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 * 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 "get_bits.h" 40#include "dsputil.h" 41#include "fft.h" 42#include "mpegaudio.h" 43 44#include "qdm2data.h" 45#include "qdm2_tablegen.h" 46 47#undef NDEBUG 48#include <assert.h> 49 50 51#define QDM2_LIST_ADD(list, size, packet) \ 52do { \ 53 if (size > 0) { \ 54 list[size - 1].next = &list[size]; \ 55 } \ 56 list[size].packet = packet; \ 57 list[size].next = NULL; \ 58 size++; \ 59} while(0) 60 61// Result is 8, 16 or 30 62#define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling)) 63 64#define FIX_NOISE_IDX(noise_idx) \ 65 if ((noise_idx) >= 3840) \ 66 (noise_idx) -= 3840; \ 67 68#define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)]) 69 70#define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb))) 71 72#define SAMPLES_NEEDED \ 73 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n"); 74 75#define SAMPLES_NEEDED_2(why) \ 76 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why); 77 78 79typedef int8_t sb_int8_array[2][30][64]; 80 81/** 82 * Subpacket 83 */ 84typedef struct { 85 int type; ///< subpacket type 86 unsigned int size; ///< subpacket size 87 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy) 88} QDM2SubPacket; 89 90/** 91 * A node in the subpacket list 92 */ 93typedef struct QDM2SubPNode { 94 QDM2SubPacket *packet; ///< packet 95 struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node 96} QDM2SubPNode; 97 98typedef struct { 99 float re; 100 float im; 101} QDM2Complex; 102 103typedef struct { 104 float level; 105 QDM2Complex *complex; 106 const float *table; 107 int phase; 108 int phase_shift; 109 int duration; 110 short time_index; 111 short cutoff; 112} FFTTone; 113 114typedef struct { 115 int16_t sub_packet; 116 uint8_t channel; 117 int16_t offset; 118 int16_t exp; 119 uint8_t phase; 120} FFTCoefficient; 121 122typedef struct { 123 DECLARE_ALIGNED(16, QDM2Complex, complex)[MPA_MAX_CHANNELS][256]; 124} QDM2FFT; 125 126/** 127 * QDM2 decoder context 128 */ 129typedef struct { 130 /// Parameters from codec header, do not change during playback 131 int nb_channels; ///< number of channels 132 int channels; ///< number of channels 133 int group_size; ///< size of frame group (16 frames per group) 134 int fft_size; ///< size of FFT, in complex numbers 135 int checksum_size; ///< size of data block, used also for checksum 136 137 /// Parameters built from header parameters, do not change during playback 138 int group_order; ///< order of frame group 139 int fft_order; ///< order of FFT (actually fftorder+1) 140 int fft_frame_size; ///< size of fft frame, in components (1 comples = re + im) 141 int frame_size; ///< size of data frame 142 int frequency_range; 143 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */ 144 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2 145 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4) 146 147 /// Packets and packet lists 148 QDM2SubPacket sub_packets[16]; ///< the packets themselves 149 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets 150 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list 151 int sub_packets_B; ///< number of packets on 'B' list 152 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors? 153 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets 154 155 /// FFT and tones 156 FFTTone fft_tones[1000]; 157 int fft_tone_start; 158 int fft_tone_end; 159 FFTCoefficient fft_coefs[1000]; 160 int fft_coefs_index; 161 int fft_coefs_min_index[5]; 162 int fft_coefs_max_index[5]; 163 int fft_level_exp[6]; 164 RDFTContext rdft_ctx; 165 QDM2FFT fft; 166 167 /// I/O data 168 const uint8_t *compressed_data; 169 int compressed_size; 170 float output_buffer[1024]; 171 172 /// Synthesis filter 173 DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512*2]; 174 int synth_buf_offset[MPA_MAX_CHANNELS]; 175 DECLARE_ALIGNED(16, int32_t, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT]; 176 177 /// Mixed temporary data used in decoding 178 float tone_level[MPA_MAX_CHANNELS][30][64]; 179 int8_t coding_method[MPA_MAX_CHANNELS][30][64]; 180 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8]; 181 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8]; 182 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8]; 183 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8]; 184 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26]; 185 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64]; 186 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]; 187 188 // Flags 189 int has_errors; ///< packet has errors 190 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type 191 int do_synth_filter; ///< used to perform or skip synthesis filter 192 193 int sub_packet; 194 int noise_idx; ///< index for dithering noise table 195} QDM2Context; 196 197 198static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE]; 199 200static VLC vlc_tab_level; 201static VLC vlc_tab_diff; 202static VLC vlc_tab_run; 203static VLC fft_level_exp_alt_vlc; 204static VLC fft_level_exp_vlc; 205static VLC fft_stereo_exp_vlc; 206static VLC fft_stereo_phase_vlc; 207static VLC vlc_tab_tone_level_idx_hi1; 208static VLC vlc_tab_tone_level_idx_mid; 209static VLC vlc_tab_tone_level_idx_hi2; 210static VLC vlc_tab_type30; 211static VLC vlc_tab_type34; 212static VLC vlc_tab_fft_tone_offset[5]; 213 214static const uint16_t qdm2_vlc_offs[] = { 215 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838, 216}; 217 218static av_cold void qdm2_init_vlc(void) 219{ 220 static int vlcs_initialized = 0; 221 static VLC_TYPE qdm2_table[3838][2]; 222 223 if (!vlcs_initialized) { 224 225 vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]]; 226 vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0]; 227 init_vlc (&vlc_tab_level, 8, 24, 228 vlc_tab_level_huffbits, 1, 1, 229 vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 230 231 vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]]; 232 vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1]; 233 init_vlc (&vlc_tab_diff, 8, 37, 234 vlc_tab_diff_huffbits, 1, 1, 235 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 236 237 vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]]; 238 vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2]; 239 init_vlc (&vlc_tab_run, 5, 6, 240 vlc_tab_run_huffbits, 1, 1, 241 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 242 243 fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]]; 244 fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3]; 245 init_vlc (&fft_level_exp_alt_vlc, 8, 28, 246 fft_level_exp_alt_huffbits, 1, 1, 247 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 248 249 250 fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]]; 251 fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4]; 252 init_vlc (&fft_level_exp_vlc, 8, 20, 253 fft_level_exp_huffbits, 1, 1, 254 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 255 256 fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]]; 257 fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5]; 258 init_vlc (&fft_stereo_exp_vlc, 6, 7, 259 fft_stereo_exp_huffbits, 1, 1, 260 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 261 262 fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]]; 263 fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6]; 264 init_vlc (&fft_stereo_phase_vlc, 6, 9, 265 fft_stereo_phase_huffbits, 1, 1, 266 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 267 268 vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]]; 269 vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7]; 270 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20, 271 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1, 272 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 273 274 vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]]; 275 vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8]; 276 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24, 277 vlc_tab_tone_level_idx_mid_huffbits, 1, 1, 278 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 279 280 vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]]; 281 vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9]; 282 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24, 283 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1, 284 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 285 286 vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]]; 287 vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10]; 288 init_vlc (&vlc_tab_type30, 6, 9, 289 vlc_tab_type30_huffbits, 1, 1, 290 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 291 292 vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]]; 293 vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11]; 294 init_vlc (&vlc_tab_type34, 5, 10, 295 vlc_tab_type34_huffbits, 1, 1, 296 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 297 298 vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]]; 299 vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12]; 300 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23, 301 vlc_tab_fft_tone_offset_0_huffbits, 1, 1, 302 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 303 304 vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]]; 305 vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13]; 306 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28, 307 vlc_tab_fft_tone_offset_1_huffbits, 1, 1, 308 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 309 310 vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]]; 311 vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14]; 312 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32, 313 vlc_tab_fft_tone_offset_2_huffbits, 1, 1, 314 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 315 316 vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]]; 317 vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15]; 318 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35, 319 vlc_tab_fft_tone_offset_3_huffbits, 1, 1, 320 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 321 322 vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]]; 323 vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16]; 324 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38, 325 vlc_tab_fft_tone_offset_4_huffbits, 1, 1, 326 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 327 328 vlcs_initialized=1; 329 } 330} 331 332 333/* for floating point to fixed point conversion */ 334static const float f2i_scale = (float) (1 << (FRAC_BITS - 15)); 335 336 337static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth) 338{ 339 int value; 340 341 value = get_vlc2(gb, vlc->table, vlc->bits, depth); 342 343 /* stage-2, 3 bits exponent escape sequence */ 344 if (value-- == 0) 345 value = get_bits (gb, get_bits (gb, 3) + 1); 346 347 /* stage-3, optional */ 348 if (flag) { 349 int tmp = vlc_stage3_values[value]; 350 351 if ((value & ~3) > 0) 352 tmp += get_bits (gb, (value >> 2)); 353 value = tmp; 354 } 355 356 return value; 357} 358 359 360static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth) 361{ 362 int value = qdm2_get_vlc (gb, vlc, 0, depth); 363 364 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1); 365} 366 367 368/** 369 * QDM2 checksum 370 * 371 * @param data pointer to data to be checksum'ed 372 * @param length data length 373 * @param value checksum value 374 * 375 * @return 0 if checksum is OK 376 */ 377static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) { 378 int i; 379 380 for (i=0; i < length; i++) 381 value -= data[i]; 382 383 return (uint16_t)(value & 0xffff); 384} 385 386 387/** 388 * Fills a QDM2SubPacket structure with packet type, size, and data pointer. 389 * 390 * @param gb bitreader context 391 * @param sub_packet packet under analysis 392 */ 393static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet) 394{ 395 sub_packet->type = get_bits (gb, 8); 396 397 if (sub_packet->type == 0) { 398 sub_packet->size = 0; 399 sub_packet->data = NULL; 400 } else { 401 sub_packet->size = get_bits (gb, 8); 402 403 if (sub_packet->type & 0x80) { 404 sub_packet->size <<= 8; 405 sub_packet->size |= get_bits (gb, 8); 406 sub_packet->type &= 0x7f; 407 } 408 409 if (sub_packet->type == 0x7f) 410 sub_packet->type |= (get_bits (gb, 8) << 8); 411 412 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data 413 } 414 415 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n", 416 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8); 417} 418 419 420/** 421 * Return node pointer to first packet of requested type in list. 422 * 423 * @param list list of subpackets to be scanned 424 * @param type type of searched subpacket 425 * @return node pointer for subpacket if found, else NULL 426 */ 427static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type) 428{ 429 while (list != NULL && list->packet != NULL) { 430 if (list->packet->type == type) 431 return list; 432 list = list->next; 433 } 434 return NULL; 435} 436 437 438/** 439 * Replaces 8 elements with their average value. 440 * Called by qdm2_decode_superblock before starting subblock decoding. 441 * 442 * @param q context 443 */ 444static void average_quantized_coeffs (QDM2Context *q) 445{ 446 int i, j, n, ch, sum; 447 448 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; 449 450 for (ch = 0; ch < q->nb_channels; ch++) 451 for (i = 0; i < n; i++) { 452 sum = 0; 453 454 for (j = 0; j < 8; j++) 455 sum += q->quantized_coeffs[ch][i][j]; 456 457 sum /= 8; 458 if (sum > 0) 459 sum--; 460 461 for (j=0; j < 8; j++) 462 q->quantized_coeffs[ch][i][j] = sum; 463 } 464} 465 466 467/** 468 * Build subband samples with noise weighted by q->tone_level. 469 * Called by synthfilt_build_sb_samples. 470 * 471 * @param q context 472 * @param sb subband index 473 */ 474static void build_sb_samples_from_noise (QDM2Context *q, int sb) 475{ 476 int ch, j; 477 478 FIX_NOISE_IDX(q->noise_idx); 479 480 if (!q->nb_channels) 481 return; 482 483 for (ch = 0; ch < q->nb_channels; ch++) 484 for (j = 0; j < 64; j++) { 485 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); 486 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); 487 } 488} 489 490 491/** 492 * Called while processing data from subpackets 11 and 12. 493 * Used after making changes to coding_method array. 494 * 495 * @param sb subband index 496 * @param channels number of channels 497 * @param coding_method q->coding_method[0][0][0] 498 */ 499static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method) 500{ 501 int j,k; 502 int ch; 503 int run, case_val; 504 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}; 505 506 for (ch = 0; ch < channels; ch++) { 507 for (j = 0; j < 64; ) { 508 if((coding_method[ch][sb][j] - 8) > 22) { 509 run = 1; 510 case_val = 8; 511 } else { 512 switch (switchtable[coding_method[ch][sb][j]-8]) { 513 case 0: run = 10; case_val = 10; break; 514 case 1: run = 1; case_val = 16; break; 515 case 2: run = 5; case_val = 24; break; 516 case 3: run = 3; case_val = 30; break; 517 case 4: run = 1; case_val = 30; break; 518 case 5: run = 1; case_val = 8; break; 519 default: run = 1; case_val = 8; break; 520 } 521 } 522 for (k = 0; k < run; k++) 523 if (j + k < 128) 524 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) 525 if (k > 0) { 526 SAMPLES_NEEDED 527 //not debugged, almost never used 528 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t)); 529 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t)); 530 } 531 j += run; 532 } 533 } 534} 535 536 537/** 538 * Related to synthesis filter 539 * Called by process_subpacket_10 540 * 541 * @param q context 542 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10 543 */ 544static void fill_tone_level_array (QDM2Context *q, int flag) 545{ 546 int i, sb, ch, sb_used; 547 int tmp, tab; 548 549 // This should never happen 550 if (q->nb_channels <= 0) 551 return; 552 553 for (ch = 0; ch < q->nb_channels; ch++) 554 for (sb = 0; sb < 30; sb++) 555 for (i = 0; i < 8; i++) { 556 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1)) 557 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+ 558 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; 559 else 560 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; 561 if(tmp < 0) 562 tmp += 0xff; 563 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff; 564 } 565 566 sb_used = QDM2_SB_USED(q->sub_sampling); 567 568 if ((q->superblocktype_2_3 != 0) && !flag) { 569 for (sb = 0; sb < sb_used; sb++) 570 for (ch = 0; ch < q->nb_channels; ch++) 571 for (i = 0; i < 64; i++) { 572 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; 573 if (q->tone_level_idx[ch][sb][i] < 0) 574 q->tone_level[ch][sb][i] = 0; 575 else 576 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f]; 577 } 578 } else { 579 tab = q->superblocktype_2_3 ? 0 : 1; 580 for (sb = 0; sb < sb_used; sb++) { 581 if ((sb >= 4) && (sb <= 23)) { 582 for (ch = 0; ch < q->nb_channels; ch++) 583 for (i = 0; i < 64; i++) { 584 tmp = q->tone_level_idx_base[ch][sb][i / 8] - 585 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] - 586 q->tone_level_idx_mid[ch][sb - 4][i / 8] - 587 q->tone_level_idx_hi2[ch][sb - 4]; 588 q->tone_level_idx[ch][sb][i] = tmp & 0xff; 589 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 590 q->tone_level[ch][sb][i] = 0; 591 else 592 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 593 } 594 } else { 595 if (sb > 4) { 596 for (ch = 0; ch < q->nb_channels; ch++) 597 for (i = 0; i < 64; i++) { 598 tmp = q->tone_level_idx_base[ch][sb][i / 8] - 599 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] - 600 q->tone_level_idx_hi2[ch][sb - 4]; 601 q->tone_level_idx[ch][sb][i] = tmp & 0xff; 602 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 603 q->tone_level[ch][sb][i] = 0; 604 else 605 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 606 } 607 } else { 608 for (ch = 0; ch < q->nb_channels; ch++) 609 for (i = 0; i < 64; i++) { 610 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; 611 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 612 q->tone_level[ch][sb][i] = 0; 613 else 614 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 615 } 616 } 617 } 618 } 619 } 620 621 return; 622} 623 624 625/** 626 * Related to synthesis filter 627 * Called by process_subpacket_11 628 * c is built with data from subpacket 11 629 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples 630 * 631 * @param tone_level_idx 632 * @param tone_level_idx_temp 633 * @param coding_method q->coding_method[0][0][0] 634 * @param nb_channels number of channels 635 * @param c coming from subpacket 11, passed as 8*c 636 * @param superblocktype_2_3 flag based on superblock packet type 637 * @param cm_table_select q->cm_table_select 638 */ 639static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp, 640 sb_int8_array coding_method, int nb_channels, 641 int c, int superblocktype_2_3, int cm_table_select) 642{ 643 int ch, sb, j; 644 int tmp, acc, esp_40, comp; 645 int add1, add2, add3, add4; 646 int64_t multres; 647 648 // This should never happen 649 if (nb_channels <= 0) 650 return; 651 652 if (!superblocktype_2_3) { 653 /* This case is untested, no samples available */ 654 SAMPLES_NEEDED 655 for (ch = 0; ch < nb_channels; ch++) 656 for (sb = 0; sb < 30; sb++) { 657 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer 658 add1 = tone_level_idx[ch][sb][j] - 10; 659 if (add1 < 0) 660 add1 = 0; 661 add2 = add3 = add4 = 0; 662 if (sb > 1) { 663 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6; 664 if (add2 < 0) 665 add2 = 0; 666 } 667 if (sb > 0) { 668 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6; 669 if (add3 < 0) 670 add3 = 0; 671 } 672 if (sb < 29) { 673 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6; 674 if (add4 < 0) 675 add4 = 0; 676 } 677 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1; 678 if (tmp < 0) 679 tmp = 0; 680 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff; 681 } 682 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1]; 683 } 684 acc = 0; 685 for (ch = 0; ch < nb_channels; ch++) 686 for (sb = 0; sb < 30; sb++) 687 for (j = 0; j < 64; j++) 688 acc += tone_level_idx_temp[ch][sb][j]; 689 690 multres = 0x66666667 * (acc * 10); 691 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31); 692 for (ch = 0; ch < nb_channels; ch++) 693 for (sb = 0; sb < 30; sb++) 694 for (j = 0; j < 64; j++) { 695 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10; 696 if (comp < 0) 697 comp += 0xff; 698 comp /= 256; // signed shift 699 switch(sb) { 700 case 0: 701 if (comp < 30) 702 comp = 30; 703 comp += 15; 704 break; 705 case 1: 706 if (comp < 24) 707 comp = 24; 708 comp += 10; 709 break; 710 case 2: 711 case 3: 712 case 4: 713 if (comp < 16) 714 comp = 16; 715 } 716 if (comp <= 5) 717 tmp = 0; 718 else if (comp <= 10) 719 tmp = 10; 720 else if (comp <= 16) 721 tmp = 16; 722 else if (comp <= 24) 723 tmp = -1; 724 else 725 tmp = 0; 726 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff; 727 } 728 for (sb = 0; sb < 30; sb++) 729 fix_coding_method_array(sb, nb_channels, coding_method); 730 for (ch = 0; ch < nb_channels; ch++) 731 for (sb = 0; sb < 30; sb++) 732 for (j = 0; j < 64; j++) 733 if (sb >= 10) { 734 if (coding_method[ch][sb][j] < 10) 735 coding_method[ch][sb][j] = 10; 736 } else { 737 if (sb >= 2) { 738 if (coding_method[ch][sb][j] < 16) 739 coding_method[ch][sb][j] = 16; 740 } else { 741 if (coding_method[ch][sb][j] < 30) 742 coding_method[ch][sb][j] = 30; 743 } 744 } 745 } else { // superblocktype_2_3 != 0 746 for (ch = 0; ch < nb_channels; ch++) 747 for (sb = 0; sb < 30; sb++) 748 for (j = 0; j < 64; j++) 749 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb]; 750 } 751 752 return; 753} 754 755 756/** 757 * 758 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8 759 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used 760 * 761 * @param q context 762 * @param gb bitreader context 763 * @param length packet length in bits 764 * @param sb_min lower subband processed (sb_min included) 765 * @param sb_max higher subband processed (sb_max excluded) 766 */ 767static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max) 768{ 769 int sb, j, k, n, ch, run, channels; 770 int joined_stereo, zero_encoding, chs; 771 int type34_first; 772 float type34_div = 0; 773 float type34_predictor; 774 float samples[10], sign_bits[16]; 775 776 if (length == 0) { 777 // If no data use noise 778 for (sb=sb_min; sb < sb_max; sb++) 779 build_sb_samples_from_noise (q, sb); 780 781 return; 782 } 783 784 for (sb = sb_min; sb < sb_max; sb++) { 785 FIX_NOISE_IDX(q->noise_idx); 786 787 channels = q->nb_channels; 788 789 if (q->nb_channels <= 1 || sb < 12) 790 joined_stereo = 0; 791 else if (sb >= 24) 792 joined_stereo = 1; 793 else 794 joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0; 795 796 if (joined_stereo) { 797 if (BITS_LEFT(length,gb) >= 16) 798 for (j = 0; j < 16; j++) 799 sign_bits[j] = get_bits1 (gb); 800 801 for (j = 0; j < 64; j++) 802 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j]) 803 q->coding_method[0][sb][j] = q->coding_method[1][sb][j]; 804 805 fix_coding_method_array(sb, q->nb_channels, q->coding_method); 806 channels = 1; 807 } 808 809 for (ch = 0; ch < channels; ch++) { 810 zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0; 811 type34_predictor = 0.0; 812 type34_first = 1; 813 814 for (j = 0; j < 128; ) { 815 switch (q->coding_method[ch][sb][j / 2]) { 816 case 8: 817 if (BITS_LEFT(length,gb) >= 10) { 818 if (zero_encoding) { 819 for (k = 0; k < 5; k++) { 820 if ((j + 2 * k) >= 128) 821 break; 822 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0; 823 } 824 } else { 825 n = get_bits(gb, 8); 826 for (k = 0; k < 5; k++) 827 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; 828 } 829 for (k = 0; k < 5; k++) 830 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx); 831 } else { 832 for (k = 0; k < 10; k++) 833 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 834 } 835 run = 10; 836 break; 837 838 case 10: 839 if (BITS_LEFT(length,gb) >= 1) { 840 float f = 0.81; 841 842 if (get_bits1(gb)) 843 f = -f; 844 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0; 845 samples[0] = f; 846 } else { 847 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 848 } 849 run = 1; 850 break; 851 852 case 16: 853 if (BITS_LEFT(length,gb) >= 10) { 854 if (zero_encoding) { 855 for (k = 0; k < 5; k++) { 856 if ((j + k) >= 128) 857 break; 858 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)]; 859 } 860 } else { 861 n = get_bits (gb, 8); 862 for (k = 0; k < 5; k++) 863 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; 864 } 865 } else { 866 for (k = 0; k < 5; k++) 867 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 868 } 869 run = 5; 870 break; 871 872 case 24: 873 if (BITS_LEFT(length,gb) >= 7) { 874 n = get_bits(gb, 7); 875 for (k = 0; k < 3; k++) 876 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5; 877 } else { 878 for (k = 0; k < 3; k++) 879 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 880 } 881 run = 3; 882 break; 883 884 case 30: 885 if (BITS_LEFT(length,gb) >= 4) 886 samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)]; 887 else 888 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 889 890 run = 1; 891 break; 892 893 case 34: 894 if (BITS_LEFT(length,gb) >= 7) { 895 if (type34_first) { 896 type34_div = (float)(1 << get_bits(gb, 2)); 897 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0; 898 type34_predictor = samples[0]; 899 type34_first = 0; 900 } else { 901 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor; 902 type34_predictor = samples[0]; 903 } 904 } else { 905 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 906 } 907 run = 1; 908 break; 909 910 default: 911 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 912 run = 1; 913 break; 914 } 915 916 if (joined_stereo) { 917 float tmp[10][MPA_MAX_CHANNELS]; 918 919 for (k = 0; k < run; k++) { 920 tmp[k][0] = samples[k]; 921 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k]; 922 } 923 for (chs = 0; chs < q->nb_channels; chs++) 924 for (k = 0; k < run; k++) 925 if ((j + k) < 128) 926 q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5); 927 } else { 928 for (k = 0; k < run; k++) 929 if ((j + k) < 128) 930 q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5); 931 } 932 933 j += run; 934 } // j loop 935 } // channel loop 936 } // subband loop 937} 938 939 940/** 941 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]). 942 * This is similar to process_subpacket_9, but for a single channel and for element [0] 943 * same VLC tables as process_subpacket_9 are used. 944 * 945 * @param q context 946 * @param quantized_coeffs pointer to quantized_coeffs[ch][0] 947 * @param gb bitreader context 948 * @param length packet length in bits 949 */ 950static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length) 951{ 952 int i, k, run, level, diff; 953 954 if (BITS_LEFT(length,gb) < 16) 955 return; 956 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); 957 958 quantized_coeffs[0] = level; 959 960 for (i = 0; i < 7; ) { 961 if (BITS_LEFT(length,gb) < 16) 962 break; 963 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; 964 965 if (BITS_LEFT(length,gb) < 16) 966 break; 967 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); 968 969 for (k = 1; k <= run; k++) 970 quantized_coeffs[i + k] = (level + ((k * diff) / run)); 971 972 level += diff; 973 i += run; 974 } 975} 976 977 978/** 979 * Related to synthesis filter, process data from packet 10 980 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0 981 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10 982 * 983 * @param q context 984 * @param gb bitreader context 985 * @param length packet length in bits 986 */ 987static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length) 988{ 989 int sb, j, k, n, ch; 990 991 for (ch = 0; ch < q->nb_channels; ch++) { 992 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length); 993 994 if (BITS_LEFT(length,gb) < 16) { 995 memset(q->quantized_coeffs[ch][0], 0, 8); 996 break; 997 } 998 } 999 1000 n = q->sub_sampling + 1; 1001 1002 for (sb = 0; sb < n; sb++) 1003 for (ch = 0; ch < q->nb_channels; ch++) 1004 for (j = 0; j < 8; j++) { 1005 if (BITS_LEFT(length,gb) < 1) 1006 break; 1007 if (get_bits1(gb)) { 1008 for (k=0; k < 8; k++) { 1009 if (BITS_LEFT(length,gb) < 16) 1010 break; 1011 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2); 1012 } 1013 } else { 1014 for (k=0; k < 8; k++) 1015 q->tone_level_idx_hi1[ch][sb][j][k] = 0; 1016 } 1017 } 1018 1019 n = QDM2_SB_USED(q->sub_sampling) - 4; 1020 1021 for (sb = 0; sb < n; sb++) 1022 for (ch = 0; ch < q->nb_channels; ch++) { 1023 if (BITS_LEFT(length,gb) < 16) 1024 break; 1025 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2); 1026 if (sb > 19) 1027 q->tone_level_idx_hi2[ch][sb] -= 16; 1028 else 1029 for (j = 0; j < 8; j++) 1030 q->tone_level_idx_mid[ch][sb][j] = -16; 1031 } 1032 1033 n = QDM2_SB_USED(q->sub_sampling) - 5; 1034 1035 for (sb = 0; sb < n; sb++) 1036 for (ch = 0; ch < q->nb_channels; ch++) 1037 for (j = 0; j < 8; j++) { 1038 if (BITS_LEFT(length,gb) < 16) 1039 break; 1040 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32; 1041 } 1042} 1043 1044/** 1045 * Process subpacket 9, init quantized_coeffs with data from it 1046 * 1047 * @param q context 1048 * @param node pointer to node with packet 1049 */ 1050static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node) 1051{ 1052 GetBitContext gb; 1053 int i, j, k, n, ch, run, level, diff; 1054 1055 init_get_bits(&gb, node->packet->data, node->packet->size*8); 1056 1057 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function 1058 1059 for (i = 1; i < n; i++) 1060 for (ch=0; ch < q->nb_channels; ch++) { 1061 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2); 1062 q->quantized_coeffs[ch][i][0] = level; 1063 1064 for (j = 0; j < (8 - 1); ) { 1065 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1; 1066 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2); 1067 1068 for (k = 1; k <= run; k++) 1069 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run)); 1070 1071 level += diff; 1072 j += run; 1073 } 1074 } 1075 1076 for (ch = 0; ch < q->nb_channels; ch++) 1077 for (i = 0; i < 8; i++) 1078 q->quantized_coeffs[ch][0][i] = 0; 1079} 1080 1081 1082/** 1083 * Process subpacket 10 if not null, else 1084 * 1085 * @param q context 1086 * @param node pointer to node with packet 1087 * @param length packet length in bits 1088 */ 1089static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length) 1090{ 1091 GetBitContext gb; 1092 1093 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); 1094 1095 if (length != 0) { 1096 init_tone_level_dequantization(q, &gb, length); 1097 fill_tone_level_array(q, 1); 1098 } else { 1099 fill_tone_level_array(q, 0); 1100 } 1101} 1102 1103 1104/** 1105 * Process subpacket 11 1106 * 1107 * @param q context 1108 * @param node pointer to node with packet 1109 * @param length packet length in bit 1110 */ 1111static void process_subpacket_11 (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 if (length >= 32) { 1117 int c = get_bits (&gb, 13); 1118 1119 if (c > 3) 1120 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method, 1121 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select); 1122 } 1123 1124 synthfilt_build_sb_samples(q, &gb, length, 0, 8); 1125} 1126 1127 1128/** 1129 * Process subpacket 12 1130 * 1131 * @param q context 1132 * @param node pointer to node with packet 1133 * @param length packet length in bits 1134 */ 1135static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length) 1136{ 1137 GetBitContext gb; 1138 1139 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); 1140 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling)); 1141} 1142 1143/* 1144 * Process new subpackets for synthesis filter 1145 * 1146 * @param q context 1147 * @param list list with synthesis filter packets (list D) 1148 */ 1149static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list) 1150{ 1151 QDM2SubPNode *nodes[4]; 1152 1153 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9); 1154 if (nodes[0] != NULL) 1155 process_subpacket_9(q, nodes[0]); 1156 1157 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10); 1158 if (nodes[1] != NULL) 1159 process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3); 1160 else 1161 process_subpacket_10(q, NULL, 0); 1162 1163 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11); 1164 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL) 1165 process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3)); 1166 else 1167 process_subpacket_11(q, NULL, 0); 1168 1169 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12); 1170 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL) 1171 process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3)); 1172 else 1173 process_subpacket_12(q, NULL, 0); 1174} 1175 1176 1177/* 1178 * Decode superblock, fill packet lists. 1179 * 1180 * @param q context 1181 */ 1182static void qdm2_decode_super_block (QDM2Context *q) 1183{ 1184 GetBitContext gb; 1185 QDM2SubPacket header, *packet; 1186 int i, packet_bytes, sub_packet_size, sub_packets_D; 1187 unsigned int next_index = 0; 1188 1189 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1)); 1190 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid)); 1191 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2)); 1192 1193 q->sub_packets_B = 0; 1194 sub_packets_D = 0; 1195 1196 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8] 1197 1198 init_get_bits(&gb, q->compressed_data, q->compressed_size*8); 1199 qdm2_decode_sub_packet_header(&gb, &header); 1200 1201 if (header.type < 2 || header.type >= 8) { 1202 q->has_errors = 1; 1203 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n"); 1204 return; 1205 } 1206 1207 q->superblocktype_2_3 = (header.type == 2 || header.type == 3); 1208 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8); 1209 1210 init_get_bits(&gb, header.data, header.size*8); 1211 1212 if (header.type == 2 || header.type == 4 || header.type == 5) { 1213 int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8); 1214 1215 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); 1216 1217 if (csum != 0) { 1218 q->has_errors = 1; 1219 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n"); 1220 return; 1221 } 1222 } 1223 1224 q->sub_packet_list_B[0].packet = NULL; 1225 q->sub_packet_list_D[0].packet = NULL; 1226 1227 for (i = 0; i < 6; i++) 1228 if (--q->fft_level_exp[i] < 0) 1229 q->fft_level_exp[i] = 0; 1230 1231 for (i = 0; packet_bytes > 0; i++) { 1232 int j; 1233 1234 q->sub_packet_list_A[i].next = NULL; 1235 1236 if (i > 0) { 1237 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i]; 1238 1239 /* seek to next block */ 1240 init_get_bits(&gb, header.data, header.size*8); 1241 skip_bits(&gb, next_index*8); 1242 1243 if (next_index >= header.size) 1244 break; 1245 } 1246 1247 /* decode subpacket */ 1248 packet = &q->sub_packets[i]; 1249 qdm2_decode_sub_packet_header(&gb, packet); 1250 next_index = packet->size + get_bits_count(&gb) / 8; 1251 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2; 1252 1253 if (packet->type == 0) 1254 break; 1255 1256 if (sub_packet_size > packet_bytes) { 1257 if (packet->type != 10 && packet->type != 11 && packet->type != 12) 1258 break; 1259 packet->size += packet_bytes - sub_packet_size; 1260 } 1261 1262 packet_bytes -= sub_packet_size; 1263 1264 /* add subpacket to 'all subpackets' list */ 1265 q->sub_packet_list_A[i].packet = packet; 1266 1267 /* add subpacket to related list */ 1268 if (packet->type == 8) { 1269 SAMPLES_NEEDED_2("packet type 8"); 1270 return; 1271 } else if (packet->type >= 9 && packet->type <= 12) { 1272 /* packets for MPEG Audio like Synthesis Filter */ 1273 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet); 1274 } else if (packet->type == 13) { 1275 for (j = 0; j < 6; j++) 1276 q->fft_level_exp[j] = get_bits(&gb, 6); 1277 } else if (packet->type == 14) { 1278 for (j = 0; j < 6; j++) 1279 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2); 1280 } else if (packet->type == 15) { 1281 SAMPLES_NEEDED_2("packet type 15") 1282 return; 1283 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) { 1284 /* packets for FFT */ 1285 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet); 1286 } 1287 } // Packet bytes loop 1288 1289/* **************************************************************** */ 1290 if (q->sub_packet_list_D[0].packet != NULL) { 1291 process_synthesis_subpackets(q, q->sub_packet_list_D); 1292 q->do_synth_filter = 1; 1293 } else if (q->do_synth_filter) { 1294 process_subpacket_10(q, NULL, 0); 1295 process_subpacket_11(q, NULL, 0); 1296 process_subpacket_12(q, NULL, 0); 1297 } 1298/* **************************************************************** */ 1299} 1300 1301 1302static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet, 1303 int offset, int duration, int channel, 1304 int exp, int phase) 1305{ 1306 if (q->fft_coefs_min_index[duration] < 0) 1307 q->fft_coefs_min_index[duration] = q->fft_coefs_index; 1308 1309 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); 1310 q->fft_coefs[q->fft_coefs_index].channel = channel; 1311 q->fft_coefs[q->fft_coefs_index].offset = offset; 1312 q->fft_coefs[q->fft_coefs_index].exp = exp; 1313 q->fft_coefs[q->fft_coefs_index].phase = phase; 1314 q->fft_coefs_index++; 1315} 1316 1317 1318static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b) 1319{ 1320 int channel, stereo, phase, exp; 1321 int local_int_4, local_int_8, stereo_phase, local_int_10; 1322 int local_int_14, stereo_exp, local_int_20, local_int_28; 1323 int n, offset; 1324 1325 local_int_4 = 0; 1326 local_int_28 = 0; 1327 local_int_20 = 2; 1328 local_int_8 = (4 - duration); 1329 local_int_10 = 1 << (q->group_order - duration - 1); 1330 offset = 1; 1331 1332 while (1) { 1333 if (q->superblocktype_2_3) { 1334 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { 1335 offset = 1; 1336 if (n == 0) { 1337 local_int_4 += local_int_10; 1338 local_int_28 += (1 << local_int_8); 1339 } else { 1340 local_int_4 += 8*local_int_10; 1341 local_int_28 += (8 << local_int_8); 1342 } 1343 } 1344 offset += (n - 2); 1345 } else { 1346 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2); 1347 while (offset >= (local_int_10 - 1)) { 1348 offset += (1 - (local_int_10 - 1)); 1349 local_int_4 += local_int_10; 1350 local_int_28 += (1 << local_int_8); 1351 } 1352 } 1353 1354 if (local_int_4 >= q->group_size) 1355 return; 1356 1357 local_int_14 = (offset >> local_int_8); 1358 1359 if (q->nb_channels > 1) { 1360 channel = get_bits1(gb); 1361 stereo = get_bits1(gb); 1362 } else { 1363 channel = 0; 1364 stereo = 0; 1365 } 1366 1367 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); 1368 exp += q->fft_level_exp[fft_level_index_table[local_int_14]]; 1369 exp = (exp < 0) ? 0 : exp; 1370 1371 phase = get_bits(gb, 3); 1372 stereo_exp = 0; 1373 stereo_phase = 0; 1374 1375 if (stereo) { 1376 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1)); 1377 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1)); 1378 if (stereo_phase < 0) 1379 stereo_phase += 8; 1380 } 1381 1382 if (q->frequency_range > (local_int_14 + 1)) { 1383 int sub_packet = (local_int_20 + local_int_28); 1384 1385 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase); 1386 if (stereo) 1387 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase); 1388 } 1389 1390 offset++; 1391 } 1392} 1393 1394 1395static void qdm2_decode_fft_packets (QDM2Context *q) 1396{ 1397 int i, j, min, max, value, type, unknown_flag; 1398 GetBitContext gb; 1399 1400 if (q->sub_packet_list_B[0].packet == NULL) 1401 return; 1402 1403 /* reset minimum indexes for FFT coefficients */ 1404 q->fft_coefs_index = 0; 1405 for (i=0; i < 5; i++) 1406 q->fft_coefs_min_index[i] = -1; 1407 1408 /* process subpackets ordered by type, largest type first */ 1409 for (i = 0, max = 256; i < q->sub_packets_B; i++) { 1410 QDM2SubPacket *packet= NULL; 1411 1412 /* find subpacket with largest type less than max */ 1413 for (j = 0, min = 0; j < q->sub_packets_B; j++) { 1414 value = q->sub_packet_list_B[j].packet->type; 1415 if (value > min && value < max) { 1416 min = value; 1417 packet = q->sub_packet_list_B[j].packet; 1418 } 1419 } 1420 1421 max = min; 1422 1423 /* check for errors (?) */ 1424 if (!packet) 1425 return; 1426 1427 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16])) 1428 return; 1429 1430 /* decode FFT tones */ 1431 init_get_bits (&gb, packet->data, packet->size*8); 1432 1433 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) 1434 unknown_flag = 1; 1435 else 1436 unknown_flag = 0; 1437 1438 type = packet->type; 1439 1440 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) { 1441 int duration = q->sub_sampling + 5 - (type & 15); 1442 1443 if (duration >= 0 && duration < 4) 1444 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); 1445 } else if (type == 31) { 1446 for (j=0; j < 4; j++) 1447 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); 1448 } else if (type == 46) { 1449 for (j=0; j < 6; j++) 1450 q->fft_level_exp[j] = get_bits(&gb, 6); 1451 for (j=0; j < 4; j++) 1452 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); 1453 } 1454 } // Loop on B packets 1455 1456 /* calculate maximum indexes for FFT coefficients */ 1457 for (i = 0, j = -1; i < 5; i++) 1458 if (q->fft_coefs_min_index[i] >= 0) { 1459 if (j >= 0) 1460 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i]; 1461 j = i; 1462 } 1463 if (j >= 0) 1464 q->fft_coefs_max_index[j] = q->fft_coefs_index; 1465} 1466 1467 1468static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone) 1469{ 1470 float level, f[6]; 1471 int i; 1472 QDM2Complex c; 1473 const double iscale = 2.0*M_PI / 512.0; 1474 1475 tone->phase += tone->phase_shift; 1476 1477 /* calculate current level (maximum amplitude) of tone */ 1478 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level; 1479 c.im = level * sin(tone->phase*iscale); 1480 c.re = level * cos(tone->phase*iscale); 1481 1482 /* generate FFT coefficients for tone */ 1483 if (tone->duration >= 3 || tone->cutoff >= 3) { 1484 tone->complex[0].im += c.im; 1485 tone->complex[0].re += c.re; 1486 tone->complex[1].im -= c.im; 1487 tone->complex[1].re -= c.re; 1488 } else { 1489 f[1] = -tone->table[4]; 1490 f[0] = tone->table[3] - tone->table[0]; 1491 f[2] = 1.0 - tone->table[2] - tone->table[3]; 1492 f[3] = tone->table[1] + tone->table[4] - 1.0; 1493 f[4] = tone->table[0] - tone->table[1]; 1494 f[5] = tone->table[2]; 1495 for (i = 0; i < 2; i++) { 1496 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i]; 1497 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]); 1498 } 1499 for (i = 0; i < 4; i++) { 1500 tone->complex[i].re += c.re * f[i+2]; 1501 tone->complex[i].im += c.im * f[i+2]; 1502 } 1503 } 1504 1505 /* copy the tone if it has not yet died out */ 1506 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) { 1507 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone)); 1508 q->fft_tone_end = (q->fft_tone_end + 1) % 1000; 1509 } 1510} 1511 1512 1513static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) 1514{ 1515 int i, j, ch; 1516 const double iscale = 0.25 * M_PI; 1517 1518 for (ch = 0; ch < q->channels; ch++) { 1519 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex)); 1520 } 1521 1522 1523 /* apply FFT tones with duration 4 (1 FFT period) */ 1524 if (q->fft_coefs_min_index[4] >= 0) 1525 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) { 1526 float level; 1527 QDM2Complex c; 1528 1529 if (q->fft_coefs[i].sub_packet != sub_packet) 1530 break; 1531 1532 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel; 1533 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]; 1534 1535 c.re = level * cos(q->fft_coefs[i].phase * iscale); 1536 c.im = level * sin(q->fft_coefs[i].phase * iscale); 1537 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re; 1538 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im; 1539 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re; 1540 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im; 1541 } 1542 1543 /* generate existing FFT tones */ 1544 for (i = q->fft_tone_end; i != q->fft_tone_start; ) { 1545 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]); 1546 q->fft_tone_start = (q->fft_tone_start + 1) % 1000; 1547 } 1548 1549 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */ 1550 for (i = 0; i < 4; i++) 1551 if (q->fft_coefs_min_index[i] >= 0) { 1552 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) { 1553 int offset, four_i; 1554 FFTTone tone; 1555 1556 if (q->fft_coefs[j].sub_packet != sub_packet) 1557 break; 1558 1559 four_i = (4 - i); 1560 offset = q->fft_coefs[j].offset >> four_i; 1561 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel; 1562 1563 if (offset < q->frequency_range) { 1564 if (offset < 2) 1565 tone.cutoff = offset; 1566 else 1567 tone.cutoff = (offset >= 60) ? 3 : 2; 1568 1569 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]; 1570 tone.complex = &q->fft.complex[ch][offset]; 1571 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)]; 1572 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128; 1573 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i); 1574 tone.duration = i; 1575 tone.time_index = 0; 1576 1577 qdm2_fft_generate_tone(q, &tone); 1578 } 1579 } 1580 q->fft_coefs_min_index[i] = j; 1581 } 1582} 1583 1584 1585static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) 1586{ 1587 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f; 1588 int i; 1589 q->fft.complex[channel][0].re *= 2.0f; 1590 q->fft.complex[channel][0].im = 0.0f; 1591 ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]); 1592 /* add samples to output buffer */ 1593 for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++) 1594 q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain; 1595} 1596 1597 1598/** 1599 * @param q context 1600 * @param index subpacket number 1601 */ 1602static void qdm2_synthesis_filter (QDM2Context *q, int index) 1603{ 1604 OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; 1605 int i, k, ch, sb_used, sub_sampling, dither_state = 0; 1606 1607 /* copy sb_samples */ 1608 sb_used = QDM2_SB_USED(q->sub_sampling); 1609 1610 for (ch = 0; ch < q->channels; ch++) 1611 for (i = 0; i < 8; i++) 1612 for (k=sb_used; k < SBLIMIT; k++) 1613 q->sb_samples[ch][(8 * index) + i][k] = 0; 1614 1615 for (ch = 0; ch < q->nb_channels; ch++) { 1616 OUT_INT *samples_ptr = samples + ch; 1617 1618 for (i = 0; i < 8; i++) { 1619 ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]), 1620 ff_mpa_synth_window, &dither_state, 1621 samples_ptr, q->nb_channels, 1622 q->sb_samples[ch][(8 * index) + i]); 1623 samples_ptr += 32 * q->nb_channels; 1624 } 1625 } 1626 1627 /* add samples to output buffer */ 1628 sub_sampling = (4 >> q->sub_sampling); 1629 1630 for (ch = 0; ch < q->channels; ch++) 1631 for (i = 0; i < q->frame_size; i++) 1632 q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16)); 1633} 1634 1635 1636/** 1637 * Init static data (does not depend on specific file) 1638 * 1639 * @param q context 1640 */ 1641static av_cold void qdm2_init(QDM2Context *q) { 1642 static int initialized = 0; 1643 1644 if (initialized != 0) 1645 return; 1646 initialized = 1; 1647 1648 qdm2_init_vlc(); 1649 ff_mpa_synth_init(ff_mpa_synth_window); 1650 softclip_table_init(); 1651 rnd_table_init(); 1652 init_noise_samples(); 1653 1654 av_log(NULL, AV_LOG_DEBUG, "init done\n"); 1655} 1656 1657 1658#if 0 1659static void dump_context(QDM2Context *q) 1660{ 1661 int i; 1662#define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b); 1663 PRINT("compressed_data",q->compressed_data); 1664 PRINT("compressed_size",q->compressed_size); 1665 PRINT("frame_size",q->frame_size); 1666 PRINT("checksum_size",q->checksum_size); 1667 PRINT("channels",q->channels); 1668 PRINT("nb_channels",q->nb_channels); 1669 PRINT("fft_frame_size",q->fft_frame_size); 1670 PRINT("fft_size",q->fft_size); 1671 PRINT("sub_sampling",q->sub_sampling); 1672 PRINT("fft_order",q->fft_order); 1673 PRINT("group_order",q->group_order); 1674 PRINT("group_size",q->group_size); 1675 PRINT("sub_packet",q->sub_packet); 1676 PRINT("frequency_range",q->frequency_range); 1677 PRINT("has_errors",q->has_errors); 1678 PRINT("fft_tone_end",q->fft_tone_end); 1679 PRINT("fft_tone_start",q->fft_tone_start); 1680 PRINT("fft_coefs_index",q->fft_coefs_index); 1681 PRINT("coeff_per_sb_select",q->coeff_per_sb_select); 1682 PRINT("cm_table_select",q->cm_table_select); 1683 PRINT("noise_idx",q->noise_idx); 1684 1685 for (i = q->fft_tone_start; i < q->fft_tone_end; i++) 1686 { 1687 FFTTone *t = &q->fft_tones[i]; 1688 1689 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i); 1690 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level); 1691// PRINT(" level", t->level); 1692 PRINT(" phase", t->phase); 1693 PRINT(" phase_shift", t->phase_shift); 1694 PRINT(" duration", t->duration); 1695 PRINT(" samples_im", t->samples_im); 1696 PRINT(" samples_re", t->samples_re); 1697 PRINT(" table", t->table); 1698 } 1699 1700} 1701#endif 1702 1703 1704/** 1705 * Init parameters from codec extradata 1706 */ 1707static av_cold int qdm2_decode_init(AVCodecContext *avctx) 1708{ 1709 QDM2Context *s = avctx->priv_data; 1710 uint8_t *extradata; 1711 int extradata_size; 1712 int tmp_val, tmp, size; 1713 1714 /* extradata parsing 1715 1716 Structure: 1717 wave { 1718 frma (QDM2) 1719 QDCA 1720 QDCP 1721 } 1722 1723 32 size (including this field) 1724 32 tag (=frma) 1725 32 type (=QDM2 or QDMC) 1726 1727 32 size (including this field, in bytes) 1728 32 tag (=QDCA) // maybe mandatory parameters 1729 32 unknown (=1) 1730 32 channels (=2) 1731 32 samplerate (=44100) 1732 32 bitrate (=96000) 1733 32 block size (=4096) 1734 32 frame size (=256) (for one channel) 1735 32 packet size (=1300) 1736 1737 32 size (including this field, in bytes) 1738 32 tag (=QDCP) // maybe some tuneable parameters 1739 32 float1 (=1.0) 1740 32 zero ? 1741 32 float2 (=1.0) 1742 32 float3 (=1.0) 1743 32 unknown (27) 1744 32 unknown (8) 1745 32 zero ? 1746 */ 1747 1748 if (!avctx->extradata || (avctx->extradata_size < 48)) { 1749 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n"); 1750 return -1; 1751 } 1752 1753 extradata = avctx->extradata; 1754 extradata_size = avctx->extradata_size; 1755 1756 while (extradata_size > 7) { 1757 if (!memcmp(extradata, "frmaQDM", 7)) 1758 break; 1759 extradata++; 1760 extradata_size--; 1761 } 1762 1763 if (extradata_size < 12) { 1764 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n", 1765 extradata_size); 1766 return -1; 1767 } 1768 1769 if (memcmp(extradata, "frmaQDM", 7)) { 1770 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n"); 1771 return -1; 1772 } 1773 1774 if (extradata[7] == 'C') { 1775// s->is_qdmc = 1; 1776 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n"); 1777 return -1; 1778 } 1779 1780 extradata += 8; 1781 extradata_size -= 8; 1782 1783 size = AV_RB32(extradata); 1784 1785 if(size > extradata_size){ 1786 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n", 1787 extradata_size, size); 1788 return -1; 1789 } 1790 1791 extradata += 4; 1792 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size); 1793 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) { 1794 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n"); 1795 return -1; 1796 } 1797 1798 extradata += 8; 1799 1800 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata); 1801 extradata += 4; 1802 1803 avctx->sample_rate = AV_RB32(extradata); 1804 extradata += 4; 1805 1806 avctx->bit_rate = AV_RB32(extradata); 1807 extradata += 4; 1808 1809 s->group_size = AV_RB32(extradata); 1810 extradata += 4; 1811 1812 s->fft_size = AV_RB32(extradata); 1813 extradata += 4; 1814 1815 s->checksum_size = AV_RB32(extradata); 1816 1817 s->fft_order = av_log2(s->fft_size) + 1; 1818 s->fft_frame_size = 2 * s->fft_size; // complex has two floats 1819 1820 // something like max decodable tones 1821 s->group_order = av_log2(s->group_size) + 1; 1822 s->frame_size = s->group_size / 16; // 16 iterations per super block 1823 1824 s->sub_sampling = s->fft_order - 7; 1825 s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); 1826 1827 switch ((s->sub_sampling * 2 + s->channels - 1)) { 1828 case 0: tmp = 40; break; 1829 case 1: tmp = 48; break; 1830 case 2: tmp = 56; break; 1831 case 3: tmp = 72; break; 1832 case 4: tmp = 80; break; 1833 case 5: tmp = 100;break; 1834 default: tmp=s->sub_sampling; break; 1835 } 1836 tmp_val = 0; 1837 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1; 1838 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2; 1839 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3; 1840 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4; 1841 s->cm_table_select = tmp_val; 1842 1843 if (s->sub_sampling == 0) 1844 tmp = 7999; 1845 else 1846 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000; 1847 /* 1848 0: 7999 -> 0 1849 1: 20000 -> 2 1850 2: 28000 -> 2 1851 */ 1852 if (tmp < 8000) 1853 s->coeff_per_sb_select = 0; 1854 else if (tmp <= 16000) 1855 s->coeff_per_sb_select = 1; 1856 else 1857 s->coeff_per_sb_select = 2; 1858 1859 // Fail on unknown fft order 1860 if ((s->fft_order < 7) || (s->fft_order > 9)) { 1861 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order); 1862 return -1; 1863 } 1864 1865 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R); 1866 1867 qdm2_init(s); 1868 1869 avctx->sample_fmt = SAMPLE_FMT_S16; 1870 1871// dump_context(s); 1872 return 0; 1873} 1874 1875 1876static av_cold int qdm2_decode_close(AVCodecContext *avctx) 1877{ 1878 QDM2Context *s = avctx->priv_data; 1879 1880 ff_rdft_end(&s->rdft_ctx); 1881 1882 return 0; 1883} 1884 1885 1886static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) 1887{ 1888 int ch, i; 1889 const int frame_size = (q->frame_size * q->channels); 1890 1891 /* select input buffer */ 1892 q->compressed_data = in; 1893 q->compressed_size = q->checksum_size; 1894 1895// dump_context(q); 1896 1897 /* copy old block, clear new block of output samples */ 1898 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float)); 1899 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float)); 1900 1901 /* decode block of QDM2 compressed data */ 1902 if (q->sub_packet == 0) { 1903 q->has_errors = 0; // zero it for a new super block 1904 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n"); 1905 qdm2_decode_super_block(q); 1906 } 1907 1908 /* parse subpackets */ 1909 if (!q->has_errors) { 1910 if (q->sub_packet == 2) 1911 qdm2_decode_fft_packets(q); 1912 1913 qdm2_fft_tone_synthesizer(q, q->sub_packet); 1914 } 1915 1916 /* sound synthesis stage 1 (FFT) */ 1917 for (ch = 0; ch < q->channels; ch++) { 1918 qdm2_calculate_fft(q, ch, q->sub_packet); 1919 1920 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) { 1921 SAMPLES_NEEDED_2("has errors, and C list is not empty") 1922 return; 1923 } 1924 } 1925 1926 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */ 1927 if (!q->has_errors && q->do_synth_filter) 1928 qdm2_synthesis_filter(q, q->sub_packet); 1929 1930 q->sub_packet = (q->sub_packet + 1) % 16; 1931 1932 /* clip and convert output float[] to 16bit signed samples */ 1933 for (i = 0; i < frame_size; i++) { 1934 int value = (int)q->output_buffer[i]; 1935 1936 if (value > SOFTCLIP_THRESHOLD) 1937 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD]; 1938 else if (value < -SOFTCLIP_THRESHOLD) 1939 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD]; 1940 1941 out[i] = value; 1942 } 1943} 1944 1945 1946static int qdm2_decode_frame(AVCodecContext *avctx, 1947 void *data, int *data_size, 1948 AVPacket *avpkt) 1949{ 1950 const uint8_t *buf = avpkt->data; 1951 int buf_size = avpkt->size; 1952 QDM2Context *s = avctx->priv_data; 1953 1954 if(!buf) 1955 return 0; 1956 if(buf_size < s->checksum_size) 1957 return -1; 1958 1959 *data_size = s->channels * s->frame_size * sizeof(int16_t); 1960 1961 av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n", 1962 buf_size, buf, s->checksum_size, data, *data_size); 1963 1964 qdm2_decode(s, buf, data); 1965 1966 // reading only when next superblock found 1967 if (s->sub_packet == 0) { 1968 return s->checksum_size; 1969 } 1970 1971 return 0; 1972} 1973 1974AVCodec qdm2_decoder = 1975{ 1976 .name = "qdm2", 1977 .type = AVMEDIA_TYPE_AUDIO, 1978 .id = CODEC_ID_QDM2, 1979 .priv_data_size = sizeof(QDM2Context), 1980 .init = qdm2_decode_init, 1981 .close = qdm2_decode_close, 1982 .decode = qdm2_decode_frame, 1983 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), 1984}; 1985