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 Libav. 9 * 10 * Libav 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 * Libav 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 Libav; if not, write to the Free Software 22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 23 */ 24 25/** 26 * @file 27 * QDM2 decoder 28 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni 29 * 30 * The decoder is not perfect yet, there are still some distortions 31 * especially on files encoded with 16 or 8 subbands. 32 */ 33 34#include <math.h> 35#include <stddef.h> 36#include <stdio.h> 37 38#define BITSTREAM_READER_LE 39#include "avcodec.h" 40#include "get_bits.h" 41#include "dsputil.h" 42#include "rdft.h" 43#include "mpegaudiodsp.h" 44#include "mpegaudio.h" 45 46#include "qdm2data.h" 47#include "qdm2_tablegen.h" 48 49#undef NDEBUG 50#include <assert.h> 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#define QDM2_MAX_FRAME_SIZE 512 81 82typedef int8_t sb_int8_array[2][30][64]; 83 84/** 85 * Subpacket 86 */ 87typedef struct { 88 int type; ///< subpacket type 89 unsigned int size; ///< subpacket size 90 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy) 91} QDM2SubPacket; 92 93/** 94 * A node in the subpacket list 95 */ 96typedef struct QDM2SubPNode { 97 QDM2SubPacket *packet; ///< packet 98 struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node 99} QDM2SubPNode; 100 101typedef struct { 102 float re; 103 float im; 104} QDM2Complex; 105 106typedef struct { 107 float level; 108 QDM2Complex *complex; 109 const float *table; 110 int phase; 111 int phase_shift; 112 int duration; 113 short time_index; 114 short cutoff; 115} FFTTone; 116 117typedef struct { 118 int16_t sub_packet; 119 uint8_t channel; 120 int16_t offset; 121 int16_t exp; 122 uint8_t phase; 123} FFTCoefficient; 124 125typedef struct { 126 DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256]; 127} QDM2FFT; 128 129/** 130 * QDM2 decoder context 131 */ 132typedef struct { 133 AVFrame frame; 134 135 /// Parameters from codec header, do not change during playback 136 int nb_channels; ///< number of channels 137 int channels; ///< number of channels 138 int group_size; ///< size of frame group (16 frames per group) 139 int fft_size; ///< size of FFT, in complex numbers 140 int checksum_size; ///< size of data block, used also for checksum 141 142 /// Parameters built from header parameters, do not change during playback 143 int group_order; ///< order of frame group 144 int fft_order; ///< order of FFT (actually fftorder+1) 145 int fft_frame_size; ///< size of fft frame, in components (1 comples = re + im) 146 int frame_size; ///< size of data frame 147 int frequency_range; 148 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */ 149 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2 150 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4) 151 152 /// Packets and packet lists 153 QDM2SubPacket sub_packets[16]; ///< the packets themselves 154 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets 155 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list 156 int sub_packets_B; ///< number of packets on 'B' list 157 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors? 158 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets 159 160 /// FFT and tones 161 FFTTone fft_tones[1000]; 162 int fft_tone_start; 163 int fft_tone_end; 164 FFTCoefficient fft_coefs[1000]; 165 int fft_coefs_index; 166 int fft_coefs_min_index[5]; 167 int fft_coefs_max_index[5]; 168 int fft_level_exp[6]; 169 RDFTContext rdft_ctx; 170 QDM2FFT fft; 171 172 /// I/O data 173 const uint8_t *compressed_data; 174 int compressed_size; 175 float output_buffer[QDM2_MAX_FRAME_SIZE * 2]; 176 177 /// Synthesis filter 178 MPADSPContext mpadsp; 179 DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2]; 180 int synth_buf_offset[MPA_MAX_CHANNELS]; 181 DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT]; 182 DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; 183 184 /// Mixed temporary data used in decoding 185 float tone_level[MPA_MAX_CHANNELS][30][64]; 186 int8_t coding_method[MPA_MAX_CHANNELS][30][64]; 187 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8]; 188 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8]; 189 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8]; 190 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8]; 191 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26]; 192 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64]; 193 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]; 194 195 // Flags 196 int has_errors; ///< packet has errors 197 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type 198 int do_synth_filter; ///< used to perform or skip synthesis filter 199 200 int sub_packet; 201 int noise_idx; ///< index for dithering noise table 202} QDM2Context; 203 204 205static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE]; 206 207static VLC vlc_tab_level; 208static VLC vlc_tab_diff; 209static VLC vlc_tab_run; 210static VLC fft_level_exp_alt_vlc; 211static VLC fft_level_exp_vlc; 212static VLC fft_stereo_exp_vlc; 213static VLC fft_stereo_phase_vlc; 214static VLC vlc_tab_tone_level_idx_hi1; 215static VLC vlc_tab_tone_level_idx_mid; 216static VLC vlc_tab_tone_level_idx_hi2; 217static VLC vlc_tab_type30; 218static VLC vlc_tab_type34; 219static VLC vlc_tab_fft_tone_offset[5]; 220 221static const uint16_t qdm2_vlc_offs[] = { 222 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838, 223}; 224 225static av_cold void qdm2_init_vlc(void) 226{ 227 static int vlcs_initialized = 0; 228 static VLC_TYPE qdm2_table[3838][2]; 229 230 if (!vlcs_initialized) { 231 232 vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]]; 233 vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0]; 234 init_vlc (&vlc_tab_level, 8, 24, 235 vlc_tab_level_huffbits, 1, 1, 236 vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 237 238 vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]]; 239 vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1]; 240 init_vlc (&vlc_tab_diff, 8, 37, 241 vlc_tab_diff_huffbits, 1, 1, 242 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 243 244 vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]]; 245 vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2]; 246 init_vlc (&vlc_tab_run, 5, 6, 247 vlc_tab_run_huffbits, 1, 1, 248 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 249 250 fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]]; 251 fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3]; 252 init_vlc (&fft_level_exp_alt_vlc, 8, 28, 253 fft_level_exp_alt_huffbits, 1, 1, 254 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 255 256 257 fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]]; 258 fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4]; 259 init_vlc (&fft_level_exp_vlc, 8, 20, 260 fft_level_exp_huffbits, 1, 1, 261 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 262 263 fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]]; 264 fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5]; 265 init_vlc (&fft_stereo_exp_vlc, 6, 7, 266 fft_stereo_exp_huffbits, 1, 1, 267 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 268 269 fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]]; 270 fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6]; 271 init_vlc (&fft_stereo_phase_vlc, 6, 9, 272 fft_stereo_phase_huffbits, 1, 1, 273 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 274 275 vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]]; 276 vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7]; 277 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20, 278 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1, 279 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 280 281 vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]]; 282 vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8]; 283 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24, 284 vlc_tab_tone_level_idx_mid_huffbits, 1, 1, 285 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 286 287 vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]]; 288 vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9]; 289 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24, 290 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1, 291 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 292 293 vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]]; 294 vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10]; 295 init_vlc (&vlc_tab_type30, 6, 9, 296 vlc_tab_type30_huffbits, 1, 1, 297 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 298 299 vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]]; 300 vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11]; 301 init_vlc (&vlc_tab_type34, 5, 10, 302 vlc_tab_type34_huffbits, 1, 1, 303 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 304 305 vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]]; 306 vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12]; 307 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23, 308 vlc_tab_fft_tone_offset_0_huffbits, 1, 1, 309 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 310 311 vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]]; 312 vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13]; 313 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28, 314 vlc_tab_fft_tone_offset_1_huffbits, 1, 1, 315 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 316 317 vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]]; 318 vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14]; 319 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32, 320 vlc_tab_fft_tone_offset_2_huffbits, 1, 1, 321 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 322 323 vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]]; 324 vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15]; 325 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35, 326 vlc_tab_fft_tone_offset_3_huffbits, 1, 1, 327 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 328 329 vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]]; 330 vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16]; 331 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38, 332 vlc_tab_fft_tone_offset_4_huffbits, 1, 1, 333 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 334 335 vlcs_initialized=1; 336 } 337} 338 339static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth) 340{ 341 int value; 342 343 value = get_vlc2(gb, vlc->table, vlc->bits, depth); 344 345 /* stage-2, 3 bits exponent escape sequence */ 346 if (value-- == 0) 347 value = get_bits (gb, get_bits (gb, 3) + 1); 348 349 /* stage-3, optional */ 350 if (flag) { 351 int tmp = vlc_stage3_values[value]; 352 353 if ((value & ~3) > 0) 354 tmp += get_bits (gb, (value >> 2)); 355 value = tmp; 356 } 357 358 return value; 359} 360 361 362static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth) 363{ 364 int value = qdm2_get_vlc (gb, vlc, 0, depth); 365 366 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1); 367} 368 369 370/** 371 * QDM2 checksum 372 * 373 * @param data pointer to data to be checksum'ed 374 * @param length data length 375 * @param value checksum value 376 * 377 * @return 0 if checksum is OK 378 */ 379static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) { 380 int i; 381 382 for (i=0; i < length; i++) 383 value -= data[i]; 384 385 return (uint16_t)(value & 0xffff); 386} 387 388 389/** 390 * Fill a QDM2SubPacket structure with packet type, size, and data pointer. 391 * 392 * @param gb bitreader context 393 * @param sub_packet packet under analysis 394 */ 395static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet) 396{ 397 sub_packet->type = get_bits (gb, 8); 398 399 if (sub_packet->type == 0) { 400 sub_packet->size = 0; 401 sub_packet->data = NULL; 402 } else { 403 sub_packet->size = get_bits (gb, 8); 404 405 if (sub_packet->type & 0x80) { 406 sub_packet->size <<= 8; 407 sub_packet->size |= get_bits (gb, 8); 408 sub_packet->type &= 0x7f; 409 } 410 411 if (sub_packet->type == 0x7f) 412 sub_packet->type |= (get_bits (gb, 8) << 8); 413 414 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data 415 } 416 417 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n", 418 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8); 419} 420 421 422/** 423 * Return node pointer to first packet of requested type in list. 424 * 425 * @param list list of subpackets to be scanned 426 * @param type type of searched subpacket 427 * @return node pointer for subpacket if found, else NULL 428 */ 429static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type) 430{ 431 while (list != NULL && list->packet != NULL) { 432 if (list->packet->type == type) 433 return list; 434 list = list->next; 435 } 436 return NULL; 437} 438 439 440/** 441 * Replace 8 elements with their average value. 442 * Called by qdm2_decode_superblock before starting subblock decoding. 443 * 444 * @param q context 445 */ 446static void average_quantized_coeffs (QDM2Context *q) 447{ 448 int i, j, n, ch, sum; 449 450 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; 451 452 for (ch = 0; ch < q->nb_channels; ch++) 453 for (i = 0; i < n; i++) { 454 sum = 0; 455 456 for (j = 0; j < 8; j++) 457 sum += q->quantized_coeffs[ch][i][j]; 458 459 sum /= 8; 460 if (sum > 0) 461 sum--; 462 463 for (j=0; j < 8; j++) 464 q->quantized_coeffs[ch][i][j] = sum; 465 } 466} 467 468 469/** 470 * Build subband samples with noise weighted by q->tone_level. 471 * Called by synthfilt_build_sb_samples. 472 * 473 * @param q context 474 * @param sb subband index 475 */ 476static void build_sb_samples_from_noise (QDM2Context *q, int sb) 477{ 478 int ch, j; 479 480 FIX_NOISE_IDX(q->noise_idx); 481 482 if (!q->nb_channels) 483 return; 484 485 for (ch = 0; ch < q->nb_channels; ch++) 486 for (j = 0; j < 64; j++) { 487 q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j]; 488 q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j]; 489 } 490} 491 492 493/** 494 * Called while processing data from subpackets 11 and 12. 495 * Used after making changes to coding_method array. 496 * 497 * @param sb subband index 498 * @param channels number of channels 499 * @param coding_method q->coding_method[0][0][0] 500 */ 501static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method) 502{ 503 int j,k; 504 int ch; 505 int run, case_val; 506 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}; 507 508 for (ch = 0; ch < channels; ch++) { 509 for (j = 0; j < 64; ) { 510 if((coding_method[ch][sb][j] - 8) > 22) { 511 run = 1; 512 case_val = 8; 513 } else { 514 switch (switchtable[coding_method[ch][sb][j]-8]) { 515 case 0: run = 10; case_val = 10; break; 516 case 1: run = 1; case_val = 16; break; 517 case 2: run = 5; case_val = 24; break; 518 case 3: run = 3; case_val = 30; break; 519 case 4: run = 1; case_val = 30; break; 520 case 5: run = 1; case_val = 8; break; 521 default: run = 1; case_val = 8; break; 522 } 523 } 524 for (k = 0; k < run; k++) 525 if (j + k < 128) 526 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) 527 if (k > 0) { 528 SAMPLES_NEEDED 529 //not debugged, almost never used 530 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t)); 531 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t)); 532 } 533 j += run; 534 } 535 } 536} 537 538 539/** 540 * Related to synthesis filter 541 * Called by process_subpacket_10 542 * 543 * @param q context 544 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10 545 */ 546static void fill_tone_level_array (QDM2Context *q, int flag) 547{ 548 int i, sb, ch, sb_used; 549 int tmp, tab; 550 551 // This should never happen 552 if (q->nb_channels <= 0) 553 return; 554 555 for (ch = 0; ch < q->nb_channels; ch++) 556 for (sb = 0; sb < 30; sb++) 557 for (i = 0; i < 8; i++) { 558 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1)) 559 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+ 560 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; 561 else 562 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; 563 if(tmp < 0) 564 tmp += 0xff; 565 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff; 566 } 567 568 sb_used = QDM2_SB_USED(q->sub_sampling); 569 570 if ((q->superblocktype_2_3 != 0) && !flag) { 571 for (sb = 0; sb < sb_used; sb++) 572 for (ch = 0; ch < q->nb_channels; ch++) 573 for (i = 0; i < 64; i++) { 574 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; 575 if (q->tone_level_idx[ch][sb][i] < 0) 576 q->tone_level[ch][sb][i] = 0; 577 else 578 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f]; 579 } 580 } else { 581 tab = q->superblocktype_2_3 ? 0 : 1; 582 for (sb = 0; sb < sb_used; sb++) { 583 if ((sb >= 4) && (sb <= 23)) { 584 for (ch = 0; ch < q->nb_channels; ch++) 585 for (i = 0; i < 64; i++) { 586 tmp = q->tone_level_idx_base[ch][sb][i / 8] - 587 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] - 588 q->tone_level_idx_mid[ch][sb - 4][i / 8] - 589 q->tone_level_idx_hi2[ch][sb - 4]; 590 q->tone_level_idx[ch][sb][i] = tmp & 0xff; 591 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 592 q->tone_level[ch][sb][i] = 0; 593 else 594 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 595 } 596 } else { 597 if (sb > 4) { 598 for (ch = 0; ch < q->nb_channels; ch++) 599 for (i = 0; i < 64; i++) { 600 tmp = q->tone_level_idx_base[ch][sb][i / 8] - 601 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] - 602 q->tone_level_idx_hi2[ch][sb - 4]; 603 q->tone_level_idx[ch][sb][i] = tmp & 0xff; 604 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 605 q->tone_level[ch][sb][i] = 0; 606 else 607 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 608 } 609 } else { 610 for (ch = 0; ch < q->nb_channels; ch++) 611 for (i = 0; i < 64; i++) { 612 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; 613 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 614 q->tone_level[ch][sb][i] = 0; 615 else 616 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 617 } 618 } 619 } 620 } 621 } 622 623 return; 624} 625 626 627/** 628 * Related to synthesis filter 629 * Called by process_subpacket_11 630 * c is built with data from subpacket 11 631 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples 632 * 633 * @param tone_level_idx 634 * @param tone_level_idx_temp 635 * @param coding_method q->coding_method[0][0][0] 636 * @param nb_channels number of channels 637 * @param c coming from subpacket 11, passed as 8*c 638 * @param superblocktype_2_3 flag based on superblock packet type 639 * @param cm_table_select q->cm_table_select 640 */ 641static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp, 642 sb_int8_array coding_method, int nb_channels, 643 int c, int superblocktype_2_3, int cm_table_select) 644{ 645 int ch, sb, j; 646 int tmp, acc, esp_40, comp; 647 int add1, add2, add3, add4; 648 int64_t multres; 649 650 // This should never happen 651 if (nb_channels <= 0) 652 return; 653 654 if (!superblocktype_2_3) { 655 /* This case is untested, no samples available */ 656 SAMPLES_NEEDED 657 for (ch = 0; ch < nb_channels; ch++) 658 for (sb = 0; sb < 30; sb++) { 659 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer 660 add1 = tone_level_idx[ch][sb][j] - 10; 661 if (add1 < 0) 662 add1 = 0; 663 add2 = add3 = add4 = 0; 664 if (sb > 1) { 665 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6; 666 if (add2 < 0) 667 add2 = 0; 668 } 669 if (sb > 0) { 670 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6; 671 if (add3 < 0) 672 add3 = 0; 673 } 674 if (sb < 29) { 675 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6; 676 if (add4 < 0) 677 add4 = 0; 678 } 679 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1; 680 if (tmp < 0) 681 tmp = 0; 682 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff; 683 } 684 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1]; 685 } 686 acc = 0; 687 for (ch = 0; ch < nb_channels; ch++) 688 for (sb = 0; sb < 30; sb++) 689 for (j = 0; j < 64; j++) 690 acc += tone_level_idx_temp[ch][sb][j]; 691 692 multres = 0x66666667 * (acc * 10); 693 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31); 694 for (ch = 0; ch < nb_channels; ch++) 695 for (sb = 0; sb < 30; sb++) 696 for (j = 0; j < 64; j++) { 697 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10; 698 if (comp < 0) 699 comp += 0xff; 700 comp /= 256; // signed shift 701 switch(sb) { 702 case 0: 703 if (comp < 30) 704 comp = 30; 705 comp += 15; 706 break; 707 case 1: 708 if (comp < 24) 709 comp = 24; 710 comp += 10; 711 break; 712 case 2: 713 case 3: 714 case 4: 715 if (comp < 16) 716 comp = 16; 717 } 718 if (comp <= 5) 719 tmp = 0; 720 else if (comp <= 10) 721 tmp = 10; 722 else if (comp <= 16) 723 tmp = 16; 724 else if (comp <= 24) 725 tmp = -1; 726 else 727 tmp = 0; 728 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff; 729 } 730 for (sb = 0; sb < 30; sb++) 731 fix_coding_method_array(sb, nb_channels, coding_method); 732 for (ch = 0; ch < nb_channels; ch++) 733 for (sb = 0; sb < 30; sb++) 734 for (j = 0; j < 64; j++) 735 if (sb >= 10) { 736 if (coding_method[ch][sb][j] < 10) 737 coding_method[ch][sb][j] = 10; 738 } else { 739 if (sb >= 2) { 740 if (coding_method[ch][sb][j] < 16) 741 coding_method[ch][sb][j] = 16; 742 } else { 743 if (coding_method[ch][sb][j] < 30) 744 coding_method[ch][sb][j] = 30; 745 } 746 } 747 } else { // superblocktype_2_3 != 0 748 for (ch = 0; ch < nb_channels; ch++) 749 for (sb = 0; sb < 30; sb++) 750 for (j = 0; j < 64; j++) 751 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb]; 752 } 753 754 return; 755} 756 757 758/** 759 * 760 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8 761 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used 762 * 763 * @param q context 764 * @param gb bitreader context 765 * @param length packet length in bits 766 * @param sb_min lower subband processed (sb_min included) 767 * @param sb_max higher subband processed (sb_max excluded) 768 */ 769static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max) 770{ 771 int sb, j, k, n, ch, run, channels; 772 int joined_stereo, zero_encoding, chs; 773 int type34_first; 774 float type34_div = 0; 775 float type34_predictor; 776 float samples[10], sign_bits[16]; 777 778 if (length == 0) { 779 // If no data use noise 780 for (sb=sb_min; sb < sb_max; sb++) 781 build_sb_samples_from_noise (q, sb); 782 783 return; 784 } 785 786 for (sb = sb_min; sb < sb_max; sb++) { 787 FIX_NOISE_IDX(q->noise_idx); 788 789 channels = q->nb_channels; 790 791 if (q->nb_channels <= 1 || sb < 12) 792 joined_stereo = 0; 793 else if (sb >= 24) 794 joined_stereo = 1; 795 else 796 joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0; 797 798 if (joined_stereo) { 799 if (BITS_LEFT(length,gb) >= 16) 800 for (j = 0; j < 16; j++) 801 sign_bits[j] = get_bits1 (gb); 802 803 for (j = 0; j < 64; j++) 804 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j]) 805 q->coding_method[0][sb][j] = q->coding_method[1][sb][j]; 806 807 fix_coding_method_array(sb, q->nb_channels, q->coding_method); 808 channels = 1; 809 } 810 811 for (ch = 0; ch < channels; ch++) { 812 zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0; 813 type34_predictor = 0.0; 814 type34_first = 1; 815 816 for (j = 0; j < 128; ) { 817 switch (q->coding_method[ch][sb][j / 2]) { 818 case 8: 819 if (BITS_LEFT(length,gb) >= 10) { 820 if (zero_encoding) { 821 for (k = 0; k < 5; k++) { 822 if ((j + 2 * k) >= 128) 823 break; 824 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0; 825 } 826 } else { 827 n = get_bits(gb, 8); 828 for (k = 0; k < 5; k++) 829 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; 830 } 831 for (k = 0; k < 5; k++) 832 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx); 833 } else { 834 for (k = 0; k < 10; k++) 835 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 836 } 837 run = 10; 838 break; 839 840 case 10: 841 if (BITS_LEFT(length,gb) >= 1) { 842 float f = 0.81; 843 844 if (get_bits1(gb)) 845 f = -f; 846 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0; 847 samples[0] = f; 848 } else { 849 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 850 } 851 run = 1; 852 break; 853 854 case 16: 855 if (BITS_LEFT(length,gb) >= 10) { 856 if (zero_encoding) { 857 for (k = 0; k < 5; k++) { 858 if ((j + k) >= 128) 859 break; 860 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)]; 861 } 862 } else { 863 n = get_bits (gb, 8); 864 for (k = 0; k < 5; k++) 865 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; 866 } 867 } else { 868 for (k = 0; k < 5; k++) 869 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 870 } 871 run = 5; 872 break; 873 874 case 24: 875 if (BITS_LEFT(length,gb) >= 7) { 876 n = get_bits(gb, 7); 877 for (k = 0; k < 3; k++) 878 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5; 879 } else { 880 for (k = 0; k < 3; k++) 881 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 882 } 883 run = 3; 884 break; 885 886 case 30: 887 if (BITS_LEFT(length,gb) >= 4) { 888 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1); 889 if (index < FF_ARRAY_ELEMS(type30_dequant)) { 890 samples[0] = type30_dequant[index]; 891 } else 892 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 893 } else 894 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 895 896 run = 1; 897 break; 898 899 case 34: 900 if (BITS_LEFT(length,gb) >= 7) { 901 if (type34_first) { 902 type34_div = (float)(1 << get_bits(gb, 2)); 903 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0; 904 type34_predictor = samples[0]; 905 type34_first = 0; 906 } else { 907 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1); 908 if (index < FF_ARRAY_ELEMS(type34_delta)) { 909 samples[0] = type34_delta[index] / type34_div + type34_predictor; 910 type34_predictor = samples[0]; 911 } else 912 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 913 } 914 } else { 915 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 916 } 917 run = 1; 918 break; 919 920 default: 921 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 922 run = 1; 923 break; 924 } 925 926 if (joined_stereo) { 927 float tmp[10][MPA_MAX_CHANNELS]; 928 929 for (k = 0; k < run; k++) { 930 tmp[k][0] = samples[k]; 931 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k]; 932 } 933 for (chs = 0; chs < q->nb_channels; chs++) 934 for (k = 0; k < run; k++) 935 if ((j + k) < 128) 936 q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs]; 937 } else { 938 for (k = 0; k < run; k++) 939 if ((j + k) < 128) 940 q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k]; 941 } 942 943 j += run; 944 } // j loop 945 } // channel loop 946 } // subband loop 947} 948 949 950/** 951 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]). 952 * This is similar to process_subpacket_9, but for a single channel and for element [0] 953 * same VLC tables as process_subpacket_9 are used. 954 * 955 * @param quantized_coeffs pointer to quantized_coeffs[ch][0] 956 * @param gb bitreader context 957 * @param length packet length in bits 958 */ 959static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length) 960{ 961 int i, k, run, level, diff; 962 963 if (BITS_LEFT(length,gb) < 16) 964 return; 965 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); 966 967 quantized_coeffs[0] = level; 968 969 for (i = 0; i < 7; ) { 970 if (BITS_LEFT(length,gb) < 16) 971 break; 972 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; 973 974 if (BITS_LEFT(length,gb) < 16) 975 break; 976 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); 977 978 for (k = 1; k <= run; k++) 979 quantized_coeffs[i + k] = (level + ((k * diff) / run)); 980 981 level += diff; 982 i += run; 983 } 984} 985 986 987/** 988 * Related to synthesis filter, process data from packet 10 989 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0 990 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10 991 * 992 * @param q context 993 * @param gb bitreader context 994 * @param length packet length in bits 995 */ 996static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length) 997{ 998 int sb, j, k, n, ch; 999 1000 for (ch = 0; ch < q->nb_channels; ch++) { 1001 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length); 1002 1003 if (BITS_LEFT(length,gb) < 16) { 1004 memset(q->quantized_coeffs[ch][0], 0, 8); 1005 break; 1006 } 1007 } 1008 1009 n = q->sub_sampling + 1; 1010 1011 for (sb = 0; sb < n; sb++) 1012 for (ch = 0; ch < q->nb_channels; ch++) 1013 for (j = 0; j < 8; j++) { 1014 if (BITS_LEFT(length,gb) < 1) 1015 break; 1016 if (get_bits1(gb)) { 1017 for (k=0; k < 8; k++) { 1018 if (BITS_LEFT(length,gb) < 16) 1019 break; 1020 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2); 1021 } 1022 } else { 1023 for (k=0; k < 8; k++) 1024 q->tone_level_idx_hi1[ch][sb][j][k] = 0; 1025 } 1026 } 1027 1028 n = QDM2_SB_USED(q->sub_sampling) - 4; 1029 1030 for (sb = 0; sb < n; sb++) 1031 for (ch = 0; ch < q->nb_channels; ch++) { 1032 if (BITS_LEFT(length,gb) < 16) 1033 break; 1034 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2); 1035 if (sb > 19) 1036 q->tone_level_idx_hi2[ch][sb] -= 16; 1037 else 1038 for (j = 0; j < 8; j++) 1039 q->tone_level_idx_mid[ch][sb][j] = -16; 1040 } 1041 1042 n = QDM2_SB_USED(q->sub_sampling) - 5; 1043 1044 for (sb = 0; sb < n; sb++) 1045 for (ch = 0; ch < q->nb_channels; ch++) 1046 for (j = 0; j < 8; j++) { 1047 if (BITS_LEFT(length,gb) < 16) 1048 break; 1049 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32; 1050 } 1051} 1052 1053/** 1054 * Process subpacket 9, init quantized_coeffs with data from it 1055 * 1056 * @param q context 1057 * @param node pointer to node with packet 1058 */ 1059static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node) 1060{ 1061 GetBitContext gb; 1062 int i, j, k, n, ch, run, level, diff; 1063 1064 init_get_bits(&gb, node->packet->data, node->packet->size*8); 1065 1066 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function 1067 1068 for (i = 1; i < n; i++) 1069 for (ch=0; ch < q->nb_channels; ch++) { 1070 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2); 1071 q->quantized_coeffs[ch][i][0] = level; 1072 1073 for (j = 0; j < (8 - 1); ) { 1074 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1; 1075 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2); 1076 1077 for (k = 1; k <= run; k++) 1078 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run)); 1079 1080 level += diff; 1081 j += run; 1082 } 1083 } 1084 1085 for (ch = 0; ch < q->nb_channels; ch++) 1086 for (i = 0; i < 8; i++) 1087 q->quantized_coeffs[ch][0][i] = 0; 1088} 1089 1090 1091/** 1092 * Process subpacket 10 if not null, else 1093 * 1094 * @param q context 1095 * @param node pointer to node with packet 1096 * @param length packet length in bits 1097 */ 1098static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length) 1099{ 1100 GetBitContext gb; 1101 1102 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); 1103 1104 if (length != 0) { 1105 init_tone_level_dequantization(q, &gb, length); 1106 fill_tone_level_array(q, 1); 1107 } else { 1108 fill_tone_level_array(q, 0); 1109 } 1110} 1111 1112 1113/** 1114 * Process subpacket 11 1115 * 1116 * @param q context 1117 * @param node pointer to node with packet 1118 * @param length packet length in bit 1119 */ 1120static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length) 1121{ 1122 GetBitContext gb; 1123 1124 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); 1125 if (length >= 32) { 1126 int c = get_bits (&gb, 13); 1127 1128 if (c > 3) 1129 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method, 1130 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select); 1131 } 1132 1133 synthfilt_build_sb_samples(q, &gb, length, 0, 8); 1134} 1135 1136 1137/** 1138 * Process subpacket 12 1139 * 1140 * @param q context 1141 * @param node pointer to node with packet 1142 * @param length packet length in bits 1143 */ 1144static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length) 1145{ 1146 GetBitContext gb; 1147 1148 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); 1149 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling)); 1150} 1151 1152/* 1153 * Process new subpackets for synthesis filter 1154 * 1155 * @param q context 1156 * @param list list with synthesis filter packets (list D) 1157 */ 1158static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list) 1159{ 1160 QDM2SubPNode *nodes[4]; 1161 1162 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9); 1163 if (nodes[0] != NULL) 1164 process_subpacket_9(q, nodes[0]); 1165 1166 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10); 1167 if (nodes[1] != NULL) 1168 process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3); 1169 else 1170 process_subpacket_10(q, NULL, 0); 1171 1172 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11); 1173 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL) 1174 process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3)); 1175 else 1176 process_subpacket_11(q, NULL, 0); 1177 1178 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12); 1179 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL) 1180 process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3)); 1181 else 1182 process_subpacket_12(q, NULL, 0); 1183} 1184 1185 1186/* 1187 * Decode superblock, fill packet lists. 1188 * 1189 * @param q context 1190 */ 1191static void qdm2_decode_super_block (QDM2Context *q) 1192{ 1193 GetBitContext gb; 1194 QDM2SubPacket header, *packet; 1195 int i, packet_bytes, sub_packet_size, sub_packets_D; 1196 unsigned int next_index = 0; 1197 1198 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1)); 1199 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid)); 1200 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2)); 1201 1202 q->sub_packets_B = 0; 1203 sub_packets_D = 0; 1204 1205 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8] 1206 1207 init_get_bits(&gb, q->compressed_data, q->compressed_size*8); 1208 qdm2_decode_sub_packet_header(&gb, &header); 1209 1210 if (header.type < 2 || header.type >= 8) { 1211 q->has_errors = 1; 1212 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n"); 1213 return; 1214 } 1215 1216 q->superblocktype_2_3 = (header.type == 2 || header.type == 3); 1217 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8); 1218 1219 init_get_bits(&gb, header.data, header.size*8); 1220 1221 if (header.type == 2 || header.type == 4 || header.type == 5) { 1222 int csum = 257 * get_bits(&gb, 8); 1223 csum += 2 * get_bits(&gb, 8); 1224 1225 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); 1226 1227 if (csum != 0) { 1228 q->has_errors = 1; 1229 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n"); 1230 return; 1231 } 1232 } 1233 1234 q->sub_packet_list_B[0].packet = NULL; 1235 q->sub_packet_list_D[0].packet = NULL; 1236 1237 for (i = 0; i < 6; i++) 1238 if (--q->fft_level_exp[i] < 0) 1239 q->fft_level_exp[i] = 0; 1240 1241 for (i = 0; packet_bytes > 0; i++) { 1242 int j; 1243 1244 q->sub_packet_list_A[i].next = NULL; 1245 1246 if (i > 0) { 1247 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i]; 1248 1249 /* seek to next block */ 1250 init_get_bits(&gb, header.data, header.size*8); 1251 skip_bits(&gb, next_index*8); 1252 1253 if (next_index >= header.size) 1254 break; 1255 } 1256 1257 /* decode subpacket */ 1258 packet = &q->sub_packets[i]; 1259 qdm2_decode_sub_packet_header(&gb, packet); 1260 next_index = packet->size + get_bits_count(&gb) / 8; 1261 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2; 1262 1263 if (packet->type == 0) 1264 break; 1265 1266 if (sub_packet_size > packet_bytes) { 1267 if (packet->type != 10 && packet->type != 11 && packet->type != 12) 1268 break; 1269 packet->size += packet_bytes - sub_packet_size; 1270 } 1271 1272 packet_bytes -= sub_packet_size; 1273 1274 /* add subpacket to 'all subpackets' list */ 1275 q->sub_packet_list_A[i].packet = packet; 1276 1277 /* add subpacket to related list */ 1278 if (packet->type == 8) { 1279 SAMPLES_NEEDED_2("packet type 8"); 1280 return; 1281 } else if (packet->type >= 9 && packet->type <= 12) { 1282 /* packets for MPEG Audio like Synthesis Filter */ 1283 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet); 1284 } else if (packet->type == 13) { 1285 for (j = 0; j < 6; j++) 1286 q->fft_level_exp[j] = get_bits(&gb, 6); 1287 } else if (packet->type == 14) { 1288 for (j = 0; j < 6; j++) 1289 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2); 1290 } else if (packet->type == 15) { 1291 SAMPLES_NEEDED_2("packet type 15") 1292 return; 1293 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) { 1294 /* packets for FFT */ 1295 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet); 1296 } 1297 } // Packet bytes loop 1298 1299/* **************************************************************** */ 1300 if (q->sub_packet_list_D[0].packet != NULL) { 1301 process_synthesis_subpackets(q, q->sub_packet_list_D); 1302 q->do_synth_filter = 1; 1303 } else if (q->do_synth_filter) { 1304 process_subpacket_10(q, NULL, 0); 1305 process_subpacket_11(q, NULL, 0); 1306 process_subpacket_12(q, NULL, 0); 1307 } 1308/* **************************************************************** */ 1309} 1310 1311 1312static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet, 1313 int offset, int duration, int channel, 1314 int exp, int phase) 1315{ 1316 if (q->fft_coefs_min_index[duration] < 0) 1317 q->fft_coefs_min_index[duration] = q->fft_coefs_index; 1318 1319 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); 1320 q->fft_coefs[q->fft_coefs_index].channel = channel; 1321 q->fft_coefs[q->fft_coefs_index].offset = offset; 1322 q->fft_coefs[q->fft_coefs_index].exp = exp; 1323 q->fft_coefs[q->fft_coefs_index].phase = phase; 1324 q->fft_coefs_index++; 1325} 1326 1327 1328static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b) 1329{ 1330 int channel, stereo, phase, exp; 1331 int local_int_4, local_int_8, stereo_phase, local_int_10; 1332 int local_int_14, stereo_exp, local_int_20, local_int_28; 1333 int n, offset; 1334 1335 local_int_4 = 0; 1336 local_int_28 = 0; 1337 local_int_20 = 2; 1338 local_int_8 = (4 - duration); 1339 local_int_10 = 1 << (q->group_order - duration - 1); 1340 offset = 1; 1341 1342 while (1) { 1343 if (q->superblocktype_2_3) { 1344 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { 1345 offset = 1; 1346 if (n == 0) { 1347 local_int_4 += local_int_10; 1348 local_int_28 += (1 << local_int_8); 1349 } else { 1350 local_int_4 += 8*local_int_10; 1351 local_int_28 += (8 << local_int_8); 1352 } 1353 } 1354 offset += (n - 2); 1355 } else { 1356 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2); 1357 while (offset >= (local_int_10 - 1)) { 1358 offset += (1 - (local_int_10 - 1)); 1359 local_int_4 += local_int_10; 1360 local_int_28 += (1 << local_int_8); 1361 } 1362 } 1363 1364 if (local_int_4 >= q->group_size) 1365 return; 1366 1367 local_int_14 = (offset >> local_int_8); 1368 if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table)) 1369 return; 1370 1371 if (q->nb_channels > 1) { 1372 channel = get_bits1(gb); 1373 stereo = get_bits1(gb); 1374 } else { 1375 channel = 0; 1376 stereo = 0; 1377 } 1378 1379 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); 1380 exp += q->fft_level_exp[fft_level_index_table[local_int_14]]; 1381 exp = (exp < 0) ? 0 : exp; 1382 1383 phase = get_bits(gb, 3); 1384 stereo_exp = 0; 1385 stereo_phase = 0; 1386 1387 if (stereo) { 1388 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1)); 1389 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1)); 1390 if (stereo_phase < 0) 1391 stereo_phase += 8; 1392 } 1393 1394 if (q->frequency_range > (local_int_14 + 1)) { 1395 int sub_packet = (local_int_20 + local_int_28); 1396 1397 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase); 1398 if (stereo) 1399 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase); 1400 } 1401 1402 offset++; 1403 } 1404} 1405 1406 1407static void qdm2_decode_fft_packets (QDM2Context *q) 1408{ 1409 int i, j, min, max, value, type, unknown_flag; 1410 GetBitContext gb; 1411 1412 if (q->sub_packet_list_B[0].packet == NULL) 1413 return; 1414 1415 /* reset minimum indexes for FFT coefficients */ 1416 q->fft_coefs_index = 0; 1417 for (i=0; i < 5; i++) 1418 q->fft_coefs_min_index[i] = -1; 1419 1420 /* process subpackets ordered by type, largest type first */ 1421 for (i = 0, max = 256; i < q->sub_packets_B; i++) { 1422 QDM2SubPacket *packet= NULL; 1423 1424 /* find subpacket with largest type less than max */ 1425 for (j = 0, min = 0; j < q->sub_packets_B; j++) { 1426 value = q->sub_packet_list_B[j].packet->type; 1427 if (value > min && value < max) { 1428 min = value; 1429 packet = q->sub_packet_list_B[j].packet; 1430 } 1431 } 1432 1433 max = min; 1434 1435 /* check for errors (?) */ 1436 if (!packet) 1437 return; 1438 1439 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16])) 1440 return; 1441 1442 /* decode FFT tones */ 1443 init_get_bits (&gb, packet->data, packet->size*8); 1444 1445 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) 1446 unknown_flag = 1; 1447 else 1448 unknown_flag = 0; 1449 1450 type = packet->type; 1451 1452 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) { 1453 int duration = q->sub_sampling + 5 - (type & 15); 1454 1455 if (duration >= 0 && duration < 4) 1456 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); 1457 } else if (type == 31) { 1458 for (j=0; j < 4; j++) 1459 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); 1460 } else if (type == 46) { 1461 for (j=0; j < 6; j++) 1462 q->fft_level_exp[j] = get_bits(&gb, 6); 1463 for (j=0; j < 4; j++) 1464 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); 1465 } 1466 } // Loop on B packets 1467 1468 /* calculate maximum indexes for FFT coefficients */ 1469 for (i = 0, j = -1; i < 5; i++) 1470 if (q->fft_coefs_min_index[i] >= 0) { 1471 if (j >= 0) 1472 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i]; 1473 j = i; 1474 } 1475 if (j >= 0) 1476 q->fft_coefs_max_index[j] = q->fft_coefs_index; 1477} 1478 1479 1480static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone) 1481{ 1482 float level, f[6]; 1483 int i; 1484 QDM2Complex c; 1485 const double iscale = 2.0*M_PI / 512.0; 1486 1487 tone->phase += tone->phase_shift; 1488 1489 /* calculate current level (maximum amplitude) of tone */ 1490 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level; 1491 c.im = level * sin(tone->phase*iscale); 1492 c.re = level * cos(tone->phase*iscale); 1493 1494 /* generate FFT coefficients for tone */ 1495 if (tone->duration >= 3 || tone->cutoff >= 3) { 1496 tone->complex[0].im += c.im; 1497 tone->complex[0].re += c.re; 1498 tone->complex[1].im -= c.im; 1499 tone->complex[1].re -= c.re; 1500 } else { 1501 f[1] = -tone->table[4]; 1502 f[0] = tone->table[3] - tone->table[0]; 1503 f[2] = 1.0 - tone->table[2] - tone->table[3]; 1504 f[3] = tone->table[1] + tone->table[4] - 1.0; 1505 f[4] = tone->table[0] - tone->table[1]; 1506 f[5] = tone->table[2]; 1507 for (i = 0; i < 2; i++) { 1508 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i]; 1509 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]); 1510 } 1511 for (i = 0; i < 4; i++) { 1512 tone->complex[i].re += c.re * f[i+2]; 1513 tone->complex[i].im += c.im * f[i+2]; 1514 } 1515 } 1516 1517 /* copy the tone if it has not yet died out */ 1518 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) { 1519 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone)); 1520 q->fft_tone_end = (q->fft_tone_end + 1) % 1000; 1521 } 1522} 1523 1524 1525static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) 1526{ 1527 int i, j, ch; 1528 const double iscale = 0.25 * M_PI; 1529 1530 for (ch = 0; ch < q->channels; ch++) { 1531 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex)); 1532 } 1533 1534 1535 /* apply FFT tones with duration 4 (1 FFT period) */ 1536 if (q->fft_coefs_min_index[4] >= 0) 1537 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) { 1538 float level; 1539 QDM2Complex c; 1540 1541 if (q->fft_coefs[i].sub_packet != sub_packet) 1542 break; 1543 1544 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel; 1545 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]; 1546 1547 c.re = level * cos(q->fft_coefs[i].phase * iscale); 1548 c.im = level * sin(q->fft_coefs[i].phase * iscale); 1549 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re; 1550 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im; 1551 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re; 1552 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im; 1553 } 1554 1555 /* generate existing FFT tones */ 1556 for (i = q->fft_tone_end; i != q->fft_tone_start; ) { 1557 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]); 1558 q->fft_tone_start = (q->fft_tone_start + 1) % 1000; 1559 } 1560 1561 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */ 1562 for (i = 0; i < 4; i++) 1563 if (q->fft_coefs_min_index[i] >= 0) { 1564 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) { 1565 int offset, four_i; 1566 FFTTone tone; 1567 1568 if (q->fft_coefs[j].sub_packet != sub_packet) 1569 break; 1570 1571 four_i = (4 - i); 1572 offset = q->fft_coefs[j].offset >> four_i; 1573 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel; 1574 1575 if (offset < q->frequency_range) { 1576 if (offset < 2) 1577 tone.cutoff = offset; 1578 else 1579 tone.cutoff = (offset >= 60) ? 3 : 2; 1580 1581 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]; 1582 tone.complex = &q->fft.complex[ch][offset]; 1583 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)]; 1584 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128; 1585 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i); 1586 tone.duration = i; 1587 tone.time_index = 0; 1588 1589 qdm2_fft_generate_tone(q, &tone); 1590 } 1591 } 1592 q->fft_coefs_min_index[i] = j; 1593 } 1594} 1595 1596 1597static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) 1598{ 1599 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f; 1600 int i; 1601 q->fft.complex[channel][0].re *= 2.0f; 1602 q->fft.complex[channel][0].im = 0.0f; 1603 q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]); 1604 /* add samples to output buffer */ 1605 for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++) 1606 q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain; 1607} 1608 1609 1610/** 1611 * @param q context 1612 * @param index subpacket number 1613 */ 1614static void qdm2_synthesis_filter (QDM2Context *q, int index) 1615{ 1616 int i, k, ch, sb_used, sub_sampling, dither_state = 0; 1617 1618 /* copy sb_samples */ 1619 sb_used = QDM2_SB_USED(q->sub_sampling); 1620 1621 for (ch = 0; ch < q->channels; ch++) 1622 for (i = 0; i < 8; i++) 1623 for (k=sb_used; k < SBLIMIT; k++) 1624 q->sb_samples[ch][(8 * index) + i][k] = 0; 1625 1626 for (ch = 0; ch < q->nb_channels; ch++) { 1627 float *samples_ptr = q->samples + ch; 1628 1629 for (i = 0; i < 8; i++) { 1630 ff_mpa_synth_filter_float(&q->mpadsp, 1631 q->synth_buf[ch], &(q->synth_buf_offset[ch]), 1632 ff_mpa_synth_window_float, &dither_state, 1633 samples_ptr, q->nb_channels, 1634 q->sb_samples[ch][(8 * index) + i]); 1635 samples_ptr += 32 * q->nb_channels; 1636 } 1637 } 1638 1639 /* add samples to output buffer */ 1640 sub_sampling = (4 >> q->sub_sampling); 1641 1642 for (ch = 0; ch < q->channels; ch++) 1643 for (i = 0; i < q->frame_size; i++) 1644 q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch]; 1645} 1646 1647 1648/** 1649 * Init static data (does not depend on specific file) 1650 * 1651 * @param q context 1652 */ 1653static av_cold void qdm2_init(QDM2Context *q) { 1654 static int initialized = 0; 1655 1656 if (initialized != 0) 1657 return; 1658 initialized = 1; 1659 1660 qdm2_init_vlc(); 1661 ff_mpa_synth_init_float(ff_mpa_synth_window_float); 1662 softclip_table_init(); 1663 rnd_table_init(); 1664 init_noise_samples(); 1665 1666 av_log(NULL, AV_LOG_DEBUG, "init done\n"); 1667} 1668 1669 1670#if 0 1671static void dump_context(QDM2Context *q) 1672{ 1673 int i; 1674#define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b); 1675 PRINT("compressed_data",q->compressed_data); 1676 PRINT("compressed_size",q->compressed_size); 1677 PRINT("frame_size",q->frame_size); 1678 PRINT("checksum_size",q->checksum_size); 1679 PRINT("channels",q->channels); 1680 PRINT("nb_channels",q->nb_channels); 1681 PRINT("fft_frame_size",q->fft_frame_size); 1682 PRINT("fft_size",q->fft_size); 1683 PRINT("sub_sampling",q->sub_sampling); 1684 PRINT("fft_order",q->fft_order); 1685 PRINT("group_order",q->group_order); 1686 PRINT("group_size",q->group_size); 1687 PRINT("sub_packet",q->sub_packet); 1688 PRINT("frequency_range",q->frequency_range); 1689 PRINT("has_errors",q->has_errors); 1690 PRINT("fft_tone_end",q->fft_tone_end); 1691 PRINT("fft_tone_start",q->fft_tone_start); 1692 PRINT("fft_coefs_index",q->fft_coefs_index); 1693 PRINT("coeff_per_sb_select",q->coeff_per_sb_select); 1694 PRINT("cm_table_select",q->cm_table_select); 1695 PRINT("noise_idx",q->noise_idx); 1696 1697 for (i = q->fft_tone_start; i < q->fft_tone_end; i++) 1698 { 1699 FFTTone *t = &q->fft_tones[i]; 1700 1701 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i); 1702 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level); 1703// PRINT(" level", t->level); 1704 PRINT(" phase", t->phase); 1705 PRINT(" phase_shift", t->phase_shift); 1706 PRINT(" duration", t->duration); 1707 PRINT(" samples_im", t->samples_im); 1708 PRINT(" samples_re", t->samples_re); 1709 PRINT(" table", t->table); 1710 } 1711 1712} 1713#endif 1714 1715 1716/** 1717 * Init parameters from codec extradata 1718 */ 1719static av_cold int qdm2_decode_init(AVCodecContext *avctx) 1720{ 1721 QDM2Context *s = avctx->priv_data; 1722 uint8_t *extradata; 1723 int extradata_size; 1724 int tmp_val, tmp, size; 1725 1726 /* extradata parsing 1727 1728 Structure: 1729 wave { 1730 frma (QDM2) 1731 QDCA 1732 QDCP 1733 } 1734 1735 32 size (including this field) 1736 32 tag (=frma) 1737 32 type (=QDM2 or QDMC) 1738 1739 32 size (including this field, in bytes) 1740 32 tag (=QDCA) // maybe mandatory parameters 1741 32 unknown (=1) 1742 32 channels (=2) 1743 32 samplerate (=44100) 1744 32 bitrate (=96000) 1745 32 block size (=4096) 1746 32 frame size (=256) (for one channel) 1747 32 packet size (=1300) 1748 1749 32 size (including this field, in bytes) 1750 32 tag (=QDCP) // maybe some tuneable parameters 1751 32 float1 (=1.0) 1752 32 zero ? 1753 32 float2 (=1.0) 1754 32 float3 (=1.0) 1755 32 unknown (27) 1756 32 unknown (8) 1757 32 zero ? 1758 */ 1759 1760 if (!avctx->extradata || (avctx->extradata_size < 48)) { 1761 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n"); 1762 return -1; 1763 } 1764 1765 extradata = avctx->extradata; 1766 extradata_size = avctx->extradata_size; 1767 1768 while (extradata_size > 7) { 1769 if (!memcmp(extradata, "frmaQDM", 7)) 1770 break; 1771 extradata++; 1772 extradata_size--; 1773 } 1774 1775 if (extradata_size < 12) { 1776 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n", 1777 extradata_size); 1778 return -1; 1779 } 1780 1781 if (memcmp(extradata, "frmaQDM", 7)) { 1782 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n"); 1783 return -1; 1784 } 1785 1786 if (extradata[7] == 'C') { 1787// s->is_qdmc = 1; 1788 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n"); 1789 return -1; 1790 } 1791 1792 extradata += 8; 1793 extradata_size -= 8; 1794 1795 size = AV_RB32(extradata); 1796 1797 if(size > extradata_size){ 1798 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n", 1799 extradata_size, size); 1800 return -1; 1801 } 1802 1803 extradata += 4; 1804 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size); 1805 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) { 1806 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n"); 1807 return -1; 1808 } 1809 1810 extradata += 8; 1811 1812 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata); 1813 extradata += 4; 1814 if (s->channels > MPA_MAX_CHANNELS) 1815 return AVERROR_INVALIDDATA; 1816 1817 avctx->sample_rate = AV_RB32(extradata); 1818 extradata += 4; 1819 1820 avctx->bit_rate = AV_RB32(extradata); 1821 extradata += 4; 1822 1823 s->group_size = AV_RB32(extradata); 1824 extradata += 4; 1825 1826 s->fft_size = AV_RB32(extradata); 1827 extradata += 4; 1828 1829 s->checksum_size = AV_RB32(extradata); 1830 if (s->checksum_size >= 1U << 28) { 1831 av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size); 1832 return AVERROR_INVALIDDATA; 1833 } 1834 1835 s->fft_order = av_log2(s->fft_size) + 1; 1836 s->fft_frame_size = 2 * s->fft_size; // complex has two floats 1837 1838 // something like max decodable tones 1839 s->group_order = av_log2(s->group_size) + 1; 1840 s->frame_size = s->group_size / 16; // 16 iterations per super block 1841 if (s->frame_size > QDM2_MAX_FRAME_SIZE) 1842 return AVERROR_INVALIDDATA; 1843 1844 s->sub_sampling = s->fft_order - 7; 1845 s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); 1846 1847 switch ((s->sub_sampling * 2 + s->channels - 1)) { 1848 case 0: tmp = 40; break; 1849 case 1: tmp = 48; break; 1850 case 2: tmp = 56; break; 1851 case 3: tmp = 72; break; 1852 case 4: tmp = 80; break; 1853 case 5: tmp = 100;break; 1854 default: tmp=s->sub_sampling; break; 1855 } 1856 tmp_val = 0; 1857 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1; 1858 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2; 1859 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3; 1860 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4; 1861 s->cm_table_select = tmp_val; 1862 1863 if (s->sub_sampling == 0) 1864 tmp = 7999; 1865 else 1866 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000; 1867 /* 1868 0: 7999 -> 0 1869 1: 20000 -> 2 1870 2: 28000 -> 2 1871 */ 1872 if (tmp < 8000) 1873 s->coeff_per_sb_select = 0; 1874 else if (tmp <= 16000) 1875 s->coeff_per_sb_select = 1; 1876 else 1877 s->coeff_per_sb_select = 2; 1878 1879 // Fail on unknown fft order 1880 if ((s->fft_order < 7) || (s->fft_order > 9)) { 1881 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order); 1882 return -1; 1883 } 1884 if (s->fft_size != (1 << (s->fft_order - 1))) { 1885 av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size); 1886 return AVERROR_INVALIDDATA; 1887 } 1888 1889 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R); 1890 ff_mpadsp_init(&s->mpadsp); 1891 1892 qdm2_init(s); 1893 1894 avctx->sample_fmt = AV_SAMPLE_FMT_S16; 1895 1896 avcodec_get_frame_defaults(&s->frame); 1897 avctx->coded_frame = &s->frame; 1898 1899// dump_context(s); 1900 return 0; 1901} 1902 1903 1904static av_cold int qdm2_decode_close(AVCodecContext *avctx) 1905{ 1906 QDM2Context *s = avctx->priv_data; 1907 1908 ff_rdft_end(&s->rdft_ctx); 1909 1910 return 0; 1911} 1912 1913 1914static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) 1915{ 1916 int ch, i; 1917 const int frame_size = (q->frame_size * q->channels); 1918 1919 /* select input buffer */ 1920 q->compressed_data = in; 1921 q->compressed_size = q->checksum_size; 1922 1923// dump_context(q); 1924 1925 /* copy old block, clear new block of output samples */ 1926 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float)); 1927 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float)); 1928 1929 /* decode block of QDM2 compressed data */ 1930 if (q->sub_packet == 0) { 1931 q->has_errors = 0; // zero it for a new super block 1932 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n"); 1933 qdm2_decode_super_block(q); 1934 } 1935 1936 /* parse subpackets */ 1937 if (!q->has_errors) { 1938 if (q->sub_packet == 2) 1939 qdm2_decode_fft_packets(q); 1940 1941 qdm2_fft_tone_synthesizer(q, q->sub_packet); 1942 } 1943 1944 /* sound synthesis stage 1 (FFT) */ 1945 for (ch = 0; ch < q->channels; ch++) { 1946 qdm2_calculate_fft(q, ch, q->sub_packet); 1947 1948 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) { 1949 SAMPLES_NEEDED_2("has errors, and C list is not empty") 1950 return -1; 1951 } 1952 } 1953 1954 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */ 1955 if (!q->has_errors && q->do_synth_filter) 1956 qdm2_synthesis_filter(q, q->sub_packet); 1957 1958 q->sub_packet = (q->sub_packet + 1) % 16; 1959 1960 /* clip and convert output float[] to 16bit signed samples */ 1961 for (i = 0; i < frame_size; i++) { 1962 int value = (int)q->output_buffer[i]; 1963 1964 if (value > SOFTCLIP_THRESHOLD) 1965 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD]; 1966 else if (value < -SOFTCLIP_THRESHOLD) 1967 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD]; 1968 1969 out[i] = value; 1970 } 1971 1972 return 0; 1973} 1974 1975 1976static int qdm2_decode_frame(AVCodecContext *avctx, void *data, 1977 int *got_frame_ptr, AVPacket *avpkt) 1978{ 1979 const uint8_t *buf = avpkt->data; 1980 int buf_size = avpkt->size; 1981 QDM2Context *s = avctx->priv_data; 1982 int16_t *out; 1983 int i, ret; 1984 1985 if(!buf) 1986 return 0; 1987 if(buf_size < s->checksum_size) 1988 return -1; 1989 1990 /* get output buffer */ 1991 s->frame.nb_samples = 16 * s->frame_size; 1992 if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { 1993 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); 1994 return ret; 1995 } 1996 out = (int16_t *)s->frame.data[0]; 1997 1998 for (i = 0; i < 16; i++) { 1999 if (qdm2_decode(s, buf, out) < 0) 2000 return -1; 2001 out += s->channels * s->frame_size; 2002 } 2003 2004 *got_frame_ptr = 1; 2005 *(AVFrame *)data = s->frame; 2006 2007 return s->checksum_size; 2008} 2009 2010AVCodec ff_qdm2_decoder = 2011{ 2012 .name = "qdm2", 2013 .type = AVMEDIA_TYPE_AUDIO, 2014 .id = CODEC_ID_QDM2, 2015 .priv_data_size = sizeof(QDM2Context), 2016 .init = qdm2_decode_init, 2017 .close = qdm2_decode_close, 2018 .decode = qdm2_decode_frame, 2019 .capabilities = CODEC_CAP_DR1, 2020 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), 2021}; 2022