 decoder context 130 */ 131typedef struct { 132 /// Parameters from codec header, do not change during playback 133 int nb_channels; ///< number of channels 134 int channels; ///< number of channels 135 int group_size; ///< size of frame group (16 frames per group) 136 int fft_size; ///< size of FFT, in complex numbers 137 int checksum_size; ///< size of data block, used also for checksum 138 139 /// Parameters built from header parameters, do not change during playback 140 int group_order; ///< order of frame group 141 int fft_order; ///< order of FFT (actually fftorder+1) 142 int frame_size; ///< size of data frame 143 int frequency_range; 144 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */ 145 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2 146 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4) 147 148 /// Packets and packet lists 149 QDM2SubPacket sub_packets[16]; ///< the packets themselves 150 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets 151 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list 152 int sub_packets_B; ///< number of packets on 'B' list 153 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors? 154 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets 155 156 /// FFT and tones 157 FFTTone fft_tones[1000]; 158 int fft_tone_start; 159 int fft_tone_end; 160 FFTCoefficient fft_coefs[1000]; 161 int fft_coefs_index; 162 int fft_coefs_min_index[5]; 163 int fft_coefs_max_index[5]; 164 int fft_level_exp[6]; 165 RDFTContext rdft_ctx; 166 QDM2FFT fft; 167 168 /// I/O data 169 const uint8_t *compressed_data; 170 int compressed_size; 171 float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2]; 172 173 /// Synthesis filter 174 MPADSPContext mpadsp; 175 DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2]; 176 int synth_buf_offset[MPA_MAX_CHANNELS]; 177 DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT]; 178 DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; 179 180 /// Mixed temporary data used in decoding 181 float tone_level[MPA_MAX_CHANNELS][30][64]; 182 int8_t coding_method[MPA_MAX_CHANNELS][30][64]; 183 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8]; 184 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8]; 185 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8]; 186 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8]; 187 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26]; 188 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64]; 189 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]; 190 191 // Flags 192 int has_errors; ///< packet has errors 193 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type 194 int do_synth_filter; ///< used to perform or skip synthesis filter 195 196 int sub_packet; 197 int noise_idx; ///< index for dithering noise table 198} QDM2Context; 199 200 201static VLC vlc_tab_level; 202static VLC vlc_tab_diff; 203static VLC vlc_tab_run; 204static VLC fft_level_exp_alt_vlc; 205static VLC fft_level_exp_vlc; 206static VLC fft_stereo_exp_vlc; 207static VLC fft_stereo_phase_vlc; 208static VLC vlc_tab_tone_level_idx_hi1; 209static VLC vlc_tab_tone_level_idx_mid; 210static VLC vlc_tab_tone_level_idx_hi2; 211static VLC vlc_tab_type30; 212static VLC vlc_tab_type34; 213static VLC vlc_tab_fft_tone_offset[5]; 214 215static const uint16_t qdm2_vlc_offs[] = { 216 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838, 217}; 218 219static const int switchtable[23] = { 220 0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4 221}; 222 223static av_cold void qdm2_init_vlc(void) 224{ 225 static VLC_TYPE qdm2_table[3838][2]; 226 227 vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]]; 228 vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0]; 229 init_vlc(&vlc_tab_level, 8, 24, 230 vlc_tab_level_huffbits, 1, 1, 231 vlc_tab_level_huffcodes, 2, 2, 232 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 233 234 vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]]; 235 vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1]; 236 init_vlc(&vlc_tab_diff, 8, 37, 237 vlc_tab_diff_huffbits, 1, 1, 238 vlc_tab_diff_huffcodes, 2, 2, 239 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 240 241 vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]]; 242 vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2]; 243 init_vlc(&vlc_tab_run, 5, 6, 244 vlc_tab_run_huffbits, 1, 1, 245 vlc_tab_run_huffcodes, 1, 1, 246 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 247 248 fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]]; 249 fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - 250 qdm2_vlc_offs[3]; 251 init_vlc(&fft_level_exp_alt_vlc, 8, 28, 252 fft_level_exp_alt_huffbits, 1, 1, 253 fft_level_exp_alt_huffcodes, 2, 2, 254 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 255 256 fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]]; 257 fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4]; 258 init_vlc(&fft_level_exp_vlc, 8, 20, 259 fft_level_exp_huffbits, 1, 1, 260 fft_level_exp_huffcodes, 2, 2, 261 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 262 263 fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]]; 264 fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - 265 qdm2_vlc_offs[5]; 266 init_vlc(&fft_stereo_exp_vlc, 6, 7, 267 fft_stereo_exp_huffbits, 1, 1, 268 fft_stereo_exp_huffcodes, 1, 1, 269 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 270 271 fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]]; 272 fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - 273 qdm2_vlc_offs[6]; 274 init_vlc(&fft_stereo_phase_vlc, 6, 9, 275 fft_stereo_phase_huffbits, 1, 1, 276 fft_stereo_phase_huffcodes, 1, 1, 277 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 278 279 vlc_tab_tone_level_idx_hi1.table = 280 &qdm2_table[qdm2_vlc_offs[7]]; 281 vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - 282 qdm2_vlc_offs[7]; 283 init_vlc(&vlc_tab_tone_level_idx_hi1, 8, 20, 284 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1, 285 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, 286 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 287 288 vlc_tab_tone_level_idx_mid.table = 289 &qdm2_table[qdm2_vlc_offs[8]]; 290 vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - 291 qdm2_vlc_offs[8]; 292 init_vlc(&vlc_tab_tone_level_idx_mid, 8, 24, 293 vlc_tab_tone_level_idx_mid_huffbits, 1, 1, 294 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, 295 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 296 297 vlc_tab_tone_level_idx_hi2.table = 298 &qdm2_table[qdm2_vlc_offs[9]]; 299 vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - 300 qdm2_vlc_offs[9]; 301 init_vlc(&vlc_tab_tone_level_idx_hi2, 8, 24, 302 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1, 303 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, 304 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 305 306 vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]]; 307 vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10]; 308 init_vlc(&vlc_tab_type30, 6, 9, 309 vlc_tab_type30_huffbits, 1, 1, 310 vlc_tab_type30_huffcodes, 1, 1, 311 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 312 313 vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]]; 314 vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11]; 315 init_vlc(&vlc_tab_type34, 5, 10, 316 vlc_tab_type34_huffbits, 1, 1, 317 vlc_tab_type34_huffcodes, 1, 1, 318 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 319 320 vlc_tab_fft_tone_offset[0].table = 321 &qdm2_table[qdm2_vlc_offs[12]]; 322 vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - 323 qdm2_vlc_offs[12]; 324 init_vlc(&vlc_tab_fft_tone_offset[0], 8, 23, 325 vlc_tab_fft_tone_offset_0_huffbits, 1, 1, 326 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, 327 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 328 329 vlc_tab_fft_tone_offset[1].table = 330 &qdm2_table[qdm2_vlc_offs[13]]; 331 vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - 332 qdm2_vlc_offs[13]; 333 init_vlc(&vlc_tab_fft_tone_offset[1], 8, 28, 334 vlc_tab_fft_tone_offset_1_huffbits, 1, 1, 335 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, 336 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 337 338 vlc_tab_fft_tone_offset[2].table = 339 &qdm2_table[qdm2_vlc_offs[14]]; 340 vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - 341 qdm2_vlc_offs[14]; 342 init_vlc(&vlc_tab_fft_tone_offset[2], 8, 32, 343 vlc_tab_fft_tone_offset_2_huffbits, 1, 1, 344 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, 345 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 346 347 vlc_tab_fft_tone_offset[3].table = 348 &qdm2_table[qdm2_vlc_offs[15]]; 349 vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - 350 qdm2_vlc_offs[15]; 351 init_vlc(&vlc_tab_fft_tone_offset[3], 8, 35, 352 vlc_tab_fft_tone_offset_3_huffbits, 1, 1, 353 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, 354 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 355 356 vlc_tab_fft_tone_offset[4].table = 357 &qdm2_table[qdm2_vlc_offs[16]]; 358 vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - 359 qdm2_vlc_offs[16]; 360 init_vlc(&vlc_tab_fft_tone_offset[4], 8, 38, 361 vlc_tab_fft_tone_offset_4_huffbits, 1, 1, 362 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, 363 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 364} 365 366static int qdm2_get_vlc(GetBitContext *gb, VLC *vlc, int flag, int depth) 367{ 368 int value; 369 370 value = get_vlc2(gb, vlc->table, vlc->bits, depth); 371 372 /* stage-2, 3 bits exponent escape sequence */ 373 if (value-- == 0) 374 value = get_bits(gb, get_bits(gb, 3) + 1); 375 376 /* stage-3, optional */ 377 if (flag) { 378 int tmp; 379 380 if (value >= 60) { 381 av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value); 382 return 0; 383 } 384 385 tmp= vlc_stage3_values[value]; 386 387 if ((value & ~3) > 0) 388 tmp += get_bits(gb, (value >> 2)); 389 value = tmp; 390 } 391 392 return value; 393} 394 395static int qdm2_get_se_vlc(VLC *vlc, GetBitContext *gb, int depth) 396{ 397 int value = qdm2_get_vlc(gb, vlc, 0, depth); 398 399 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1); 400} 401 402/** 403 * QDM2 checksum 404 * 405 * @param data pointer to data to be checksum'ed 406 * @param length data length 407 * @param value checksum value 408 * 409 * @return 0 if checksum is OK 410 */ 411static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value) 412{ 413 int i; 414 415 for (i = 0; i < length; i++) 416 value -= data[i]; 417 418 return (uint16_t)(value & 0xffff); 419} 420 421/** 422 * Fill a QDM2SubPacket structure with packet type, size, and data pointer. 423 * 424 * @param gb bitreader context 425 * @param sub_packet packet under analysis 426 */ 427static void qdm2_decode_sub_packet_header(GetBitContext *gb, 428 QDM2SubPacket *sub_packet) 429{ 430 sub_packet->type = get_bits(gb, 8); 431 432 if (sub_packet->type == 0) { 433 sub_packet->size = 0; 434 sub_packet->data = NULL; 435 } else { 436 sub_packet->size = get_bits(gb, 8); 437 438 if (sub_packet->type & 0x80) { 439 sub_packet->size <<= 8; 440 sub_packet->size |= get_bits(gb, 8); 441 sub_packet->type &= 0x7f; 442 } 443 444 if (sub_packet->type == 0x7f) 445 sub_packet->type |= (get_bits(gb, 8) << 8); 446 447 // FIXME: this depends on bitreader-internal data 448 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; 449 } 450 451 av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n", 452 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8); 453} 454 455/** 456 * Return node pointer to first packet of requested type in list. 457 * 458 * @param list list of subpackets to be scanned 459 * @param type type of searched subpacket 460 * @return node pointer for subpacket if found, else NULL 461 */ 462static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list, 463 int type) 464{ 465 while (list != NULL && list->packet != NULL) { 466 if (list->packet->type == type) 467 return list; 468 list = list->next; 469 } 470 return NULL; 471} 472 473/** 474 * Replace 8 elements with their average value. 475 * Called by qdm2_decode_superblock before starting subblock decoding. 476 * 477 * @param q context 478 */ 479static void average_quantized_coeffs(QDM2Context *q) 480{ 481 int i, j, n, ch, sum; 482 483 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; 484 485 for (ch = 0; ch < q->nb_channels; ch++) 486 for (i = 0; i < n; i++) { 487 sum = 0; 488 489 for (j = 0; j < 8; j++) 490 sum += q->quantized_coeffs[ch][i][j]; 491 492 sum /= 8; 493 if (sum > 0) 494 sum--; 495 496 for (j = 0; j < 8; j++) 497 q->quantized_coeffs[ch][i][j] = sum; 498 } 499} 500 501/** 502 * Build subband samples with noise weighted by q->tone_level. 503 * Called by synthfilt_build_sb_samples. 504 * 505 * @param q context 506 * @param sb subband index 507 */ 508static void build_sb_samples_from_noise(QDM2Context *q, int sb) 509{ 510 int ch, j; 511 512 FIX_NOISE_IDX(q->noise_idx); 513 514 if (!q->nb_channels) 515 return; 516 517 for (ch = 0; ch < q->nb_channels; ch++) { 518 for (j = 0; j < 64; j++) { 519 q->sb_samples[ch][j * 2][sb] = 520 SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j]; 521 q->sb_samples[ch][j * 2 + 1][sb] = 522 SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j]; 523 } 524 } 525} 526 527/** 528 * Called while processing data from subpackets 11 and 12. 529 * Used after making changes to coding_method array. 530 * 531 * @param sb subband index 532 * @param channels number of channels 533 * @param coding_method q->coding_method[0][0][0] 534 */ 535static int fix_coding_method_array(int sb, int channels, 536 sb_int8_array coding_method) 537{ 538 int j, k; 539 int ch; 540 int run, case_val; 541 542 for (ch = 0; ch < channels; ch++) { 543 for (j = 0; j < 64; ) { 544 if (coding_method[ch][sb][j] < 8) 545 return -1; 546 if ((coding_method[ch][sb][j] - 8) > 22) { 547 run = 1; 548 case_val = 8; 549 } else { 550 switch (switchtable[coding_method[ch][sb][j] - 8]) { 551 case 0: run = 10; 552 case_val = 10; 553 break; 554 case 1: run = 1; 555 case_val = 16; 556 break; 557 case 2: run = 5; 558 case_val = 24; 559 break; 560 case 3: run = 3; 561 case_val = 30; 562 break; 563 case 4: run = 1; 564 case_val = 30; 565 break; 566 case 5: run = 1; 567 case_val = 8; 568 break; 569 default: run = 1; 570 case_val = 8; 571 break; 572 } 573 } 574 for (k = 0; k < run; k++) { 575 if (j + k < 128) { 576 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) { 577 if (k > 0) { 578 SAMPLES_NEEDED 579 //not debugged, almost never used 580 memset(&coding_method[ch][sb][j + k], case_val, 581 k *sizeof(int8_t)); 582 memset(&coding_method[ch][sb][j + k], case_val, 583 3 * sizeof(int8_t)); 584 } 585 } 586 } 587 } 588 j += run; 589 } 590 } 591 return 0; 592} 593 594/** 595 * Related to synthesis filter 596 * Called by process_subpacket_10 597 * 598 * @param q context 599 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10 600 */ 601static void fill_tone_level_array(QDM2Context *q, int flag) 602{ 603 int i, sb, ch, sb_used; 604 int tmp, tab; 605 606 for (ch = 0; ch < q->nb_channels; ch++) 607 for (sb = 0; sb < 30; sb++) 608 for (i = 0; i < 8; i++) { 609 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1)) 610 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+ 611 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; 612 else 613 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; 614 if(tmp < 0) 615 tmp += 0xff; 616 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff; 617 } 618 619 sb_used = QDM2_SB_USED(q->sub_sampling); 620 621 if ((q->superblocktype_2_3 != 0) && !flag) { 622 for (sb = 0; sb < sb_used; sb++) 623 for (ch = 0; ch < q->nb_channels; ch++) 624 for (i = 0; i < 64; i++) { 625 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; 626 if (q->tone_level_idx[ch][sb][i] < 0) 627 q->tone_level[ch][sb][i] = 0; 628 else 629 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f]; 630 } 631 } else { 632 tab = q->superblocktype_2_3 ? 0 : 1; 633 for (sb = 0; sb < sb_used; sb++) { 634 if ((sb >= 4) && (sb <= 23)) { 635 for (ch = 0; ch < q->nb_channels; ch++) 636 for (i = 0; i < 64; i++) { 637 tmp = q->tone_level_idx_base[ch][sb][i / 8] - 638 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] - 639 q->tone_level_idx_mid[ch][sb - 4][i / 8] - 640 q->tone_level_idx_hi2[ch][sb - 4]; 641 q->tone_level_idx[ch][sb][i] = tmp & 0xff; 642 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 643 q->tone_level[ch][sb][i] = 0; 644 else 645 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 646 } 647 } else { 648 if (sb > 4) { 649 for (ch = 0; ch < q->nb_channels; ch++) 650 for (i = 0; i < 64; i++) { 651 tmp = q->tone_level_idx_base[ch][sb][i / 8] - 652 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] - 653 q->tone_level_idx_hi2[ch][sb - 4]; 654 q->tone_level_idx[ch][sb][i] = tmp & 0xff; 655 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 656 q->tone_level[ch][sb][i] = 0; 657 else 658 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 659 } 660 } else { 661 for (ch = 0; ch < q->nb_channels; ch++) 662 for (i = 0; i < 64; i++) { 663 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; 664 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 665 q->tone_level[ch][sb][i] = 0; 666 else 667 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 668 } 669 } 670 } 671 } 672 } 673} 674 675/** 676 * Related to synthesis filter 677 * Called by process_subpacket_11 678 * c is built with data from subpacket 11 679 * Most of this function is used only if superblock_type_2_3 == 0, 680 * never seen it in samples. 681 * 682 * @param tone_level_idx 683 * @param tone_level_idx_temp 684 * @param coding_method q->coding_method[0][0][0] 685 * @param nb_channels number of channels 686 * @param c coming from subpacket 11, passed as 8*c 687 * @param superblocktype_2_3 flag based on superblock packet type 688 * @param cm_table_select q->cm_table_select 689 */ 690static void fill_coding_method_array(sb_int8_array tone_level_idx, 691 sb_int8_array tone_level_idx_temp, 692 sb_int8_array coding_method, 693 int nb_channels, 694 int c, int superblocktype_2_3, 695 int cm_table_select) 696{ 697 int ch, sb, j; 698 int tmp, acc, esp_40, comp; 699 int add1, add2, add3, add4; 700 int64_t multres; 701 702 if (!superblocktype_2_3) { 703 /* This case is untested, no samples available */ 704 avpriv_request_sample(NULL, "!superblocktype_2_3"); 705 return; 706 for (ch = 0; ch < nb_channels; ch++) 707 for (sb = 0; sb < 30; sb++) { 708 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer 709 add1 = tone_level_idx[ch][sb][j] - 10; 710 if (add1 < 0) 711 add1 = 0; 712 add2 = add3 = add4 = 0; 713 if (sb > 1) { 714 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6; 715 if (add2 < 0) 716 add2 = 0; 717 } 718 if (sb > 0) { 719 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6; 720 if (add3 < 0) 721 add3 = 0; 722 } 723 if (sb < 29) { 724 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6; 725 if (add4 < 0) 726 add4 = 0; 727 } 728 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1; 729 if (tmp < 0) 730 tmp = 0; 731 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff; 732 } 733 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1]; 734 } 735 acc = 0; 736 for (ch = 0; ch < nb_channels; ch++) 737 for (sb = 0; sb < 30; sb++) 738 for (j = 0; j < 64; j++) 739 acc += tone_level_idx_temp[ch][sb][j]; 740 741 multres = 0x66666667LL * (acc * 10); 742 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31); 743 for (ch = 0; ch < nb_channels; ch++) 744 for (sb = 0; sb < 30; sb++) 745 for (j = 0; j < 64; j++) { 746 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10; 747 if (comp < 0) 748 comp += 0xff; 749 comp /= 256; // signed shift 750 switch(sb) { 751 case 0: 752 if (comp < 30) 753 comp = 30; 754 comp += 15; 755 break; 756 case 1: 757 if (comp < 24) 758 comp = 24; 759 comp += 10; 760 break; 761 case 2: 762 case 3: 763 case 4: 764 if (comp < 16) 765 comp = 16; 766 } 767 if (comp <= 5) 768 tmp = 0; 769 else if (comp <= 10) 770 tmp = 10; 771 else if (comp <= 16) 772 tmp = 16; 773 else if (comp <= 24) 774 tmp = -1; 775 else 776 tmp = 0; 777 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff; 778 } 779 for (sb = 0; sb < 30; sb++) 780 fix_coding_method_array(sb, nb_channels, coding_method); 781 for (ch = 0; ch < nb_channels; ch++) 782 for (sb = 0; sb < 30; sb++) 783 for (j = 0; j < 64; j++) 784 if (sb >= 10) { 785 if (coding_method[ch][sb][j] < 10) 786 coding_method[ch][sb][j] = 10; 787 } else { 788 if (sb >= 2) { 789 if (coding_method[ch][sb][j] < 16) 790 coding_method[ch][sb][j] = 16; 791 } else { 792 if (coding_method[ch][sb][j] < 30) 793 coding_method[ch][sb][j] = 30; 794 } 795 } 796 } else { // superblocktype_2_3 != 0 797 for (ch = 0; ch < nb_channels; ch++) 798 for (sb = 0; sb < 30; sb++) 799 for (j = 0; j < 64; j++) 800 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb]; 801 } 802} 803 804/** 805 * 806 * Called by process_subpacket_11 to process more data from subpacket 11 807 * with sb 0-8. 808 * Called by process_subpacket_12 to process data from subpacket 12 with 809 * sb 8-sb_used. 810 * 811 * @param q context 812 * @param gb bitreader context 813 * @param length packet length in bits 814 * @param sb_min lower subband processed (sb_min included) 815 * @param sb_max higher subband processed (sb_max excluded) 816 */ 817static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, 818 int length, int sb_min, int sb_max) 819{ 820 int sb, j, k, n, ch, run, channels; 821 int joined_stereo, zero_encoding; 822 int type34_first; 823 float type34_div = 0; 824 float type34_predictor; 825 float samples[10]; 826 int sign_bits[16] = {0}; 827 828 if (length == 0) { 829 // If no data use noise 830 for (sb=sb_min; sb < sb_max; sb++) 831 build_sb_samples_from_noise(q, sb); 832 833 return 0; 834 } 835 836 for (sb = sb_min; sb < sb_max; sb++) { 837 channels = q->nb_channels; 838 839 if (q->nb_channels <= 1 || sb < 12) 840 joined_stereo = 0; 841 else if (sb >= 24) 842 joined_stereo = 1; 843 else 844 joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0; 845 846 if (joined_stereo) { 847 if (get_bits_left(gb) >= 16) 848 for (j = 0; j < 16; j++) 849 sign_bits[j] = get_bits1(gb); 850 851 for (j = 0; j < 64; j++) 852 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j]) 853 q->coding_method[0][sb][j] = q->coding_method[1][sb][j]; 854 855 if (fix_coding_method_array(sb, q->nb_channels, 856 q->coding_method)) { 857 av_log(NULL, AV_LOG_ERROR, "coding method invalid\n"); 858 build_sb_samples_from_noise(q, sb); 859 continue; 860 } 861 channels = 1; 862 } 863 864 for (ch = 0; ch < channels; ch++) { 865 FIX_NOISE_IDX(q->noise_idx); 866 zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0; 867 type34_predictor = 0.0; 868 type34_first = 1; 869 870 for (j = 0; j < 128; ) { 871 switch (q->coding_method[ch][sb][j / 2]) { 872 case 8: 873 if (get_bits_left(gb) >= 10) { 874 if (zero_encoding) { 875 for (k = 0; k < 5; k++) { 876 if ((j + 2 * k) >= 128) 877 break; 878 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0; 879 } 880 } else { 881 n = get_bits(gb, 8); 882 if (n >= 243) { 883 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n"); 884 return AVERROR_INVALIDDATA; 885 } 886 887 for (k = 0; k < 5; k++) 888 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; 889 } 890 for (k = 0; k < 5; k++) 891 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx); 892 } else { 893 for (k = 0; k < 10; k++) 894 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 895 } 896 run = 10; 897 break; 898 899 case 10: 900 if (get_bits_left(gb) >= 1) { 901 float f = 0.81; 902 903 if (get_bits1(gb)) 904 f = -f; 905 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0; 906 samples[0] = f; 907 } else { 908 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 909 } 910 run = 1; 911 break; 912 913 case 16: 914 if (get_bits_left(gb) >= 10) { 915 if (zero_encoding) { 916 for (k = 0; k < 5; k++) { 917 if ((j + k) >= 128) 918 break; 919 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)]; 920 } 921 } else { 922 n = get_bits (gb, 8); 923 if (n >= 243) { 924 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n"); 925 return AVERROR_INVALIDDATA; 926 } 927 928 for (k = 0; k < 5; k++) 929 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; 930 } 931 } else { 932 for (k = 0; k < 5; k++) 933 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 934 } 935 run = 5; 936 break; 937 938 case 24: 939 if (get_bits_left(gb) >= 7) { 940 n = get_bits(gb, 7); 941 if (n >= 125) { 942 av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n"); 943 return AVERROR_INVALIDDATA; 944 } 945 946 for (k = 0; k < 3; k++) 947 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5; 948 } else { 949 for (k = 0; k < 3; k++) 950 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 951 } 952 run = 3; 953 break; 954 955 case 30: 956 if (get_bits_left(gb) >= 4) { 957 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1); 958 if (index >= FF_ARRAY_ELEMS(type30_dequant)) { 959 av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index); 960 return AVERROR_INVALIDDATA; 961 } 962 samples[0] = type30_dequant[index]; 963 } else 964 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 965 966 run = 1; 967 break; 968 969 case 34: 970 if (get_bits_left(gb) >= 7) { 971 if (type34_first) { 972 type34_div = (float)(1 << get_bits(gb, 2)); 973 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0; 974 type34_predictor = samples[0]; 975 type34_first = 0; 976 } else { 977 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1); 978 if (index >= FF_ARRAY_ELEMS(type34_delta)) { 979 av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index); 980 return AVERROR_INVALIDDATA; 981 } 982 samples[0] = type34_delta[index] / type34_div + type34_predictor; 983 type34_predictor = samples[0]; 984 } 985 } else { 986 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 987 } 988 run = 1; 989 break; 990 991 default: 992 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 993 run = 1; 994 break; 995 } 996 997 if (joined_stereo) { 998 for (k = 0; k < run && j + k < 128; k++) { 999 q->sb_samples[0][j + k][sb] = 1000 q->tone_level[0][sb][(j + k) / 2] * samples[k]; 1001 if (q->nb_channels == 2) { 1002 if (sign_bits[(j + k) / 8]) 1003 q->sb_samples[1][j + k][sb] = 1004 q->tone_level[1][sb][(j + k) / 2] * -samples[k]; 1005 else 1006 q->sb_samples[1][j + k][sb] = 1007 q->tone_level[1][sb][(j + k) / 2] * samples[k]; 1008 } 1009 } 1010 } else { 1011 for (k = 0; k < run; k++) 1012 if ((j + k) < 128) 1013 q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k]; 1014 } 1015 1016 j += run; 1017 } // j loop 1018 } // channel loop 1019 } // subband loop 1020 return 0; 1021} 1022 1023/** 1024 * Init the first element of a channel in quantized_coeffs with data 1025 * from packet 10 (quantized_coeffs[ch][0]). 1026 * This is similar to process_subpacket_9, but for a single channel 1027 * and for element [0] 1028 * same VLC tables as process_subpacket_9 are used. 1029 * 1030 * @param quantized_coeffs pointer to quantized_coeffs[ch][0] 1031 * @param gb bitreader context 1032 */ 1033static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs, 1034 GetBitContext *gb) 1035{ 1036 int i, k, run, level, diff; 1037 1038 if (get_bits_left(gb) < 16) 1039 return -1; 1040 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); 1041 1042 quantized_coeffs[0] = level; 1043 1044 for (i = 0; i < 7; ) { 1045 if (get_bits_left(gb) < 16) 1046 return -1; 1047 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; 1048 1049 if (i + run >= 8) 1050 return -1; 1051 1052 if (get_bits_left(gb) < 16) 1053 return -1; 1054 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); 1055 1056 for (k = 1; k <= run; k++) 1057 quantized_coeffs[i + k] = (level + ((k * diff) / run)); 1058 1059 level += diff; 1060 i += run; 1061 } 1062 return 0; 1063} 1064 1065/** 1066 * Related to synthesis filter, process data from packet 10 1067 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0 1068 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with 1069 * data from packet 10 1070 * 1071 * @param q context 1072 * @param gb bitreader context 1073 */ 1074static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb) 1075{ 1076 int sb, j, k, n, ch; 1077 1078 for (ch = 0; ch < q->nb_channels; ch++) { 1079 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb); 1080 1081 if (get_bits_left(gb) < 16) { 1082 memset(q->quantized_coeffs[ch][0], 0, 8); 1083 break; 1084 } 1085 } 1086 1087 n = q->sub_sampling + 1; 1088 1089 for (sb = 0; sb < n; sb++) 1090 for (ch = 0; ch < q->nb_channels; ch++) 1091 for (j = 0; j < 8; j++) { 1092 if (get_bits_left(gb) < 1) 1093 break; 1094 if (get_bits1(gb)) { 1095 for (k=0; k < 8; k++) { 1096 if (get_bits_left(gb) < 16) 1097 break; 1098 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2); 1099 } 1100 } else { 1101 for (k=0; k < 8; k++) 1102 q->tone_level_idx_hi1[ch][sb][j][k] = 0; 1103 } 1104 } 1105 1106 n = QDM2_SB_USED(q->sub_sampling) - 4; 1107 1108 for (sb = 0; sb < n; sb++) 1109 for (ch = 0; ch < q->nb_channels; ch++) { 1110 if (get_bits_left(gb) < 16) 1111 break; 1112 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2); 1113 if (sb > 19) 1114 q->tone_level_idx_hi2[ch][sb] -= 16; 1115 else 1116 for (j = 0; j < 8; j++) 1117 q->tone_level_idx_mid[ch][sb][j] = -16; 1118 } 1119 1120 n = QDM2_SB_USED(q->sub_sampling) - 5; 1121 1122 for (sb = 0; sb < n; sb++) 1123 for (ch = 0; ch < q->nb_channels; ch++) 1124 for (j = 0; j < 8; j++) { 1125 if (get_bits_left(gb) < 16) 1126 break; 1127 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32; 1128 } 1129} 1130 1131/** 1132 * Process subpacket 9, init quantized_coeffs with data from it 1133 * 1134 * @param q context 1135 * @param node pointer to node with packet 1136 */ 1137static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node) 1138{ 1139 GetBitContext gb; 1140 int i, j, k, n, ch, run, level, diff; 1141 1142 init_get_bits(&gb, node->packet->data, node->packet->size * 8); 1143 1144 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; 1145 1146 for (i = 1; i < n; i++) 1147 for (ch = 0; ch < q->nb_channels; ch++) { 1148 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2); 1149 q->quantized_coeffs[ch][i][0] = level; 1150 1151 for (j = 0; j < (8 - 1); ) { 1152 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1; 1153 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2); 1154 1155 if (j + run >= 8) 1156 return -1; 1157 1158 for (k = 1; k <= run; k++) 1159 q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run)); 1160 1161 level += diff; 1162 j += run; 1163 } 1164 } 1165 1166 for (ch = 0; ch < q->nb_channels; ch++) 1167 for (i = 0; i < 8; i++) 1168 q->quantized_coeffs[ch][0][i] = 0; 1169 1170 return 0; 1171} 1172 1173/** 1174 * Process subpacket 10 if not null, else 1175 * 1176 * @param q context 1177 * @param node pointer to node with packet 1178 */ 1179static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node) 1180{ 1181 GetBitContext gb; 1182 1183 if (node) { 1184 init_get_bits(&gb, node->packet->data, node->packet->size * 8); 1185 init_tone_level_dequantization(q, &gb); 1186 fill_tone_level_array(q, 1); 1187 } else { 1188 fill_tone_level_array(q, 0); 1189 } 1190} 1191 1192/** 1193 * Process subpacket 11 1194 * 1195 * @param q context 1196 * @param node pointer to node with packet 1197 */ 1198static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node) 1199{ 1200 GetBitContext gb; 1201 int length = 0; 1202 1203 if (node) { 1204 length = node->packet->size * 8; 1205 init_get_bits(&gb, node->packet->data, length); 1206 } 1207 1208 if (length >= 32) { 1209 int c = get_bits(&gb, 13); 1210 1211 if (c > 3) 1212 fill_coding_method_array(q->tone_level_idx, 1213 q->tone_level_idx_temp, q->coding_method, 1214 q->nb_channels, 8 * c, 1215 q->superblocktype_2_3, q->cm_table_select); 1216 } 1217 1218 synthfilt_build_sb_samples(q, &gb, length, 0, 8); 1219} 1220 1221/** 1222 * Process subpacket 12 1223 * 1224 * @param q context 1225 * @param node pointer to node with packet 1226 */ 1227static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node) 1228{ 1229 GetBitContext gb; 1230 int length = 0; 1231 1232 if (node) { 1233 length = node->packet->size * 8; 1234 init_get_bits(&gb, node->packet->data, length); 1235 } 1236 1237 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling)); 1238} 1239 1240/** 1241 * Process new subpackets for synthesis filter 1242 * 1243 * @param q context 1244 * @param list list with synthesis filter packets (list D) 1245 */ 1246static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list) 1247{ 1248 QDM2SubPNode *nodes[4]; 1249 1250 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9); 1251 if (nodes[0] != NULL) 1252 process_subpacket_9(q, nodes[0]); 1253 1254 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10); 1255 if (nodes[1] != NULL) 1256 process_subpacket_10(q, nodes[1]); 1257 else 1258 process_subpacket_10(q, NULL); 1259 1260 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11); 1261 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL) 1262 process_subpacket_11(q, nodes[2]); 1263 else 1264 process_subpacket_11(q, NULL); 1265 1266 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12); 1267 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL) 1268 process_subpacket_12(q, nodes[3]); 1269 else 1270 process_subpacket_12(q, NULL); 1271} 1272 1273/** 1274 * Decode superblock, fill packet lists. 1275 * 1276 * @param q context 1277 */ 1278static void qdm2_decode_super_block(QDM2Context *q) 1279{ 1280 GetBitContext gb; 1281 QDM2SubPacket header, *packet; 1282 int i, packet_bytes, sub_packet_size, sub_packets_D; 1283 unsigned int next_index = 0; 1284 1285 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1)); 1286 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid)); 1287 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2)); 1288 1289 q->sub_packets_B = 0; 1290 sub_packets_D = 0; 1291 1292 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8] 1293 1294 init_get_bits(&gb, q->compressed_data, q->compressed_size * 8); 1295 qdm2_decode_sub_packet_header(&gb, &header); 1296 1297 if (header.type < 2 || header.type >= 8) { 1298 q->has_errors = 1; 1299 av_log(NULL, AV_LOG_ERROR, "bad superblock type\n"); 1300 return; 1301 } 1302 1303 q->superblocktype_2_3 = (header.type == 2 || header.type == 3); 1304 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8); 1305 1306 init_get_bits(&gb, header.data, header.size * 8); 1307 1308 if (header.type == 2 || header.type == 4 || header.type == 5) { 1309 int csum = 257 * get_bits(&gb, 8); 1310 csum += 2 * get_bits(&gb, 8); 1311 1312 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); 1313 1314 if (csum != 0) { 1315 q->has_errors = 1; 1316 av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n"); 1317 return; 1318 } 1319 } 1320 1321 q->sub_packet_list_B[0].packet = NULL; 1322 q->sub_packet_list_D[0].packet = NULL; 1323 1324 for (i = 0; i < 6; i++) 1325 if (--q->fft_level_exp[i] < 0) 1326 q->fft_level_exp[i] = 0; 1327 1328 for (i = 0; packet_bytes > 0; i++) { 1329 int j; 1330 1331 if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) { 1332 SAMPLES_NEEDED_2("too many packet bytes"); 1333 return; 1334 } 1335 1336 q->sub_packet_list_A[i].next = NULL; 1337 1338 if (i > 0) { 1339 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i]; 1340 1341 /* seek to next block */ 1342 init_get_bits(&gb, header.data, header.size * 8); 1343 skip_bits(&gb, next_index * 8); 1344 1345 if (next_index >= header.size) 1346 break; 1347 } 1348 1349 /* decode subpacket */ 1350 packet = &q->sub_packets[i]; 1351 qdm2_decode_sub_packet_header(&gb, packet); 1352 next_index = packet->size + get_bits_count(&gb) / 8; 1353 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2; 1354 1355 if (packet->type == 0) 1356 break; 1357 1358 if (sub_packet_size > packet_bytes) { 1359 if (packet->type != 10 && packet->type != 11 && packet->type != 12) 1360 break; 1361 packet->size += packet_bytes - sub_packet_size; 1362 } 1363 1364 packet_bytes -= sub_packet_size; 1365 1366 /* add subpacket to 'all subpackets' list */ 1367 q->sub_packet_list_A[i].packet = packet; 1368 1369 /* add subpacket to related list */ 1370 if (packet->type == 8) { 1371 SAMPLES_NEEDED_2("packet type 8"); 1372 return; 1373 } else if (packet->type >= 9 && packet->type <= 12) { 1374 /* packets for MPEG Audio like Synthesis Filter */ 1375 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet); 1376 } else if (packet->type == 13) { 1377 for (j = 0; j < 6; j++) 1378 q->fft_level_exp[j] = get_bits(&gb, 6); 1379 } else if (packet->type == 14) { 1380 for (j = 0; j < 6; j++) 1381 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2); 1382 } else if (packet->type == 15) { 1383 SAMPLES_NEEDED_2("packet type 15") 1384 return; 1385 } else if (packet->type >= 16 && packet->type < 48 && 1386 !fft_subpackets[packet->type - 16]) { 1387 /* packets for FFT */ 1388 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet); 1389 } 1390 } // Packet bytes loop 1391 1392 if (q->sub_packet_list_D[0].packet != NULL) { 1393 process_synthesis_subpackets(q, q->sub_packet_list_D); 1394 q->do_synth_filter = 1; 1395 } else if (q->do_synth_filter) { 1396 process_subpacket_10(q, NULL); 1397 process_subpacket_11(q, NULL); 1398 process_subpacket_12(q, NULL); 1399 } 1400} 1401 1402static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet, 1403 int offset, int duration, int channel, 1404 int exp, int phase) 1405{ 1406 if (q->fft_coefs_min_index[duration] < 0) 1407 q->fft_coefs_min_index[duration] = q->fft_coefs_index; 1408 1409 q->fft_coefs[q->fft_coefs_index].sub_packet = 1410 ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); 1411 q->fft_coefs[q->fft_coefs_index].channel = channel; 1412 q->fft_coefs[q->fft_coefs_index].offset = offset; 1413 q->fft_coefs[q->fft_coefs_index].exp = exp; 1414 q->fft_coefs[q->fft_coefs_index].phase = phase; 1415 q->fft_coefs_index++; 1416} 1417 1418static void qdm2_fft_decode_tones(QDM2Context *q, int duration, 1419 GetBitContext *gb, int b) 1420{ 1421 int channel, stereo, phase, exp; 1422 int local_int_4, local_int_8, stereo_phase, local_int_10; 1423 int local_int_14, stereo_exp, local_int_20, local_int_28; 1424 int n, offset; 1425 1426 local_int_4 = 0; 1427 local_int_28 = 0; 1428 local_int_20 = 2; 1429 local_int_8 = (4 - duration); 1430 local_int_10 = 1 << (q->group_order - duration - 1); 1431 offset = 1; 1432 1433 while (get_bits_left(gb)>0) { 1434 if (q->superblocktype_2_3) { 1435 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { 1436 if (get_bits_left(gb)<0) { 1437 if(local_int_4 < q->group_size) 1438 av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n"); 1439 return; 1440 } 1441 offset = 1; 1442 if (n == 0) { 1443 local_int_4 += local_int_10; 1444 local_int_28 += (1 << local_int_8); 1445 } else { 1446 local_int_4 += 8 * local_int_10; 1447 local_int_28 += (8 << local_int_8); 1448 } 1449 } 1450 offset += (n - 2); 1451 } else { 1452 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2); 1453 while (offset >= (local_int_10 - 1)) { 1454 offset += (1 - (local_int_10 - 1)); 1455 local_int_4 += local_int_10; 1456 local_int_28 += (1 << local_int_8); 1457 } 1458 } 1459 1460 if (local_int_4 >= q->group_size) 1461 return; 1462 1463 local_int_14 = (offset >> local_int_8); 1464 if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table)) 1465 return; 1466 1467 if (q->nb_channels > 1) { 1468 channel = get_bits1(gb); 1469 stereo = get_bits1(gb); 1470 } else { 1471 channel = 0; 1472 stereo = 0; 1473 } 1474 1475 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); 1476 exp += q->fft_level_exp[fft_level_index_table[local_int_14]]; 1477 exp = (exp < 0) ? 0 : exp; 1478 1479 phase = get_bits(gb, 3); 1480 stereo_exp = 0; 1481 stereo_phase = 0; 1482 1483 if (stereo) { 1484 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1)); 1485 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1)); 1486 if (stereo_phase < 0) 1487 stereo_phase += 8; 1488 } 1489 1490 if (q->frequency_range > (local_int_14 + 1)) { 1491 int sub_packet = (local_int_20 + local_int_28); 1492 1493 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, 1494 channel, exp, phase); 1495 if (stereo) 1496 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, 1497 1 - channel, 1498 stereo_exp, stereo_phase); 1499 } 1500 offset++; 1501 } 1502} 1503 1504static void qdm2_decode_fft_packets(QDM2Context *q) 1505{ 1506 int i, j, min, max, value, type, unknown_flag; 1507 GetBitContext gb; 1508 1509 if (q->sub_packet_list_B[0].packet == NULL) 1510 return; 1511 1512 /* reset minimum indexes for FFT coefficients */ 1513 q->fft_coefs_index = 0; 1514 for (i = 0; i < 5; i++) 1515 q->fft_coefs_min_index[i] = -1; 1516 1517 /* process subpackets ordered by type, largest type first */ 1518 for (i = 0, max = 256; i < q->sub_packets_B; i++) { 1519 QDM2SubPacket *packet = NULL; 1520 1521 /* find subpacket with largest type less than max */ 1522 for (j = 0, min = 0; j < q->sub_packets_B; j++) { 1523 value = q->sub_packet_list_B[j].packet->type; 1524 if (value > min && value < max) { 1525 min = value; 1526 packet = q->sub_packet_list_B[j].packet; 1527 } 1528 } 1529 1530 max = min; 1531 1532 /* check for errors (?) */ 1533 if (!packet) 1534 return; 1535 1536 if (i == 0 && 1537 (packet->type < 16 || packet->type >= 48 || 1538 fft_subpackets[packet->type - 16])) 1539 return; 1540 1541 /* decode FFT tones */ 1542 init_get_bits(&gb, packet->data, packet->size * 8); 1543 1544 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) 1545 unknown_flag = 1; 1546 else 1547 unknown_flag = 0; 1548 1549 type = packet->type; 1550 1551 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) { 1552 int duration = q->sub_sampling + 5 - (type & 15); 1553 1554 if (duration >= 0 && duration < 4) 1555 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); 1556 } else if (type == 31) { 1557 for (j = 0; j < 4; j++) 1558 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); 1559 } else if (type == 46) { 1560 for (j = 0; j < 6; j++) 1561 q->fft_level_exp[j] = get_bits(&gb, 6); 1562 for (j = 0; j < 4; j++) 1563 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); 1564 } 1565 } // Loop on B packets 1566 1567 /* calculate maximum indexes for FFT coefficients */ 1568 for (i = 0, j = -1; i < 5; i++) 1569 if (q->fft_coefs_min_index[i] >= 0) { 1570 if (j >= 0) 1571 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i]; 1572 j = i; 1573 } 1574 if (j >= 0) 1575 q->fft_coefs_max_index[j] = q->fft_coefs_index; 1576} 1577 1578static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone) 1579{ 1580 float level, f[6]; 1581 int i; 1582 QDM2Complex c; 1583 const double iscale = 2.0 * M_PI / 512.0; 1584 1585 tone->phase += tone->phase_shift; 1586 1587 /* calculate current level (maximum amplitude) of tone */ 1588 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level; 1589 c.im = level * sin(tone->phase * iscale); 1590 c.re = level * cos(tone->phase * iscale); 1591 1592 /* generate FFT coefficients for tone */ 1593 if (tone->duration >= 3 || tone->cutoff >= 3) { 1594 tone->complex[0].im += c.im; 1595 tone->complex[0].re += c.re; 1596 tone->complex[1].im -= c.im; 1597 tone->complex[1].re -= c.re; 1598 } else { 1599 f[1] = -tone->table[4]; 1600 f[0] = tone->table[3] - tone->table[0]; 1601 f[2] = 1.0 - tone->table[2] - tone->table[3]; 1602 f[3] = tone->table[1] + tone->table[4] - 1.0; 1603 f[4] = tone->table[0] - tone->table[1]; 1604 f[5] = tone->table[2]; 1605 for (i = 0; i < 2; i++) { 1606 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += 1607 c.re * f[i]; 1608 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += 1609 c.im * ((tone->cutoff <= i) ? -f[i] : f[i]); 1610 } 1611 for (i = 0; i < 4; i++) { 1612 tone->complex[i].re += c.re * f[i + 2]; 1613 tone->complex[i].im += c.im * f[i + 2]; 1614 } 1615 } 1616 1617 /* copy the tone if it has not yet died out */ 1618 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) { 1619 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone)); 1620 q->fft_tone_end = (q->fft_tone_end + 1) % 1000; 1621 } 1622} 1623 1624static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet) 1625{ 1626 int i, j, ch; 1627 const double iscale = 0.25 * M_PI; 1628 1629 for (ch = 0; ch < q->channels; ch++) { 1630 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex)); 1631 } 1632 1633 1634 /* apply FFT tones with duration 4 (1 FFT period) */ 1635 if (q->fft_coefs_min_index[4] >= 0) 1636 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) { 1637 float level; 1638 QDM2Complex c; 1639 1640 if (q->fft_coefs[i].sub_packet != sub_packet) 1641 break; 1642 1643 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel; 1644 level = (q->fft_coefs[i].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[i].exp & 63]; 1645 1646 c.re = level * cos(q->fft_coefs[i].phase * iscale); 1647 c.im = level * sin(q->fft_coefs[i].phase * iscale); 1648 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re; 1649 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im; 1650 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re; 1651 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im; 1652 } 1653 1654 /* generate existing FFT tones */ 1655 for (i = q->fft_tone_end; i != q->fft_tone_start; ) { 1656 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]); 1657 q->fft_tone_start = (q->fft_tone_start + 1) % 1000; 1658 } 1659 1660 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */ 1661 for (i = 0; i < 4; i++) 1662 if (q->fft_coefs_min_index[i] >= 0) { 1663 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) { 1664 int offset, four_i; 1665 FFTTone tone; 1666 1667 if (q->fft_coefs[j].sub_packet != sub_packet) 1668 break; 1669 1670 four_i = (4 - i); 1671 offset = q->fft_coefs[j].offset >> four_i; 1672 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel; 1673 1674 if (offset < q->frequency_range) { 1675 if (offset < 2) 1676 tone.cutoff = offset; 1677 else 1678 tone.cutoff = (offset >= 60) ? 3 : 2; 1679 1680 tone.level = (q->fft_coefs[j].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[j].exp & 63]; 1681 tone.complex = &q->fft.complex[ch][offset]; 1682 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)]; 1683 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128; 1684 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i); 1685 tone.duration = i; 1686 tone.time_index = 0; 1687 1688 qdm2_fft_generate_tone(q, &tone); 1689 } 1690 } 1691 q->fft_coefs_min_index[i] = j; 1692 } 1693} 1694 1695static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet) 1696{ 1697 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f; 1698 float *out = q->output_buffer + channel; 1699 int i; 1700 q->fft.complex[channel][0].re *= 2.0f; 1701 q->fft.complex[channel][0].im = 0.0f; 1702 q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]); 1703 /* add samples to output buffer */ 1704 for (i = 0; i < FFALIGN(q->fft_size, 8); i++) { 1705 out[0] += q->fft.complex[channel][i].re * gain; 1706 out[q->channels] += q->fft.complex[channel][i].im * gain; 1707 out += 2 * q->channels; 1708 } 1709} 1710 1711/** 1712 * @param q context 1713 * @param index subpacket number 1714 */ 1715static void qdm2_synthesis_filter(QDM2Context *q, int index) 1716{ 1717 int i, k, ch, sb_used, sub_sampling, dither_state = 0; 1718 1719 /* copy sb_samples */ 1720 sb_used = QDM2_SB_USED(q->sub_sampling); 1721 1722 for (ch = 0; ch < q->channels; ch++) 1723 for (i = 0; i < 8; i++) 1724 for (k = sb_used; k < SBLIMIT; k++) 1725 q->sb_samples[ch][(8 * index) + i][k] = 0; 1726 1727 for (ch = 0; ch < q->nb_channels; ch++) { 1728 float *samples_ptr = q->samples + ch; 1729 1730 for (i = 0; i < 8; i++) { 1731 ff_mpa_synth_filter_float(&q->mpadsp, 1732 q->synth_buf[ch], &(q->synth_buf_offset[ch]), 1733 ff_mpa_synth_window_float, &dither_state, 1734 samples_ptr, q->nb_channels, 1735 q->sb_samples[ch][(8 * index) + i]); 1736 samples_ptr += 32 * q->nb_channels; 1737 } 1738 } 1739 1740 /* add samples to output buffer */ 1741 sub_sampling = (4 >> q->sub_sampling); 1742 1743 for (ch = 0; ch < q->channels; ch++) 1744 for (i = 0; i < q->frame_size; i++) 1745 q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch]; 1746} 1747 1748/** 1749 * Init static data (does not depend on specific file) 1750 * 1751 * @param q context 1752 */ 1753static av_cold void qdm2_init_static_data(void) { 1754 static int done; 1755 1756 if(done) 1757 return; 1758 1759 qdm2_init_vlc(); 1760 ff_mpa_synth_init_float(ff_mpa_synth_window_float); 1761 softclip_table_init(); 1762 rnd_table_init(); 1763 init_noise_samples(); 1764 1765 done = 1; 1766} 1767 1768/** 1769 * Init parameters from codec extradata 1770 */ 1771static av_cold int qdm2_decode_init(AVCodecContext *avctx) 1772{ 1773 QDM2Context *s = avctx->priv_data; 1774 uint8_t *extradata; 1775 int extradata_size; 1776 int tmp_val, tmp, size; 1777 1778 qdm2_init_static_data(); 1779 1780 /* extradata parsing 1781 1782 Structure: 1783 wave { 1784 frma (QDM2) 1785 QDCA 1786 QDCP 1787 } 1788 1789 32 size (including this field) 1790 32 tag (=frma) 1791 32 type (=QDM2 or QDMC) 1792 1793 32 size (including this field, in bytes) 1794 32 tag (=QDCA) // maybe mandatory parameters 1795 32 unknown (=1) 1796 32 channels (=2) 1797 32 samplerate (=44100) 1798 32 bitrate (=96000) 1799 32 block size (=4096) 1800 32 frame size (=256) (for one channel) 1801 32 packet size (=1300) 1802 1803 32 size (including this field, in bytes) 1804 32 tag (=QDCP) // maybe some tuneable parameters 1805 32 float1 (=1.0) 1806 32 zero ? 1807 32 float2 (=1.0) 1808 32 float3 (=1.0) 1809 32 unknown (27) 1810 32 unknown (8) 1811 32 zero ? 1812 */ 1813 1814 if (!avctx->extradata || (avctx->extradata_size < 48)) { 1815 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n"); 1816 return -1; 1817 } 1818 1819 extradata = avctx->extradata; 1820 extradata_size = avctx->extradata_size; 1821 1822 while (extradata_size > 7) { 1823 if (!memcmp(extradata, "frmaQDM", 7)) 1824 break; 1825 extradata++; 1826 extradata_size--; 1827 } 1828 1829 if (extradata_size < 12) { 1830 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n", 1831 extradata_size); 1832 return -1; 1833 } 1834 1835 if (memcmp(extradata, "frmaQDM", 7)) { 1836 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n"); 1837 return -1; 1838 } 1839 1840 if (extradata[7] == 'C') { 1841// s->is_qdmc = 1; 1842 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n"); 1843 return -1; 1844 } 1845 1846 extradata += 8; 1847 extradata_size -= 8; 1848 1849 size = AV_RB32(extradata); 1850 1851 if(size > extradata_size){ 1852 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n", 1853 extradata_size, size); 1854 return -1; 1855 } 1856 1857 extradata += 4; 1858 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size); 1859 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) { 1860 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n"); 1861 return -1; 1862 } 1863 1864 extradata += 8; 1865 1866 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata); 1867 extradata += 4; 1868 if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) { 1869 av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n"); 1870 return AVERROR_INVALIDDATA; 1871 } 1872 avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO : 1873 AV_CH_LAYOUT_MONO; 1874 1875 avctx->sample_rate = AV_RB32(extradata); 1876 extradata += 4; 1877 1878 avctx->bit_rate = AV_RB32(extradata); 1879 extradata += 4; 1880 1881 s->group_size = AV_RB32(extradata); 1882 extradata += 4; 1883 1884 s->fft_size = AV_RB32(extradata); 1885 extradata += 4; 1886 1887 s->checksum_size = AV_RB32(extradata); 1888 if (s->checksum_size >= 1U << 28) { 1889 av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size); 1890 return AVERROR_INVALIDDATA; 1891 } 1892 1893 s->fft_order = av_log2(s->fft_size) + 1; 1894 1895 // something like max decodable tones 1896 s->group_order = av_log2(s->group_size) + 1; 1897 s->frame_size = s->group_size / 16; // 16 iterations per super block 1898 1899 if (s->frame_size > QDM2_MAX_FRAME_SIZE) 1900 return AVERROR_INVALIDDATA; 1901 1902 s->sub_sampling = s->fft_order - 7; 1903 s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); 1904 1905 switch ((s->sub_sampling * 2 + s->channels - 1)) { 1906 case 0: tmp = 40; break; 1907 case 1: tmp = 48; break; 1908 case 2: tmp = 56; break; 1909 case 3: tmp = 72; break; 1910 case 4: tmp = 80; break; 1911 case 5: tmp = 100;break; 1912 default: tmp=s->sub_sampling; break; 1913 } 1914 tmp_val = 0; 1915 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1; 1916 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2; 1917 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3; 1918 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4; 1919 s->cm_table_select = tmp_val; 1920 1921 if (avctx->bit_rate <= 8000) 1922 s->coeff_per_sb_select = 0; 1923 else if (avctx->bit_rate < 16000) 1924 s->coeff_per_sb_select = 1; 1925 else 1926 s->coeff_per_sb_select = 2; 1927 1928 // Fail on unknown fft order 1929 if ((s->fft_order < 7) || (s->fft_order > 9)) { 1930 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order); 1931 return -1; 1932 } 1933 if (s->fft_size != (1 << (s->fft_order - 1))) { 1934 av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size); 1935 return AVERROR_INVALIDDATA; 1936 } 1937 1938 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R); 1939 ff_mpadsp_init(&s->mpadsp); 1940 1941 avctx->sample_fmt = AV_SAMPLE_FMT_S16; 1942 1943 return 0; 1944} 1945 1946static av_cold int qdm2_decode_close(AVCodecContext *avctx) 1947{ 1948 QDM2Context *s = avctx->priv_data; 1949 1950 ff_rdft_end(&s->rdft_ctx); 1951 1952 return 0; 1953} 1954 1955static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out) 1956{ 1957 int ch, i; 1958 const int frame_size = (q->frame_size * q->channels); 1959 1960 if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2) 1961 return -1; 1962 1963 /* select input buffer */ 1964 q->compressed_data = in; 1965 q->compressed_size = q->checksum_size; 1966 1967 /* copy old block, clear new block of output samples */ 1968 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float)); 1969 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float)); 1970 1971 /* decode block of QDM2 compressed data */ 1972 if (q->sub_packet == 0) { 1973 q->has_errors = 0; // zero it for a new super block 1974 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n"); 1975 qdm2_decode_super_block(q); 1976 } 1977 1978 /* parse subpackets */ 1979 if (!q->has_errors) { 1980 if (q->sub_packet == 2) 1981 qdm2_decode_fft_packets(q); 1982 1983 qdm2_fft_tone_synthesizer(q, q->sub_packet); 1984 } 1985 1986 /* sound synthesis stage 1 (FFT) */ 1987 for (ch = 0; ch < q->channels; ch++) { 1988 qdm2_calculate_fft(q, ch, q->sub_packet); 1989 1990 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) { 1991 SAMPLES_NEEDED_2("has errors, and C list is not empty") 1992 return -1; 1993 } 1994 } 1995 1996 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */ 1997 if (!q->has_errors && q->do_synth_filter) 1998 qdm2_synthesis_filter(q, q->sub_packet); 1999 2000 q->sub_packet = (q->sub_packet + 1) % 16; 2001 2002 /* clip and convert output float[] to 16bit signed samples */ 2003 for (i = 0; i < frame_size; i++) { 2004 int value = (int)q->output_buffer[i]; 2005 2006 if (value > SOFTCLIP_THRESHOLD) 2007 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD]; 2008 else if (value < -SOFTCLIP_THRESHOLD) 2009 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD]; 2010 2011 out[i] = value; 2012 } 2013 2014 return 0; 2015} 2016 2017static int qdm2_decode_frame(AVCodecContext *avctx, void *data, 2018 int *got_frame_ptr, AVPacket *avpkt) 2019{ 2020 AVFrame *frame = data; 2021 const uint8_t *buf = avpkt->data; 2022 int buf_size = avpkt->size; 2023 QDM2Context *s = avctx->priv_data; 2024 int16_t *out; 2025 int i, ret; 2026 2027 if(!buf) 2028 return 0; 2029 if(buf_size < s->checksum_size) 2030 return -1; 2031 2032 /* get output buffer */ 2033 frame->nb_samples = 16 * s->frame_size; 2034 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) 2035 return ret; 2036 out = (int16_t *)frame->data[0]; 2037 2038 for (i = 0; i < 16; i++) { 2039 if (qdm2_decode(s, buf, out) < 0) 2040 return -1; 2041 out += s->channels * s->frame_size; 2042 } 2043 2044 *got_frame_ptr = 1; 2045 2046 return s->checksum_size; 2047} 2048 2049AVCodec ff_qdm2_decoder = { 2050 .name = "qdm2", 2051 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), 2052 .type = AVMEDIA_TYPE_AUDIO, 2053 .id = AV_CODEC_ID_QDM2, 2054 .priv_data_size = sizeof(QDM2Context), 2055 .init = qdm2_decode_init, 2056 .close = qdm2_decode_close, 2057 .decode = qdm2_decode_frame, 2058 .capabilities = CODEC_CAP_DR1, 2059}; 2060