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 libavcodec/qdm2.c 27 * QDM2 decoder 28 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni 29 * The decoder is not perfect yet, there are still some distortions 30 * especially on files encoded with 16 or 8 subbands. 31 */ 32 33#include <math.h> 34#include <stddef.h> 35#include <stdio.h> 36 37#define ALT_BITSTREAM_READER_LE 38#include "avcodec.h" 39#include "bitstream.h" 40#include "dsputil.h" 41#include "mpegaudio.h" 42 43#include "qdm2data.h" 44 45#undef NDEBUG 46#include <assert.h> 47 48 49#define SOFTCLIP_THRESHOLD 27600 50#define HARDCLIP_THRESHOLD 35716 51 52 53#define QDM2_LIST_ADD(list, size, packet) \ 54do { \ 55 if (size > 0) { \ 56 list[size - 1].next = &list[size]; \ 57 } \ 58 list[size].packet = packet; \ 59 list[size].next = NULL; \ 60 size++; \ 61} while(0) 62 63// Result is 8, 16 or 30 64#define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling)) 65 66#define FIX_NOISE_IDX(noise_idx) \ 67 if ((noise_idx) >= 3840) \ 68 (noise_idx) -= 3840; \ 69 70#define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)]) 71 72#define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb))) 73 74#define SAMPLES_NEEDED \ 75 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n"); 76 77#define SAMPLES_NEEDED_2(why) \ 78 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why); 79 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_16(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 fft_frame_size; ///< size of fft frame, in components (1 comples = re + im) 143 int frame_size; ///< size of data frame 144 int frequency_range; 145 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */ 146 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2 147 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4) 148 149 /// Packets and packet lists 150 QDM2SubPacket sub_packets[16]; ///< the packets themselves 151 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets 152 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list 153 int sub_packets_B; ///< number of packets on 'B' list 154 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors? 155 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets 156 157 /// FFT and tones 158 FFTTone fft_tones[1000]; 159 int fft_tone_start; 160 int fft_tone_end; 161 FFTCoefficient fft_coefs[1000]; 162 int fft_coefs_index; 163 int fft_coefs_min_index[5]; 164 int fft_coefs_max_index[5]; 165 int fft_level_exp[6]; 166 RDFTContext rdft_ctx; 167 QDM2FFT fft; 168 169 /// I/O data 170 const uint8_t *compressed_data; 171 int compressed_size; 172 float output_buffer[1024]; 173 174 /// Synthesis filter 175 DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512*2]); 176 int synth_buf_offset[MPA_MAX_CHANNELS]; 177 DECLARE_ALIGNED_16(int32_t, sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT]); 178 179 /// Mixed temporary data used in decoding 180 float tone_level[MPA_MAX_CHANNELS][30][64]; 181 int8_t coding_method[MPA_MAX_CHANNELS][30][64]; 182 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8]; 183 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8]; 184 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8]; 185 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8]; 186 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26]; 187 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64]; 188 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]; 189 190 // Flags 191 int has_errors; ///< packet has errors 192 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type 193 int do_synth_filter; ///< used to perform or skip synthesis filter 194 195 int sub_packet; 196 int noise_idx; ///< index for dithering noise table 197} QDM2Context; 198 199 200static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE]; 201 202static VLC vlc_tab_level; 203static VLC vlc_tab_diff; 204static VLC vlc_tab_run; 205static VLC fft_level_exp_alt_vlc; 206static VLC fft_level_exp_vlc; 207static VLC fft_stereo_exp_vlc; 208static VLC fft_stereo_phase_vlc; 209static VLC vlc_tab_tone_level_idx_hi1; 210static VLC vlc_tab_tone_level_idx_mid; 211static VLC vlc_tab_tone_level_idx_hi2; 212static VLC vlc_tab_type30; 213static VLC vlc_tab_type34; 214static VLC vlc_tab_fft_tone_offset[5]; 215 216static uint16_t softclip_table[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1]; 217static float noise_table[4096]; 218static uint8_t random_dequant_index[256][5]; 219static uint8_t random_dequant_type24[128][3]; 220static float noise_samples[128]; 221 222static DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]); 223 224 225static av_cold void softclip_table_init(void) { 226 int i; 227 double dfl = SOFTCLIP_THRESHOLD - 32767; 228 float delta = 1.0 / -dfl; 229 for (i = 0; i < HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1; i++) 230 softclip_table[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF); 231} 232 233 234// random generated table 235static av_cold void rnd_table_init(void) { 236 int i,j; 237 uint32_t ldw,hdw; 238 uint64_t tmp64_1; 239 uint64_t random_seed = 0; 240 float delta = 1.0 / 16384.0; 241 for(i = 0; i < 4096 ;i++) { 242 random_seed = random_seed * 214013 + 2531011; 243 noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3; 244 } 245 246 for (i = 0; i < 256 ;i++) { 247 random_seed = 81; 248 ldw = i; 249 for (j = 0; j < 5 ;j++) { 250 random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF); 251 ldw = (uint32_t)ldw % (uint32_t)random_seed; 252 tmp64_1 = (random_seed * 0x55555556); 253 hdw = (uint32_t)(tmp64_1 >> 32); 254 random_seed = (uint64_t)(hdw + (ldw >> 31)); 255 } 256 } 257 for (i = 0; i < 128 ;i++) { 258 random_seed = 25; 259 ldw = i; 260 for (j = 0; j < 3 ;j++) { 261 random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF); 262 ldw = (uint32_t)ldw % (uint32_t)random_seed; 263 tmp64_1 = (random_seed * 0x66666667); 264 hdw = (uint32_t)(tmp64_1 >> 33); 265 random_seed = hdw + (ldw >> 31); 266 } 267 } 268} 269 270 271static av_cold void init_noise_samples(void) { 272 int i; 273 int random_seed = 0; 274 float delta = 1.0 / 16384.0; 275 for (i = 0; i < 128;i++) { 276 random_seed = random_seed * 214013 + 2531011; 277 noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0); 278 } 279} 280 281 282static av_cold void qdm2_init_vlc(void) 283{ 284 init_vlc (&vlc_tab_level, 8, 24, 285 vlc_tab_level_huffbits, 1, 1, 286 vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); 287 288 init_vlc (&vlc_tab_diff, 8, 37, 289 vlc_tab_diff_huffbits, 1, 1, 290 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); 291 292 init_vlc (&vlc_tab_run, 5, 6, 293 vlc_tab_run_huffbits, 1, 1, 294 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); 295 296 init_vlc (&fft_level_exp_alt_vlc, 8, 28, 297 fft_level_exp_alt_huffbits, 1, 1, 298 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); 299 300 init_vlc (&fft_level_exp_vlc, 8, 20, 301 fft_level_exp_huffbits, 1, 1, 302 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); 303 304 init_vlc (&fft_stereo_exp_vlc, 6, 7, 305 fft_stereo_exp_huffbits, 1, 1, 306 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); 307 308 init_vlc (&fft_stereo_phase_vlc, 6, 9, 309 fft_stereo_phase_huffbits, 1, 1, 310 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); 311 312 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20, 313 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1, 314 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); 315 316 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24, 317 vlc_tab_tone_level_idx_mid_huffbits, 1, 1, 318 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); 319 320 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24, 321 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1, 322 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); 323 324 init_vlc (&vlc_tab_type30, 6, 9, 325 vlc_tab_type30_huffbits, 1, 1, 326 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); 327 328 init_vlc (&vlc_tab_type34, 5, 10, 329 vlc_tab_type34_huffbits, 1, 1, 330 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); 331 332 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23, 333 vlc_tab_fft_tone_offset_0_huffbits, 1, 1, 334 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); 335 336 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28, 337 vlc_tab_fft_tone_offset_1_huffbits, 1, 1, 338 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); 339 340 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32, 341 vlc_tab_fft_tone_offset_2_huffbits, 1, 1, 342 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); 343 344 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35, 345 vlc_tab_fft_tone_offset_3_huffbits, 1, 1, 346 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); 347 348 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38, 349 vlc_tab_fft_tone_offset_4_huffbits, 1, 1, 350 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); 351} 352 353 354/* for floating point to fixed point conversion */ 355static const float f2i_scale = (float) (1 << (FRAC_BITS - 15)); 356 357 358static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth) 359{ 360 int value; 361 362 value = get_vlc2(gb, vlc->table, vlc->bits, depth); 363 364 /* stage-2, 3 bits exponent escape sequence */ 365 if (value-- == 0) 366 value = get_bits (gb, get_bits (gb, 3) + 1); 367 368 /* stage-3, optional */ 369 if (flag) { 370 int tmp = vlc_stage3_values[value]; 371 372 if ((value & ~3) > 0) 373 tmp += get_bits (gb, (value >> 2)); 374 value = tmp; 375 } 376 377 return value; 378} 379 380 381static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth) 382{ 383 int value = qdm2_get_vlc (gb, vlc, 0, depth); 384 385 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1); 386} 387 388 389/** 390 * QDM2 checksum 391 * 392 * @param data pointer to data to be checksum'ed 393 * @param length data length 394 * @param value checksum value 395 * 396 * @return 0 if checksum is OK 397 */ 398static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) { 399 int i; 400 401 for (i=0; i < length; i++) 402 value -= data[i]; 403 404 return (uint16_t)(value & 0xffff); 405} 406 407 408/** 409 * Fills a QDM2SubPacket structure with packet type, size, and data pointer. 410 * 411 * @param gb bitreader context 412 * @param sub_packet packet under analysis 413 */ 414static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet) 415{ 416 sub_packet->type = get_bits (gb, 8); 417 418 if (sub_packet->type == 0) { 419 sub_packet->size = 0; 420 sub_packet->data = NULL; 421 } else { 422 sub_packet->size = get_bits (gb, 8); 423 424 if (sub_packet->type & 0x80) { 425 sub_packet->size <<= 8; 426 sub_packet->size |= get_bits (gb, 8); 427 sub_packet->type &= 0x7f; 428 } 429 430 if (sub_packet->type == 0x7f) 431 sub_packet->type |= (get_bits (gb, 8) << 8); 432 433 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data 434 } 435 436 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n", 437 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8); 438} 439 440 441/** 442 * Return node pointer to first packet of requested type in list. 443 * 444 * @param list list of subpackets to be scanned 445 * @param type type of searched subpacket 446 * @return node pointer for subpacket if found, else NULL 447 */ 448static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type) 449{ 450 while (list != NULL && list->packet != NULL) { 451 if (list->packet->type == type) 452 return list; 453 list = list->next; 454 } 455 return NULL; 456} 457 458 459/** 460 * Replaces 8 elements with their average value. 461 * Called by qdm2_decode_superblock before starting subblock decoding. 462 * 463 * @param q context 464 */ 465static void average_quantized_coeffs (QDM2Context *q) 466{ 467 int i, j, n, ch, sum; 468 469 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; 470 471 for (ch = 0; ch < q->nb_channels; ch++) 472 for (i = 0; i < n; i++) { 473 sum = 0; 474 475 for (j = 0; j < 8; j++) 476 sum += q->quantized_coeffs[ch][i][j]; 477 478 sum /= 8; 479 if (sum > 0) 480 sum--; 481 482 for (j=0; j < 8; j++) 483 q->quantized_coeffs[ch][i][j] = sum; 484 } 485} 486 487 488/** 489 * Build subband samples with noise weighted by q->tone_level. 490 * Called by synthfilt_build_sb_samples. 491 * 492 * @param q context 493 * @param sb subband index 494 */ 495static void build_sb_samples_from_noise (QDM2Context *q, int sb) 496{ 497 int ch, j; 498 499 FIX_NOISE_IDX(q->noise_idx); 500 501 if (!q->nb_channels) 502 return; 503 504 for (ch = 0; ch < q->nb_channels; ch++) 505 for (j = 0; j < 64; j++) { 506 q->sb_samples[ch][j * 2][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5); 507 q->sb_samples[ch][j * 2 + 1][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5); 508 } 509} 510 511 512/** 513 * Called while processing data from subpackets 11 and 12. 514 * Used after making changes to coding_method array. 515 * 516 * @param sb subband index 517 * @param channels number of channels 518 * @param coding_method q->coding_method[0][0][0] 519 */ 520static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method) 521{ 522 int j,k; 523 int ch; 524 int run, case_val; 525 int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4}; 526 527 for (ch = 0; ch < channels; ch++) { 528 for (j = 0; j < 64; ) { 529 if((coding_method[ch][sb][j] - 8) > 22) { 530 run = 1; 531 case_val = 8; 532 } else { 533 switch (switchtable[coding_method[ch][sb][j]-8]) { 534 case 0: run = 10; case_val = 10; break; 535 case 1: run = 1; case_val = 16; break; 536 case 2: run = 5; case_val = 24; break; 537 case 3: run = 3; case_val = 30; break; 538 case 4: run = 1; case_val = 30; break; 539 case 5: run = 1; case_val = 8; break; 540 default: run = 1; case_val = 8; break; 541 } 542 } 543 for (k = 0; k < run; k++) 544 if (j + k < 128) 545 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) 546 if (k > 0) { 547 SAMPLES_NEEDED 548 //not debugged, almost never used 549 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t)); 550 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t)); 551 } 552 j += run; 553 } 554 } 555} 556 557 558/** 559 * Related to synthesis filter 560 * Called by process_subpacket_10 561 * 562 * @param q context 563 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10 564 */ 565static void fill_tone_level_array (QDM2Context *q, int flag) 566{ 567 int i, sb, ch, sb_used; 568 int tmp, tab; 569 570 // This should never happen 571 if (q->nb_channels <= 0) 572 return; 573 574 for (ch = 0; ch < q->nb_channels; ch++) 575 for (sb = 0; sb < 30; sb++) 576 for (i = 0; i < 8; i++) { 577 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1)) 578 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+ 579 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; 580 else 581 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; 582 if(tmp < 0) 583 tmp += 0xff; 584 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff; 585 } 586 587 sb_used = QDM2_SB_USED(q->sub_sampling); 588 589 if ((q->superblocktype_2_3 != 0) && !flag) { 590 for (sb = 0; sb < sb_used; sb++) 591 for (ch = 0; ch < q->nb_channels; ch++) 592 for (i = 0; i < 64; i++) { 593 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; 594 if (q->tone_level_idx[ch][sb][i] < 0) 595 q->tone_level[ch][sb][i] = 0; 596 else 597 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f]; 598 } 599 } else { 600 tab = q->superblocktype_2_3 ? 0 : 1; 601 for (sb = 0; sb < sb_used; sb++) { 602 if ((sb >= 4) && (sb <= 23)) { 603 for (ch = 0; ch < q->nb_channels; ch++) 604 for (i = 0; i < 64; i++) { 605 tmp = q->tone_level_idx_base[ch][sb][i / 8] - 606 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] - 607 q->tone_level_idx_mid[ch][sb - 4][i / 8] - 608 q->tone_level_idx_hi2[ch][sb - 4]; 609 q->tone_level_idx[ch][sb][i] = tmp & 0xff; 610 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 611 q->tone_level[ch][sb][i] = 0; 612 else 613 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 614 } 615 } else { 616 if (sb > 4) { 617 for (ch = 0; ch < q->nb_channels; ch++) 618 for (i = 0; i < 64; i++) { 619 tmp = q->tone_level_idx_base[ch][sb][i / 8] - 620 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] - 621 q->tone_level_idx_hi2[ch][sb - 4]; 622 q->tone_level_idx[ch][sb][i] = tmp & 0xff; 623 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 624 q->tone_level[ch][sb][i] = 0; 625 else 626 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 627 } 628 } else { 629 for (ch = 0; ch < q->nb_channels; ch++) 630 for (i = 0; i < 64; i++) { 631 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; 632 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 633 q->tone_level[ch][sb][i] = 0; 634 else 635 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 636 } 637 } 638 } 639 } 640 } 641 642 return; 643} 644 645 646/** 647 * Related to synthesis filter 648 * Called by process_subpacket_11 649 * c is built with data from subpacket 11 650 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples 651 * 652 * @param tone_level_idx 653 * @param tone_level_idx_temp 654 * @param coding_method q->coding_method[0][0][0] 655 * @param nb_channels number of channels 656 * @param c coming from subpacket 11, passed as 8*c 657 * @param superblocktype_2_3 flag based on superblock packet type 658 * @param cm_table_select q->cm_table_select 659 */ 660static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp, 661 sb_int8_array coding_method, int nb_channels, 662 int c, int superblocktype_2_3, int cm_table_select) 663{ 664 int ch, sb, j; 665 int tmp, acc, esp_40, comp; 666 int add1, add2, add3, add4; 667 int64_t multres; 668 669 // This should never happen 670 if (nb_channels <= 0) 671 return; 672 673 if (!superblocktype_2_3) { 674 /* This case is untested, no samples available */ 675 SAMPLES_NEEDED 676 for (ch = 0; ch < nb_channels; ch++) 677 for (sb = 0; sb < 30; sb++) { 678 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer 679 add1 = tone_level_idx[ch][sb][j] - 10; 680 if (add1 < 0) 681 add1 = 0; 682 add2 = add3 = add4 = 0; 683 if (sb > 1) { 684 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6; 685 if (add2 < 0) 686 add2 = 0; 687 } 688 if (sb > 0) { 689 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6; 690 if (add3 < 0) 691 add3 = 0; 692 } 693 if (sb < 29) { 694 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6; 695 if (add4 < 0) 696 add4 = 0; 697 } 698 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1; 699 if (tmp < 0) 700 tmp = 0; 701 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff; 702 } 703 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1]; 704 } 705 acc = 0; 706 for (ch = 0; ch < nb_channels; ch++) 707 for (sb = 0; sb < 30; sb++) 708 for (j = 0; j < 64; j++) 709 acc += tone_level_idx_temp[ch][sb][j]; 710 if (acc) 711 tmp = c * 256 / (acc & 0xffff); 712 multres = 0x66666667 * (acc * 10); 713 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31); 714 for (ch = 0; ch < nb_channels; ch++) 715 for (sb = 0; sb < 30; sb++) 716 for (j = 0; j < 64; j++) { 717 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10; 718 if (comp < 0) 719 comp += 0xff; 720 comp /= 256; // signed shift 721 switch(sb) { 722 case 0: 723 if (comp < 30) 724 comp = 30; 725 comp += 15; 726 break; 727 case 1: 728 if (comp < 24) 729 comp = 24; 730 comp += 10; 731 break; 732 case 2: 733 case 3: 734 case 4: 735 if (comp < 16) 736 comp = 16; 737 } 738 if (comp <= 5) 739 tmp = 0; 740 else if (comp <= 10) 741 tmp = 10; 742 else if (comp <= 16) 743 tmp = 16; 744 else if (comp <= 24) 745 tmp = -1; 746 else 747 tmp = 0; 748 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff; 749 } 750 for (sb = 0; sb < 30; sb++) 751 fix_coding_method_array(sb, nb_channels, coding_method); 752 for (ch = 0; ch < nb_channels; ch++) 753 for (sb = 0; sb < 30; sb++) 754 for (j = 0; j < 64; j++) 755 if (sb >= 10) { 756 if (coding_method[ch][sb][j] < 10) 757 coding_method[ch][sb][j] = 10; 758 } else { 759 if (sb >= 2) { 760 if (coding_method[ch][sb][j] < 16) 761 coding_method[ch][sb][j] = 16; 762 } else { 763 if (coding_method[ch][sb][j] < 30) 764 coding_method[ch][sb][j] = 30; 765 } 766 } 767 } else { // superblocktype_2_3 != 0 768 for (ch = 0; ch < nb_channels; ch++) 769 for (sb = 0; sb < 30; sb++) 770 for (j = 0; j < 64; j++) 771 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb]; 772 } 773 774 return; 775} 776 777 778/** 779 * 780 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8 781 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used 782 * 783 * @param q context 784 * @param gb bitreader context 785 * @param length packet length in bits 786 * @param sb_min lower subband processed (sb_min included) 787 * @param sb_max higher subband processed (sb_max excluded) 788 */ 789static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max) 790{ 791 int sb, j, k, n, ch, run, channels; 792 int joined_stereo, zero_encoding, chs; 793 int type34_first; 794 float type34_div = 0; 795 float type34_predictor; 796 float samples[10], sign_bits[16]; 797 798 if (length == 0) { 799 // If no data use noise 800 for (sb=sb_min; sb < sb_max; sb++) 801 build_sb_samples_from_noise (q, sb); 802 803 return; 804 } 805 806 for (sb = sb_min; sb < sb_max; sb++) { 807 FIX_NOISE_IDX(q->noise_idx); 808 809 channels = q->nb_channels; 810 811 if (q->nb_channels <= 1 || sb < 12) 812 joined_stereo = 0; 813 else if (sb >= 24) 814 joined_stereo = 1; 815 else 816 joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0; 817 818 if (joined_stereo) { 819 if (BITS_LEFT(length,gb) >= 16) 820 for (j = 0; j < 16; j++) 821 sign_bits[j] = get_bits1 (gb); 822 823 for (j = 0; j < 64; j++) 824 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j]) 825 q->coding_method[0][sb][j] = q->coding_method[1][sb][j]; 826 827 fix_coding_method_array(sb, q->nb_channels, q->coding_method); 828 channels = 1; 829 } 830 831 for (ch = 0; ch < channels; ch++) { 832 zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0; 833 type34_predictor = 0.0; 834 type34_first = 1; 835 836 for (j = 0; j < 128; ) { 837 switch (q->coding_method[ch][sb][j / 2]) { 838 case 8: 839 if (BITS_LEFT(length,gb) >= 10) { 840 if (zero_encoding) { 841 for (k = 0; k < 5; k++) { 842 if ((j + 2 * k) >= 128) 843 break; 844 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0; 845 } 846 } else { 847 n = get_bits(gb, 8); 848 for (k = 0; k < 5; k++) 849 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; 850 } 851 for (k = 0; k < 5; k++) 852 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx); 853 } else { 854 for (k = 0; k < 10; k++) 855 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 856 } 857 run = 10; 858 break; 859 860 case 10: 861 if (BITS_LEFT(length,gb) >= 1) { 862 float f = 0.81; 863 864 if (get_bits1(gb)) 865 f = -f; 866 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0; 867 samples[0] = f; 868 } else { 869 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 870 } 871 run = 1; 872 break; 873 874 case 16: 875 if (BITS_LEFT(length,gb) >= 10) { 876 if (zero_encoding) { 877 for (k = 0; k < 5; k++) { 878 if ((j + k) >= 128) 879 break; 880 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)]; 881 } 882 } else { 883 n = get_bits (gb, 8); 884 for (k = 0; k < 5; k++) 885 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; 886 } 887 } else { 888 for (k = 0; k < 5; k++) 889 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 890 } 891 run = 5; 892 break; 893 894 case 24: 895 if (BITS_LEFT(length,gb) >= 7) { 896 n = get_bits(gb, 7); 897 for (k = 0; k < 3; k++) 898 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5; 899 } else { 900 for (k = 0; k < 3; k++) 901 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 902 } 903 run = 3; 904 break; 905 906 case 30: 907 if (BITS_LEFT(length,gb) >= 4) 908 samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)]; 909 else 910 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 911 912 run = 1; 913 break; 914 915 case 34: 916 if (BITS_LEFT(length,gb) >= 7) { 917 if (type34_first) { 918 type34_div = (float)(1 << get_bits(gb, 2)); 919 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0; 920 type34_predictor = samples[0]; 921 type34_first = 0; 922 } else { 923 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor; 924 type34_predictor = samples[0]; 925 } 926 } else { 927 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 928 } 929 run = 1; 930 break; 931 932 default: 933 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 934 run = 1; 935 break; 936 } 937 938 if (joined_stereo) { 939 float tmp[10][MPA_MAX_CHANNELS]; 940 941 for (k = 0; k < run; k++) { 942 tmp[k][0] = samples[k]; 943 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k]; 944 } 945 for (chs = 0; chs < q->nb_channels; chs++) 946 for (k = 0; k < run; k++) 947 if ((j + k) < 128) 948 q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5); 949 } else { 950 for (k = 0; k < run; k++) 951 if ((j + k) < 128) 952 q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5); 953 } 954 955 j += run; 956 } // j loop 957 } // channel loop 958 } // subband loop 959} 960 961 962/** 963 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]). 964 * This is similar to process_subpacket_9, but for a single channel and for element [0] 965 * same VLC tables as process_subpacket_9 are used. 966 * 967 * @param q context 968 * @param quantized_coeffs pointer to quantized_coeffs[ch][0] 969 * @param gb bitreader context 970 * @param length packet length in bits 971 */ 972static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length) 973{ 974 int i, k, run, level, diff; 975 976 if (BITS_LEFT(length,gb) < 16) 977 return; 978 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); 979 980 quantized_coeffs[0] = level; 981 982 for (i = 0; i < 7; ) { 983 if (BITS_LEFT(length,gb) < 16) 984 break; 985 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; 986 987 if (BITS_LEFT(length,gb) < 16) 988 break; 989 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); 990 991 for (k = 1; k <= run; k++) 992 quantized_coeffs[i + k] = (level + ((k * diff) / run)); 993 994 level += diff; 995 i += run; 996 } 997} 998 999 1000/** 1001 * Related to synthesis filter, process data from packet 10 1002 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0 1003 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10 1004 * 1005 * @param q context 1006 * @param gb bitreader context 1007 * @param length packet length in bits 1008 */ 1009static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length) 1010{ 1011 int sb, j, k, n, ch; 1012 1013 for (ch = 0; ch < q->nb_channels; ch++) { 1014 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length); 1015 1016 if (BITS_LEFT(length,gb) < 16) { 1017 memset(q->quantized_coeffs[ch][0], 0, 8); 1018 break; 1019 } 1020 } 1021 1022 n = q->sub_sampling + 1; 1023 1024 for (sb = 0; sb < n; sb++) 1025 for (ch = 0; ch < q->nb_channels; ch++) 1026 for (j = 0; j < 8; j++) { 1027 if (BITS_LEFT(length,gb) < 1) 1028 break; 1029 if (get_bits1(gb)) { 1030 for (k=0; k < 8; k++) { 1031 if (BITS_LEFT(length,gb) < 16) 1032 break; 1033 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2); 1034 } 1035 } else { 1036 for (k=0; k < 8; k++) 1037 q->tone_level_idx_hi1[ch][sb][j][k] = 0; 1038 } 1039 } 1040 1041 n = QDM2_SB_USED(q->sub_sampling) - 4; 1042 1043 for (sb = 0; sb < n; sb++) 1044 for (ch = 0; ch < q->nb_channels; ch++) { 1045 if (BITS_LEFT(length,gb) < 16) 1046 break; 1047 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2); 1048 if (sb > 19) 1049 q->tone_level_idx_hi2[ch][sb] -= 16; 1050 else 1051 for (j = 0; j < 8; j++) 1052 q->tone_level_idx_mid[ch][sb][j] = -16; 1053 } 1054 1055 n = QDM2_SB_USED(q->sub_sampling) - 5; 1056 1057 for (sb = 0; sb < n; sb++) 1058 for (ch = 0; ch < q->nb_channels; ch++) 1059 for (j = 0; j < 8; j++) { 1060 if (BITS_LEFT(length,gb) < 16) 1061 break; 1062 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32; 1063 } 1064} 1065 1066/** 1067 * Process subpacket 9, init quantized_coeffs with data from it 1068 * 1069 * @param q context 1070 * @param node pointer to node with packet 1071 */ 1072static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node) 1073{ 1074 GetBitContext gb; 1075 int i, j, k, n, ch, run, level, diff; 1076 1077 init_get_bits(&gb, node->packet->data, node->packet->size*8); 1078 1079 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function 1080 1081 for (i = 1; i < n; i++) 1082 for (ch=0; ch < q->nb_channels; ch++) { 1083 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2); 1084 q->quantized_coeffs[ch][i][0] = level; 1085 1086 for (j = 0; j < (8 - 1); ) { 1087 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1; 1088 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2); 1089 1090 for (k = 1; k <= run; k++) 1091 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run)); 1092 1093 level += diff; 1094 j += run; 1095 } 1096 } 1097 1098 for (ch = 0; ch < q->nb_channels; ch++) 1099 for (i = 0; i < 8; i++) 1100 q->quantized_coeffs[ch][0][i] = 0; 1101} 1102 1103 1104/** 1105 * Process subpacket 10 if not null, else 1106 * 1107 * @param q context 1108 * @param node pointer to node with packet 1109 * @param length packet length in bits 1110 */ 1111static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length) 1112{ 1113 GetBitContext gb; 1114 1115 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); 1116 1117 if (length != 0) { 1118 init_tone_level_dequantization(q, &gb, length); 1119 fill_tone_level_array(q, 1); 1120 } else { 1121 fill_tone_level_array(q, 0); 1122 } 1123} 1124 1125 1126/** 1127 * Process subpacket 11 1128 * 1129 * @param q context 1130 * @param node pointer to node with packet 1131 * @param length packet length in bit 1132 */ 1133static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length) 1134{ 1135 GetBitContext gb; 1136 1137 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); 1138 if (length >= 32) { 1139 int c = get_bits (&gb, 13); 1140 1141 if (c > 3) 1142 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method, 1143 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select); 1144 } 1145 1146 synthfilt_build_sb_samples(q, &gb, length, 0, 8); 1147} 1148 1149 1150/** 1151 * Process subpacket 12 1152 * 1153 * @param q context 1154 * @param node pointer to node with packet 1155 * @param length packet length in bits 1156 */ 1157static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length) 1158{ 1159 GetBitContext gb; 1160 1161 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); 1162 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling)); 1163} 1164 1165/* 1166 * Process new subpackets for synthesis filter 1167 * 1168 * @param q context 1169 * @param list list with synthesis filter packets (list D) 1170 */ 1171static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list) 1172{ 1173 QDM2SubPNode *nodes[4]; 1174 1175 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9); 1176 if (nodes[0] != NULL) 1177 process_subpacket_9(q, nodes[0]); 1178 1179 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10); 1180 if (nodes[1] != NULL) 1181 process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3); 1182 else 1183 process_subpacket_10(q, NULL, 0); 1184 1185 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11); 1186 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL) 1187 process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3)); 1188 else 1189 process_subpacket_11(q, NULL, 0); 1190 1191 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12); 1192 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL) 1193 process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3)); 1194 else 1195 process_subpacket_12(q, NULL, 0); 1196} 1197 1198 1199/* 1200 * Decode superblock, fill packet lists. 1201 * 1202 * @param q context 1203 */ 1204static void qdm2_decode_super_block (QDM2Context *q) 1205{ 1206 GetBitContext gb; 1207 QDM2SubPacket header, *packet; 1208 int i, packet_bytes, sub_packet_size, sub_packets_D; 1209 unsigned int next_index = 0; 1210 1211 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1)); 1212 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid)); 1213 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2)); 1214 1215 q->sub_packets_B = 0; 1216 sub_packets_D = 0; 1217 1218 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8] 1219 1220 init_get_bits(&gb, q->compressed_data, q->compressed_size*8); 1221 qdm2_decode_sub_packet_header(&gb, &header); 1222 1223 if (header.type < 2 || header.type >= 8) { 1224 q->has_errors = 1; 1225 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n"); 1226 return; 1227 } 1228 1229 q->superblocktype_2_3 = (header.type == 2 || header.type == 3); 1230 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8); 1231 1232 init_get_bits(&gb, header.data, header.size*8); 1233 1234 if (header.type == 2 || header.type == 4 || header.type == 5) { 1235 int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8); 1236 1237 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); 1238 1239 if (csum != 0) { 1240 q->has_errors = 1; 1241 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n"); 1242 return; 1243 } 1244 } 1245 1246 q->sub_packet_list_B[0].packet = NULL; 1247 q->sub_packet_list_D[0].packet = NULL; 1248 1249 for (i = 0; i < 6; i++) 1250 if (--q->fft_level_exp[i] < 0) 1251 q->fft_level_exp[i] = 0; 1252 1253 for (i = 0; packet_bytes > 0; i++) { 1254 int j; 1255 1256 q->sub_packet_list_A[i].next = NULL; 1257 1258 if (i > 0) { 1259 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i]; 1260 1261 /* seek to next block */ 1262 init_get_bits(&gb, header.data, header.size*8); 1263 skip_bits(&gb, next_index*8); 1264 1265 if (next_index >= header.size) 1266 break; 1267 } 1268 1269 /* decode subpacket */ 1270 packet = &q->sub_packets[i]; 1271 qdm2_decode_sub_packet_header(&gb, packet); 1272 next_index = packet->size + get_bits_count(&gb) / 8; 1273 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2; 1274 1275 if (packet->type == 0) 1276 break; 1277 1278 if (sub_packet_size > packet_bytes) { 1279 if (packet->type != 10 && packet->type != 11 && packet->type != 12) 1280 break; 1281 packet->size += packet_bytes - sub_packet_size; 1282 } 1283 1284 packet_bytes -= sub_packet_size; 1285 1286 /* add subpacket to 'all subpackets' list */ 1287 q->sub_packet_list_A[i].packet = packet; 1288 1289 /* add subpacket to related list */ 1290 if (packet->type == 8) { 1291 SAMPLES_NEEDED_2("packet type 8"); 1292 return; 1293 } else if (packet->type >= 9 && packet->type <= 12) { 1294 /* packets for MPEG Audio like Synthesis Filter */ 1295 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet); 1296 } else if (packet->type == 13) { 1297 for (j = 0; j < 6; j++) 1298 q->fft_level_exp[j] = get_bits(&gb, 6); 1299 } else if (packet->type == 14) { 1300 for (j = 0; j < 6; j++) 1301 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2); 1302 } else if (packet->type == 15) { 1303 SAMPLES_NEEDED_2("packet type 15") 1304 return; 1305 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) { 1306 /* packets for FFT */ 1307 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet); 1308 } 1309 } // Packet bytes loop 1310 1311/* **************************************************************** */ 1312 if (q->sub_packet_list_D[0].packet != NULL) { 1313 process_synthesis_subpackets(q, q->sub_packet_list_D); 1314 q->do_synth_filter = 1; 1315 } else if (q->do_synth_filter) { 1316 process_subpacket_10(q, NULL, 0); 1317 process_subpacket_11(q, NULL, 0); 1318 process_subpacket_12(q, NULL, 0); 1319 } 1320/* **************************************************************** */ 1321} 1322 1323 1324static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet, 1325 int offset, int duration, int channel, 1326 int exp, int phase) 1327{ 1328 if (q->fft_coefs_min_index[duration] < 0) 1329 q->fft_coefs_min_index[duration] = q->fft_coefs_index; 1330 1331 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); 1332 q->fft_coefs[q->fft_coefs_index].channel = channel; 1333 q->fft_coefs[q->fft_coefs_index].offset = offset; 1334 q->fft_coefs[q->fft_coefs_index].exp = exp; 1335 q->fft_coefs[q->fft_coefs_index].phase = phase; 1336 q->fft_coefs_index++; 1337} 1338 1339 1340static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b) 1341{ 1342 int channel, stereo, phase, exp; 1343 int local_int_4, local_int_8, stereo_phase, local_int_10; 1344 int local_int_14, stereo_exp, local_int_20, local_int_28; 1345 int n, offset; 1346 1347 local_int_4 = 0; 1348 local_int_28 = 0; 1349 local_int_20 = 2; 1350 local_int_8 = (4 - duration); 1351 local_int_10 = 1 << (q->group_order - duration - 1); 1352 offset = 1; 1353 1354 while (1) { 1355 if (q->superblocktype_2_3) { 1356 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { 1357 offset = 1; 1358 if (n == 0) { 1359 local_int_4 += local_int_10; 1360 local_int_28 += (1 << local_int_8); 1361 } else { 1362 local_int_4 += 8*local_int_10; 1363 local_int_28 += (8 << local_int_8); 1364 } 1365 } 1366 offset += (n - 2); 1367 } else { 1368 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2); 1369 while (offset >= (local_int_10 - 1)) { 1370 offset += (1 - (local_int_10 - 1)); 1371 local_int_4 += local_int_10; 1372 local_int_28 += (1 << local_int_8); 1373 } 1374 } 1375 1376 if (local_int_4 >= q->group_size) 1377 return; 1378 1379 local_int_14 = (offset >> local_int_8); 1380 1381 if (q->nb_channels > 1) { 1382 channel = get_bits1(gb); 1383 stereo = get_bits1(gb); 1384 } else { 1385 channel = 0; 1386 stereo = 0; 1387 } 1388 1389 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); 1390 exp += q->fft_level_exp[fft_level_index_table[local_int_14]]; 1391 exp = (exp < 0) ? 0 : exp; 1392 1393 phase = get_bits(gb, 3); 1394 stereo_exp = 0; 1395 stereo_phase = 0; 1396 1397 if (stereo) { 1398 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1)); 1399 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1)); 1400 if (stereo_phase < 0) 1401 stereo_phase += 8; 1402 } 1403 1404 if (q->frequency_range > (local_int_14 + 1)) { 1405 int sub_packet = (local_int_20 + local_int_28); 1406 1407 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase); 1408 if (stereo) 1409 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase); 1410 } 1411 1412 offset++; 1413 } 1414} 1415 1416 1417static void qdm2_decode_fft_packets (QDM2Context *q) 1418{ 1419 int i, j, min, max, value, type, unknown_flag; 1420 GetBitContext gb; 1421 1422 if (q->sub_packet_list_B[0].packet == NULL) 1423 return; 1424 1425 /* reset minimum indexes for FFT coefficients */ 1426 q->fft_coefs_index = 0; 1427 for (i=0; i < 5; i++) 1428 q->fft_coefs_min_index[i] = -1; 1429 1430 /* process subpackets ordered by type, largest type first */ 1431 for (i = 0, max = 256; i < q->sub_packets_B; i++) { 1432 QDM2SubPacket *packet= NULL; 1433 1434 /* find subpacket with largest type less than max */ 1435 for (j = 0, min = 0; j < q->sub_packets_B; j++) { 1436 value = q->sub_packet_list_B[j].packet->type; 1437 if (value > min && value < max) { 1438 min = value; 1439 packet = q->sub_packet_list_B[j].packet; 1440 } 1441 } 1442 1443 max = min; 1444 1445 /* check for errors (?) */ 1446 if (!packet) 1447 return; 1448 1449 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16])) 1450 return; 1451 1452 /* decode FFT tones */ 1453 init_get_bits (&gb, packet->data, packet->size*8); 1454 1455 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) 1456 unknown_flag = 1; 1457 else 1458 unknown_flag = 0; 1459 1460 type = packet->type; 1461 1462 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) { 1463 int duration = q->sub_sampling + 5 - (type & 15); 1464 1465 if (duration >= 0 && duration < 4) 1466 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); 1467 } else if (type == 31) { 1468 for (j=0; j < 4; j++) 1469 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); 1470 } else if (type == 46) { 1471 for (j=0; j < 6; j++) 1472 q->fft_level_exp[j] = get_bits(&gb, 6); 1473 for (j=0; j < 4; j++) 1474 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); 1475 } 1476 } // Loop on B packets 1477 1478 /* calculate maximum indexes for FFT coefficients */ 1479 for (i = 0, j = -1; i < 5; i++) 1480 if (q->fft_coefs_min_index[i] >= 0) { 1481 if (j >= 0) 1482 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i]; 1483 j = i; 1484 } 1485 if (j >= 0) 1486 q->fft_coefs_max_index[j] = q->fft_coefs_index; 1487} 1488 1489 1490static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone) 1491{ 1492 float level, f[6]; 1493 int i; 1494 QDM2Complex c; 1495 const double iscale = 2.0*M_PI / 512.0; 1496 1497 tone->phase += tone->phase_shift; 1498 1499 /* calculate current level (maximum amplitude) of tone */ 1500 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level; 1501 c.im = level * sin(tone->phase*iscale); 1502 c.re = level * cos(tone->phase*iscale); 1503 1504 /* generate FFT coefficients for tone */ 1505 if (tone->duration >= 3 || tone->cutoff >= 3) { 1506 tone->complex[0].im += c.im; 1507 tone->complex[0].re += c.re; 1508 tone->complex[1].im -= c.im; 1509 tone->complex[1].re -= c.re; 1510 } else { 1511 f[1] = -tone->table[4]; 1512 f[0] = tone->table[3] - tone->table[0]; 1513 f[2] = 1.0 - tone->table[2] - tone->table[3]; 1514 f[3] = tone->table[1] + tone->table[4] - 1.0; 1515 f[4] = tone->table[0] - tone->table[1]; 1516 f[5] = tone->table[2]; 1517 for (i = 0; i < 2; i++) { 1518 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i]; 1519 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]); 1520 } 1521 for (i = 0; i < 4; i++) { 1522 tone->complex[i].re += c.re * f[i+2]; 1523 tone->complex[i].im += c.im * f[i+2]; 1524 } 1525 } 1526 1527 /* copy the tone if it has not yet died out */ 1528 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) { 1529 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone)); 1530 q->fft_tone_end = (q->fft_tone_end + 1) % 1000; 1531 } 1532} 1533 1534 1535static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) 1536{ 1537 int i, j, ch; 1538 const double iscale = 0.25 * M_PI; 1539 1540 for (ch = 0; ch < q->channels; ch++) { 1541 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex)); 1542 } 1543 1544 1545 /* apply FFT tones with duration 4 (1 FFT period) */ 1546 if (q->fft_coefs_min_index[4] >= 0) 1547 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) { 1548 float level; 1549 QDM2Complex c; 1550 1551 if (q->fft_coefs[i].sub_packet != sub_packet) 1552 break; 1553 1554 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel; 1555 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]; 1556 1557 c.re = level * cos(q->fft_coefs[i].phase * iscale); 1558 c.im = level * sin(q->fft_coefs[i].phase * iscale); 1559 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re; 1560 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im; 1561 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re; 1562 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im; 1563 } 1564 1565 /* generate existing FFT tones */ 1566 for (i = q->fft_tone_end; i != q->fft_tone_start; ) { 1567 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]); 1568 q->fft_tone_start = (q->fft_tone_start + 1) % 1000; 1569 } 1570 1571 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */ 1572 for (i = 0; i < 4; i++) 1573 if (q->fft_coefs_min_index[i] >= 0) { 1574 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) { 1575 int offset, four_i; 1576 FFTTone tone; 1577 1578 if (q->fft_coefs[j].sub_packet != sub_packet) 1579 break; 1580 1581 four_i = (4 - i); 1582 offset = q->fft_coefs[j].offset >> four_i; 1583 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel; 1584 1585 if (offset < q->frequency_range) { 1586 if (offset < 2) 1587 tone.cutoff = offset; 1588 else 1589 tone.cutoff = (offset >= 60) ? 3 : 2; 1590 1591 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]; 1592 tone.complex = &q->fft.complex[ch][offset]; 1593 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)]; 1594 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128; 1595 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i); 1596 tone.duration = i; 1597 tone.time_index = 0; 1598 1599 qdm2_fft_generate_tone(q, &tone); 1600 } 1601 } 1602 q->fft_coefs_min_index[i] = j; 1603 } 1604} 1605 1606 1607static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) 1608{ 1609 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f; 1610 int i; 1611 q->fft.complex[channel][0].re *= 2.0f; 1612 q->fft.complex[channel][0].im = 0.0f; 1613 ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]); 1614 /* add samples to output buffer */ 1615 for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++) 1616 q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain; 1617} 1618 1619 1620/** 1621 * @param q context 1622 * @param index subpacket number 1623 */ 1624static void qdm2_synthesis_filter (QDM2Context *q, int index) 1625{ 1626 OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; 1627 int i, k, ch, sb_used, sub_sampling, dither_state = 0; 1628 1629 /* copy sb_samples */ 1630 sb_used = QDM2_SB_USED(q->sub_sampling); 1631 1632 for (ch = 0; ch < q->channels; ch++) 1633 for (i = 0; i < 8; i++) 1634 for (k=sb_used; k < SBLIMIT; k++) 1635 q->sb_samples[ch][(8 * index) + i][k] = 0; 1636 1637 for (ch = 0; ch < q->nb_channels; ch++) { 1638 OUT_INT *samples_ptr = samples + ch; 1639 1640 for (i = 0; i < 8; i++) { 1641 ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]), 1642 mpa_window, &dither_state, 1643 samples_ptr, q->nb_channels, 1644 q->sb_samples[ch][(8 * index) + i]); 1645 samples_ptr += 32 * q->nb_channels; 1646 } 1647 } 1648 1649 /* add samples to output buffer */ 1650 sub_sampling = (4 >> q->sub_sampling); 1651 1652 for (ch = 0; ch < q->channels; ch++) 1653 for (i = 0; i < q->frame_size; i++) 1654 q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16)); 1655} 1656 1657 1658/** 1659 * Init static data (does not depend on specific file) 1660 * 1661 * @param q context 1662 */ 1663static av_cold void qdm2_init(QDM2Context *q) { 1664 static int initialized = 0; 1665 1666 if (initialized != 0) 1667 return; 1668 initialized = 1; 1669 1670 qdm2_init_vlc(); 1671 ff_mpa_synth_init(mpa_window); 1672 softclip_table_init(); 1673 rnd_table_init(); 1674 init_noise_samples(); 1675 1676 av_log(NULL, AV_LOG_DEBUG, "init done\n"); 1677} 1678 1679 1680#if 0 1681static void dump_context(QDM2Context *q) 1682{ 1683 int i; 1684#define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b); 1685 PRINT("compressed_data",q->compressed_data); 1686 PRINT("compressed_size",q->compressed_size); 1687 PRINT("frame_size",q->frame_size); 1688 PRINT("checksum_size",q->checksum_size); 1689 PRINT("channels",q->channels); 1690 PRINT("nb_channels",q->nb_channels); 1691 PRINT("fft_frame_size",q->fft_frame_size); 1692 PRINT("fft_size",q->fft_size); 1693 PRINT("sub_sampling",q->sub_sampling); 1694 PRINT("fft_order",q->fft_order); 1695 PRINT("group_order",q->group_order); 1696 PRINT("group_size",q->group_size); 1697 PRINT("sub_packet",q->sub_packet); 1698 PRINT("frequency_range",q->frequency_range); 1699 PRINT("has_errors",q->has_errors); 1700 PRINT("fft_tone_end",q->fft_tone_end); 1701 PRINT("fft_tone_start",q->fft_tone_start); 1702 PRINT("fft_coefs_index",q->fft_coefs_index); 1703 PRINT("coeff_per_sb_select",q->coeff_per_sb_select); 1704 PRINT("cm_table_select",q->cm_table_select); 1705 PRINT("noise_idx",q->noise_idx); 1706 1707 for (i = q->fft_tone_start; i < q->fft_tone_end; i++) 1708 { 1709 FFTTone *t = &q->fft_tones[i]; 1710 1711 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i); 1712 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level); 1713// PRINT(" level", t->level); 1714 PRINT(" phase", t->phase); 1715 PRINT(" phase_shift", t->phase_shift); 1716 PRINT(" duration", t->duration); 1717 PRINT(" samples_im", t->samples_im); 1718 PRINT(" samples_re", t->samples_re); 1719 PRINT(" table", t->table); 1720 } 1721 1722} 1723#endif 1724 1725 1726/** 1727 * Init parameters from codec extradata 1728 */ 1729static av_cold int qdm2_decode_init(AVCodecContext *avctx) 1730{ 1731 QDM2Context *s = avctx->priv_data; 1732 uint8_t *extradata; 1733 int extradata_size; 1734 int tmp_val, tmp, size; 1735 1736 /* extradata parsing 1737 1738 Structure: 1739 wave { 1740 frma (QDM2) 1741 QDCA 1742 QDCP 1743 } 1744 1745 32 size (including this field) 1746 32 tag (=frma) 1747 32 type (=QDM2 or QDMC) 1748 1749 32 size (including this field, in bytes) 1750 32 tag (=QDCA) // maybe mandatory parameters 1751 32 unknown (=1) 1752 32 channels (=2) 1753 32 samplerate (=44100) 1754 32 bitrate (=96000) 1755 32 block size (=4096) 1756 32 frame size (=256) (for one channel) 1757 32 packet size (=1300) 1758 1759 32 size (including this field, in bytes) 1760 32 tag (=QDCP) // maybe some tuneable parameters 1761 32 float1 (=1.0) 1762 32 zero ? 1763 32 float2 (=1.0) 1764 32 float3 (=1.0) 1765 32 unknown (27) 1766 32 unknown (8) 1767 32 zero ? 1768 */ 1769 1770 if (!avctx->extradata || (avctx->extradata_size < 48)) { 1771 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n"); 1772 return -1; 1773 } 1774 1775 extradata = avctx->extradata; 1776 extradata_size = avctx->extradata_size; 1777 1778 while (extradata_size > 7) { 1779 if (!memcmp(extradata, "frmaQDM", 7)) 1780 break; 1781 extradata++; 1782 extradata_size--; 1783 } 1784 1785 if (extradata_size < 12) { 1786 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n", 1787 extradata_size); 1788 return -1; 1789 } 1790 1791 if (memcmp(extradata, "frmaQDM", 7)) { 1792 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n"); 1793 return -1; 1794 } 1795 1796 if (extradata[7] == 'C') { 1797// s->is_qdmc = 1; 1798 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n"); 1799 return -1; 1800 } 1801 1802 extradata += 8; 1803 extradata_size -= 8; 1804 1805 size = AV_RB32(extradata); 1806 1807 if(size > extradata_size){ 1808 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n", 1809 extradata_size, size); 1810 return -1; 1811 } 1812 1813 extradata += 4; 1814 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size); 1815 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) { 1816 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n"); 1817 return -1; 1818 } 1819 1820 extradata += 8; 1821 1822 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata); 1823 extradata += 4; 1824 1825 avctx->sample_rate = AV_RB32(extradata); 1826 extradata += 4; 1827 1828 avctx->bit_rate = AV_RB32(extradata); 1829 extradata += 4; 1830 1831 s->group_size = AV_RB32(extradata); 1832 extradata += 4; 1833 1834 s->fft_size = AV_RB32(extradata); 1835 extradata += 4; 1836 1837 s->checksum_size = AV_RB32(extradata); 1838 extradata += 4; 1839 1840 s->fft_order = av_log2(s->fft_size) + 1; 1841 s->fft_frame_size = 2 * s->fft_size; // complex has two floats 1842 1843 // something like max decodable tones 1844 s->group_order = av_log2(s->group_size) + 1; 1845 s->frame_size = s->group_size / 16; // 16 iterations per super block 1846 1847 s->sub_sampling = s->fft_order - 7; 1848 s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); 1849 1850 switch ((s->sub_sampling * 2 + s->channels - 1)) { 1851 case 0: tmp = 40; break; 1852 case 1: tmp = 48; break; 1853 case 2: tmp = 56; break; 1854 case 3: tmp = 72; break; 1855 case 4: tmp = 80; break; 1856 case 5: tmp = 100;break; 1857 default: tmp=s->sub_sampling; break; 1858 } 1859 tmp_val = 0; 1860 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1; 1861 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2; 1862 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3; 1863 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4; 1864 s->cm_table_select = tmp_val; 1865 1866 if (s->sub_sampling == 0) 1867 tmp = 7999; 1868 else 1869 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000; 1870 /* 1871 0: 7999 -> 0 1872 1: 20000 -> 2 1873 2: 28000 -> 2 1874 */ 1875 if (tmp < 8000) 1876 s->coeff_per_sb_select = 0; 1877 else if (tmp <= 16000) 1878 s->coeff_per_sb_select = 1; 1879 else 1880 s->coeff_per_sb_select = 2; 1881 1882 // Fail on unknown fft order 1883 if ((s->fft_order < 7) || (s->fft_order > 9)) { 1884 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order); 1885 return -1; 1886 } 1887 1888 ff_rdft_init(&s->rdft_ctx, s->fft_order, IRDFT); 1889 1890 qdm2_init(s); 1891 1892 avctx->sample_fmt = SAMPLE_FMT_S16; 1893 1894// dump_context(s); 1895 return 0; 1896} 1897 1898 1899static av_cold int qdm2_decode_close(AVCodecContext *avctx) 1900{ 1901 QDM2Context *s = avctx->priv_data; 1902 1903 ff_rdft_end(&s->rdft_ctx); 1904 1905 return 0; 1906} 1907 1908 1909static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) 1910{ 1911 int ch, i; 1912 const int frame_size = (q->frame_size * q->channels); 1913 1914 /* select input buffer */ 1915 q->compressed_data = in; 1916 q->compressed_size = q->checksum_size; 1917 1918// dump_context(q); 1919 1920 /* copy old block, clear new block of output samples */ 1921 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float)); 1922 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float)); 1923 1924 /* decode block of QDM2 compressed data */ 1925 if (q->sub_packet == 0) { 1926 q->has_errors = 0; // zero it for a new super block 1927 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n"); 1928 qdm2_decode_super_block(q); 1929 } 1930 1931 /* parse subpackets */ 1932 if (!q->has_errors) { 1933 if (q->sub_packet == 2) 1934 qdm2_decode_fft_packets(q); 1935 1936 qdm2_fft_tone_synthesizer(q, q->sub_packet); 1937 } 1938 1939 /* sound synthesis stage 1 (FFT) */ 1940 for (ch = 0; ch < q->channels; ch++) { 1941 qdm2_calculate_fft(q, ch, q->sub_packet); 1942 1943 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) { 1944 SAMPLES_NEEDED_2("has errors, and C list is not empty") 1945 return; 1946 } 1947 } 1948 1949 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */ 1950 if (!q->has_errors && q->do_synth_filter) 1951 qdm2_synthesis_filter(q, q->sub_packet); 1952 1953 q->sub_packet = (q->sub_packet + 1) % 16; 1954 1955 /* clip and convert output float[] to 16bit signed samples */ 1956 for (i = 0; i < frame_size; i++) { 1957 int value = (int)q->output_buffer[i]; 1958 1959 if (value > SOFTCLIP_THRESHOLD) 1960 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD]; 1961 else if (value < -SOFTCLIP_THRESHOLD) 1962 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD]; 1963 1964 out[i] = value; 1965 } 1966} 1967 1968 1969static int qdm2_decode_frame(AVCodecContext *avctx, 1970 void *data, int *data_size, 1971 const uint8_t *buf, int buf_size) 1972{ 1973 QDM2Context *s = avctx->priv_data; 1974 1975 if(!buf) 1976 return 0; 1977 if(buf_size < s->checksum_size) 1978 return -1; 1979 1980 *data_size = s->channels * s->frame_size * sizeof(int16_t); 1981 1982 av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n", 1983 buf_size, buf, s->checksum_size, data, *data_size); 1984 1985 qdm2_decode(s, buf, data); 1986 1987 // reading only when next superblock found 1988 if (s->sub_packet == 0) { 1989 return s->checksum_size; 1990 } 1991 1992 return 0; 1993} 1994 1995AVCodec qdm2_decoder = 1996{ 1997 .name = "qdm2", 1998 .type = CODEC_TYPE_AUDIO, 1999 .id = CODEC_ID_QDM2, 2000 .priv_data_size = sizeof(QDM2Context), 2001 .init = qdm2_decode_init, 2002 .close = qdm2_decode_close, 2003 .decode = qdm2_decode_frame, 2004 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), 2005}; 2006