1/* 2 * ATRAC3 compatible decoder 3 * Copyright (c) 2006-2008 Maxim Poliakovski 4 * Copyright (c) 2006-2008 Benjamin Larsson 5 * 6 * This file is part of FFmpeg. 7 * 8 * FFmpeg is free software; you can redistribute it and/or 9 * modify it under the terms of the GNU Lesser General Public 10 * License as published by the Free Software Foundation; either 11 * version 2.1 of the License, or (at your option) any later version. 12 * 13 * FFmpeg is distributed in the hope that it will be useful, 14 * but WITHOUT ANY WARRANTY; without even the implied warranty of 15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 16 * Lesser General Public License for more details. 17 * 18 * You should have received a copy of the GNU Lesser General Public 19 * License along with FFmpeg; if not, write to the Free Software 20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 21 */ 22 23/** 24 * @file 25 * ATRAC3 compatible decoder. 26 * This decoder handles Sony's ATRAC3 data. 27 * 28 * Container formats used to store ATRAC3 data: 29 * RealMedia (.rm), RIFF WAV (.wav, .at3), Sony OpenMG (.oma, .aa3). 30 * 31 * To use this decoder, a calling application must supply the extradata 32 * bytes provided in the containers above. 33 */ 34 35#include <math.h> 36#include <stddef.h> 37#include <stdio.h> 38 39#include "libavutil/attributes.h" 40#include "libavutil/float_dsp.h" 41#include "libavutil/libm.h" 42#include "avcodec.h" 43#include "bytestream.h" 44#include "fft.h" 45#include "fmtconvert.h" 46#include "get_bits.h" 47#include "internal.h" 48 49#include "atrac.h" 50#include "atrac3data.h" 51 52#define JOINT_STEREO 0x12 53#define STEREO 0x2 54 55#define SAMPLES_PER_FRAME 1024 56#define MDCT_SIZE 512 57 58typedef struct GainBlock { 59 AtracGainInfo g_block[4]; 60} GainBlock; 61 62typedef struct TonalComponent { 63 int pos; 64 int num_coefs; 65 float coef[8]; 66} TonalComponent; 67 68typedef struct ChannelUnit { 69 int bands_coded; 70 int num_components; 71 float prev_frame[SAMPLES_PER_FRAME]; 72 int gc_blk_switch; 73 TonalComponent components[64]; 74 GainBlock gain_block[2]; 75 76 DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME]; 77 DECLARE_ALIGNED(32, float, imdct_buf)[SAMPLES_PER_FRAME]; 78 79 float delay_buf1[46]; ///<qmf delay buffers 80 float delay_buf2[46]; 81 float delay_buf3[46]; 82} ChannelUnit; 83 84typedef struct ATRAC3Context { 85 GetBitContext gb; 86 //@{ 87 /** stream data */ 88 int coding_mode; 89 90 ChannelUnit *units; 91 //@} 92 //@{ 93 /** joint-stereo related variables */ 94 int matrix_coeff_index_prev[4]; 95 int matrix_coeff_index_now[4]; 96 int matrix_coeff_index_next[4]; 97 int weighting_delay[6]; 98 //@} 99 //@{ 100 /** data buffers */ 101 uint8_t *decoded_bytes_buffer; 102 float temp_buf[1070]; 103 //@} 104 //@{ 105 /** extradata */ 106 int scrambled_stream; 107 //@} 108 109 AtracGCContext gainc_ctx; 110 FFTContext mdct_ctx; 111 FmtConvertContext fmt_conv; 112 AVFloatDSPContext fdsp; 113} ATRAC3Context; 114 115static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE]; 116static VLC_TYPE atrac3_vlc_table[4096][2]; 117static VLC spectral_coeff_tab[7]; 118 119/** 120 * Regular 512 points IMDCT without overlapping, with the exception of the 121 * swapping of odd bands caused by the reverse spectra of the QMF. 122 * 123 * @param odd_band 1 if the band is an odd band 124 */ 125static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band) 126{ 127 int i; 128 129 if (odd_band) { 130 /** 131 * Reverse the odd bands before IMDCT, this is an effect of the QMF 132 * transform or it gives better compression to do it this way. 133 * FIXME: It should be possible to handle this in imdct_calc 134 * for that to happen a modification of the prerotation step of 135 * all SIMD code and C code is needed. 136 * Or fix the functions before so they generate a pre reversed spectrum. 137 */ 138 for (i = 0; i < 128; i++) 139 FFSWAP(float, input[i], input[255 - i]); 140 } 141 142 q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input); 143 144 /* Perform windowing on the output. */ 145 q->fdsp.vector_fmul(output, output, mdct_window, MDCT_SIZE); 146} 147 148/* 149 * indata descrambling, only used for data coming from the rm container 150 */ 151static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes) 152{ 153 int i, off; 154 uint32_t c; 155 const uint32_t *buf; 156 uint32_t *output = (uint32_t *)out; 157 158 off = (intptr_t)input & 3; 159 buf = (const uint32_t *)(input - off); 160 if (off) 161 c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8)))); 162 else 163 c = av_be2ne32(0x537F6103U); 164 bytes += 3 + off; 165 for (i = 0; i < bytes / 4; i++) 166 output[i] = c ^ buf[i]; 167 168 if (off) 169 avpriv_request_sample(NULL, "Offset of %d", off); 170 171 return off; 172} 173 174static av_cold void init_imdct_window(void) 175{ 176 int i, j; 177 178 /* generate the mdct window, for details see 179 * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */ 180 for (i = 0, j = 255; i < 128; i++, j--) { 181 float wi = sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0; 182 float wj = sin(((j + 0.5) / 256.0 - 0.5) * M_PI) + 1.0; 183 float w = 0.5 * (wi * wi + wj * wj); 184 mdct_window[i] = mdct_window[511 - i] = wi / w; 185 mdct_window[j] = mdct_window[511 - j] = wj / w; 186 } 187} 188 189static av_cold int atrac3_decode_close(AVCodecContext *avctx) 190{ 191 ATRAC3Context *q = avctx->priv_data; 192 193 av_free(q->units); 194 av_free(q->decoded_bytes_buffer); 195 196 ff_mdct_end(&q->mdct_ctx); 197 198 return 0; 199} 200 201/** 202 * Mantissa decoding 203 * 204 * @param selector which table the output values are coded with 205 * @param coding_flag constant length coding or variable length coding 206 * @param mantissas mantissa output table 207 * @param num_codes number of values to get 208 */ 209static void read_quant_spectral_coeffs(GetBitContext *gb, int selector, 210 int coding_flag, int *mantissas, 211 int num_codes) 212{ 213 int i, code, huff_symb; 214 215 if (selector == 1) 216 num_codes /= 2; 217 218 if (coding_flag != 0) { 219 /* constant length coding (CLC) */ 220 int num_bits = clc_length_tab[selector]; 221 222 if (selector > 1) { 223 for (i = 0; i < num_codes; i++) { 224 if (num_bits) 225 code = get_sbits(gb, num_bits); 226 else 227 code = 0; 228 mantissas[i] = code; 229 } 230 } else { 231 for (i = 0; i < num_codes; i++) { 232 if (num_bits) 233 code = get_bits(gb, num_bits); // num_bits is always 4 in this case 234 else 235 code = 0; 236 mantissas[i * 2 ] = mantissa_clc_tab[code >> 2]; 237 mantissas[i * 2 + 1] = mantissa_clc_tab[code & 3]; 238 } 239 } 240 } else { 241 /* variable length coding (VLC) */ 242 if (selector != 1) { 243 for (i = 0; i < num_codes; i++) { 244 huff_symb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, 245 spectral_coeff_tab[selector-1].bits, 3); 246 huff_symb += 1; 247 code = huff_symb >> 1; 248 if (huff_symb & 1) 249 code = -code; 250 mantissas[i] = code; 251 } 252 } else { 253 for (i = 0; i < num_codes; i++) { 254 huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table, 255 spectral_coeff_tab[selector - 1].bits, 3); 256 mantissas[i * 2 ] = mantissa_vlc_tab[huff_symb * 2 ]; 257 mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1]; 258 } 259 } 260 } 261} 262 263/** 264 * Restore the quantized band spectrum coefficients 265 * 266 * @return subband count, fix for broken specification/files 267 */ 268static int decode_spectrum(GetBitContext *gb, float *output) 269{ 270 int num_subbands, coding_mode, i, j, first, last, subband_size; 271 int subband_vlc_index[32], sf_index[32]; 272 int mantissas[128]; 273 float scale_factor; 274 275 num_subbands = get_bits(gb, 5); // number of coded subbands 276 coding_mode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC 277 278 /* get the VLC selector table for the subbands, 0 means not coded */ 279 for (i = 0; i <= num_subbands; i++) 280 subband_vlc_index[i] = get_bits(gb, 3); 281 282 /* read the scale factor indexes from the stream */ 283 for (i = 0; i <= num_subbands; i++) { 284 if (subband_vlc_index[i] != 0) 285 sf_index[i] = get_bits(gb, 6); 286 } 287 288 for (i = 0; i <= num_subbands; i++) { 289 first = subband_tab[i ]; 290 last = subband_tab[i + 1]; 291 292 subband_size = last - first; 293 294 if (subband_vlc_index[i] != 0) { 295 /* decode spectral coefficients for this subband */ 296 /* TODO: This can be done faster is several blocks share the 297 * same VLC selector (subband_vlc_index) */ 298 read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode, 299 mantissas, subband_size); 300 301 /* decode the scale factor for this subband */ 302 scale_factor = ff_atrac_sf_table[sf_index[i]] * 303 inv_max_quant[subband_vlc_index[i]]; 304 305 /* inverse quantize the coefficients */ 306 for (j = 0; first < last; first++, j++) 307 output[first] = mantissas[j] * scale_factor; 308 } else { 309 /* this subband was not coded, so zero the entire subband */ 310 memset(output + first, 0, subband_size * sizeof(*output)); 311 } 312 } 313 314 /* clear the subbands that were not coded */ 315 first = subband_tab[i]; 316 memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(*output)); 317 return num_subbands; 318} 319 320/** 321 * Restore the quantized tonal components 322 * 323 * @param components tonal components 324 * @param num_bands number of coded bands 325 */ 326static int decode_tonal_components(GetBitContext *gb, 327 TonalComponent *components, int num_bands) 328{ 329 int i, b, c, m; 330 int nb_components, coding_mode_selector, coding_mode; 331 int band_flags[4], mantissa[8]; 332 int component_count = 0; 333 334 nb_components = get_bits(gb, 5); 335 336 /* no tonal components */ 337 if (nb_components == 0) 338 return 0; 339 340 coding_mode_selector = get_bits(gb, 2); 341 if (coding_mode_selector == 2) 342 return AVERROR_INVALIDDATA; 343 344 coding_mode = coding_mode_selector & 1; 345 346 for (i = 0; i < nb_components; i++) { 347 int coded_values_per_component, quant_step_index; 348 349 for (b = 0; b <= num_bands; b++) 350 band_flags[b] = get_bits1(gb); 351 352 coded_values_per_component = get_bits(gb, 3); 353 354 quant_step_index = get_bits(gb, 3); 355 if (quant_step_index <= 1) 356 return AVERROR_INVALIDDATA; 357 358 if (coding_mode_selector == 3) 359 coding_mode = get_bits1(gb); 360 361 for (b = 0; b < (num_bands + 1) * 4; b++) { 362 int coded_components; 363 364 if (band_flags[b >> 2] == 0) 365 continue; 366 367 coded_components = get_bits(gb, 3); 368 369 for (c = 0; c < coded_components; c++) { 370 TonalComponent *cmp = &components[component_count]; 371 int sf_index, coded_values, max_coded_values; 372 float scale_factor; 373 374 sf_index = get_bits(gb, 6); 375 if (component_count >= 64) 376 return AVERROR_INVALIDDATA; 377 378 cmp->pos = b * 64 + get_bits(gb, 6); 379 380 max_coded_values = SAMPLES_PER_FRAME - cmp->pos; 381 coded_values = coded_values_per_component + 1; 382 coded_values = FFMIN(max_coded_values, coded_values); 383 384 scale_factor = ff_atrac_sf_table[sf_index] * 385 inv_max_quant[quant_step_index]; 386 387 read_quant_spectral_coeffs(gb, quant_step_index, coding_mode, 388 mantissa, coded_values); 389 390 cmp->num_coefs = coded_values; 391 392 /* inverse quant */ 393 for (m = 0; m < coded_values; m++) 394 cmp->coef[m] = mantissa[m] * scale_factor; 395 396 component_count++; 397 } 398 } 399 } 400 401 return component_count; 402} 403 404/** 405 * Decode gain parameters for the coded bands 406 * 407 * @param block the gainblock for the current band 408 * @param num_bands amount of coded bands 409 */ 410static int decode_gain_control(GetBitContext *gb, GainBlock *block, 411 int num_bands) 412{ 413 int b, j; 414 int *level, *loc; 415 416 AtracGainInfo *gain = block->g_block; 417 418 for (b = 0; b <= num_bands; b++) { 419 gain[b].num_points = get_bits(gb, 3); 420 level = gain[b].lev_code; 421 loc = gain[b].loc_code; 422 423 for (j = 0; j < gain[b].num_points; j++) { 424 level[j] = get_bits(gb, 4); 425 loc[j] = get_bits(gb, 5); 426 if (j && loc[j] <= loc[j - 1]) 427 return AVERROR_INVALIDDATA; 428 } 429 } 430 431 /* Clear the unused blocks. */ 432 for (; b < 4 ; b++) 433 gain[b].num_points = 0; 434 435 return 0; 436} 437 438/** 439 * Combine the tonal band spectrum and regular band spectrum 440 * 441 * @param spectrum output spectrum buffer 442 * @param num_components number of tonal components 443 * @param components tonal components for this band 444 * @return position of the last tonal coefficient 445 */ 446static int add_tonal_components(float *spectrum, int num_components, 447 TonalComponent *components) 448{ 449 int i, j, last_pos = -1; 450 float *input, *output; 451 452 for (i = 0; i < num_components; i++) { 453 last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos); 454 input = components[i].coef; 455 output = &spectrum[components[i].pos]; 456 457 for (j = 0; j < components[i].num_coefs; j++) 458 output[j] += input[j]; 459 } 460 461 return last_pos; 462} 463 464#define INTERPOLATE(old, new, nsample) \ 465 ((old) + (nsample) * 0.125 * ((new) - (old))) 466 467static void reverse_matrixing(float *su1, float *su2, int *prev_code, 468 int *curr_code) 469{ 470 int i, nsample, band; 471 float mc1_l, mc1_r, mc2_l, mc2_r; 472 473 for (i = 0, band = 0; band < 4 * 256; band += 256, i++) { 474 int s1 = prev_code[i]; 475 int s2 = curr_code[i]; 476 nsample = band; 477 478 if (s1 != s2) { 479 /* Selector value changed, interpolation needed. */ 480 mc1_l = matrix_coeffs[s1 * 2 ]; 481 mc1_r = matrix_coeffs[s1 * 2 + 1]; 482 mc2_l = matrix_coeffs[s2 * 2 ]; 483 mc2_r = matrix_coeffs[s2 * 2 + 1]; 484 485 /* Interpolation is done over the first eight samples. */ 486 for (; nsample < band + 8; nsample++) { 487 float c1 = su1[nsample]; 488 float c2 = su2[nsample]; 489 c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample - band) + 490 c2 * INTERPOLATE(mc1_r, mc2_r, nsample - band); 491 su1[nsample] = c2; 492 su2[nsample] = c1 * 2.0 - c2; 493 } 494 } 495 496 /* Apply the matrix without interpolation. */ 497 switch (s2) { 498 case 0: /* M/S decoding */ 499 for (; nsample < band + 256; nsample++) { 500 float c1 = su1[nsample]; 501 float c2 = su2[nsample]; 502 su1[nsample] = c2 * 2.0; 503 su2[nsample] = (c1 - c2) * 2.0; 504 } 505 break; 506 case 1: 507 for (; nsample < band + 256; nsample++) { 508 float c1 = su1[nsample]; 509 float c2 = su2[nsample]; 510 su1[nsample] = (c1 + c2) * 2.0; 511 su2[nsample] = c2 * -2.0; 512 } 513 break; 514 case 2: 515 case 3: 516 for (; nsample < band + 256; nsample++) { 517 float c1 = su1[nsample]; 518 float c2 = su2[nsample]; 519 su1[nsample] = c1 + c2; 520 su2[nsample] = c1 - c2; 521 } 522 break; 523 default: 524 av_assert1(0); 525 } 526 } 527} 528 529static void get_channel_weights(int index, int flag, float ch[2]) 530{ 531 if (index == 7) { 532 ch[0] = 1.0; 533 ch[1] = 1.0; 534 } else { 535 ch[0] = (index & 7) / 7.0; 536 ch[1] = sqrt(2 - ch[0] * ch[0]); 537 if (flag) 538 FFSWAP(float, ch[0], ch[1]); 539 } 540} 541 542static void channel_weighting(float *su1, float *su2, int *p3) 543{ 544 int band, nsample; 545 /* w[x][y] y=0 is left y=1 is right */ 546 float w[2][2]; 547 548 if (p3[1] != 7 || p3[3] != 7) { 549 get_channel_weights(p3[1], p3[0], w[0]); 550 get_channel_weights(p3[3], p3[2], w[1]); 551 552 for (band = 256; band < 4 * 256; band += 256) { 553 for (nsample = band; nsample < band + 8; nsample++) { 554 su1[nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample - band); 555 su2[nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample - band); 556 } 557 for(; nsample < band + 256; nsample++) { 558 su1[nsample] *= w[1][0]; 559 su2[nsample] *= w[1][1]; 560 } 561 } 562 } 563} 564 565/** 566 * Decode a Sound Unit 567 * 568 * @param snd the channel unit to be used 569 * @param output the decoded samples before IQMF in float representation 570 * @param channel_num channel number 571 * @param coding_mode the coding mode (JOINT_STEREO or regular stereo/mono) 572 */ 573static int decode_channel_sound_unit(ATRAC3Context *q, GetBitContext *gb, 574 ChannelUnit *snd, float *output, 575 int channel_num, int coding_mode) 576{ 577 int band, ret, num_subbands, last_tonal, num_bands; 578 GainBlock *gain1 = &snd->gain_block[ snd->gc_blk_switch]; 579 GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch]; 580 581 if (coding_mode == JOINT_STEREO && channel_num == 1) { 582 if (get_bits(gb, 2) != 3) { 583 av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n"); 584 return AVERROR_INVALIDDATA; 585 } 586 } else { 587 if (get_bits(gb, 6) != 0x28) { 588 av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n"); 589 return AVERROR_INVALIDDATA; 590 } 591 } 592 593 /* number of coded QMF bands */ 594 snd->bands_coded = get_bits(gb, 2); 595 596 ret = decode_gain_control(gb, gain2, snd->bands_coded); 597 if (ret) 598 return ret; 599 600 snd->num_components = decode_tonal_components(gb, snd->components, 601 snd->bands_coded); 602 if (snd->num_components < 0) 603 return snd->num_components; 604 605 num_subbands = decode_spectrum(gb, snd->spectrum); 606 607 /* Merge the decoded spectrum and tonal components. */ 608 last_tonal = add_tonal_components(snd->spectrum, snd->num_components, 609 snd->components); 610 611 612 /* calculate number of used MLT/QMF bands according to the amount of coded 613 spectral lines */ 614 num_bands = (subband_tab[num_subbands] - 1) >> 8; 615 if (last_tonal >= 0) 616 num_bands = FFMAX((last_tonal + 256) >> 8, num_bands); 617 618 619 /* Reconstruct time domain samples. */ 620 for (band = 0; band < 4; band++) { 621 /* Perform the IMDCT step without overlapping. */ 622 if (band <= num_bands) 623 imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1); 624 else 625 memset(snd->imdct_buf, 0, 512 * sizeof(*snd->imdct_buf)); 626 627 /* gain compensation and overlapping */ 628 ff_atrac_gain_compensation(&q->gainc_ctx, snd->imdct_buf, 629 &snd->prev_frame[band * 256], 630 &gain1->g_block[band], &gain2->g_block[band], 631 256, &output[band * 256]); 632 } 633 634 /* Swap the gain control buffers for the next frame. */ 635 snd->gc_blk_switch ^= 1; 636 637 return 0; 638} 639 640static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf, 641 float **out_samples) 642{ 643 ATRAC3Context *q = avctx->priv_data; 644 int ret, i; 645 uint8_t *ptr1; 646 647 if (q->coding_mode == JOINT_STEREO) { 648 /* channel coupling mode */ 649 /* decode Sound Unit 1 */ 650 init_get_bits(&q->gb, databuf, avctx->block_align * 8); 651 652 ret = decode_channel_sound_unit(q, &q->gb, q->units, out_samples[0], 0, 653 JOINT_STEREO); 654 if (ret != 0) 655 return ret; 656 657 /* Framedata of the su2 in the joint-stereo mode is encoded in 658 * reverse byte order so we need to swap it first. */ 659 if (databuf == q->decoded_bytes_buffer) { 660 uint8_t *ptr2 = q->decoded_bytes_buffer + avctx->block_align - 1; 661 ptr1 = q->decoded_bytes_buffer; 662 for (i = 0; i < avctx->block_align / 2; i++, ptr1++, ptr2--) 663 FFSWAP(uint8_t, *ptr1, *ptr2); 664 } else { 665 const uint8_t *ptr2 = databuf + avctx->block_align - 1; 666 for (i = 0; i < avctx->block_align; i++) 667 q->decoded_bytes_buffer[i] = *ptr2--; 668 } 669 670 /* Skip the sync codes (0xF8). */ 671 ptr1 = q->decoded_bytes_buffer; 672 for (i = 4; *ptr1 == 0xF8; i++, ptr1++) { 673 if (i >= avctx->block_align) 674 return AVERROR_INVALIDDATA; 675 } 676 677 678 /* set the bitstream reader at the start of the second Sound Unit*/ 679 init_get_bits8(&q->gb, ptr1, q->decoded_bytes_buffer + avctx->block_align - ptr1); 680 681 /* Fill the Weighting coeffs delay buffer */ 682 memmove(q->weighting_delay, &q->weighting_delay[2], 683 4 * sizeof(*q->weighting_delay)); 684 q->weighting_delay[4] = get_bits1(&q->gb); 685 q->weighting_delay[5] = get_bits(&q->gb, 3); 686 687 for (i = 0; i < 4; i++) { 688 q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i]; 689 q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i]; 690 q->matrix_coeff_index_next[i] = get_bits(&q->gb, 2); 691 } 692 693 /* Decode Sound Unit 2. */ 694 ret = decode_channel_sound_unit(q, &q->gb, &q->units[1], 695 out_samples[1], 1, JOINT_STEREO); 696 if (ret != 0) 697 return ret; 698 699 /* Reconstruct the channel coefficients. */ 700 reverse_matrixing(out_samples[0], out_samples[1], 701 q->matrix_coeff_index_prev, 702 q->matrix_coeff_index_now); 703 704 channel_weighting(out_samples[0], out_samples[1], q->weighting_delay); 705 } else { 706 /* normal stereo mode or mono */ 707 /* Decode the channel sound units. */ 708 for (i = 0; i < avctx->channels; i++) { 709 /* Set the bitstream reader at the start of a channel sound unit. */ 710 init_get_bits(&q->gb, 711 databuf + i * avctx->block_align / avctx->channels, 712 avctx->block_align * 8 / avctx->channels); 713 714 ret = decode_channel_sound_unit(q, &q->gb, &q->units[i], 715 out_samples[i], i, q->coding_mode); 716 if (ret != 0) 717 return ret; 718 } 719 } 720 721 /* Apply the iQMF synthesis filter. */ 722 for (i = 0; i < avctx->channels; i++) { 723 float *p1 = out_samples[i]; 724 float *p2 = p1 + 256; 725 float *p3 = p2 + 256; 726 float *p4 = p3 + 256; 727 ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf); 728 ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf); 729 ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf); 730 } 731 732 return 0; 733} 734 735static int atrac3_decode_frame(AVCodecContext *avctx, void *data, 736 int *got_frame_ptr, AVPacket *avpkt) 737{ 738 AVFrame *frame = data; 739 const uint8_t *buf = avpkt->data; 740 int buf_size = avpkt->size; 741 ATRAC3Context *q = avctx->priv_data; 742 int ret; 743 const uint8_t *databuf; 744 745 if (buf_size < avctx->block_align) { 746 av_log(avctx, AV_LOG_ERROR, 747 "Frame too small (%d bytes). Truncated file?\n", buf_size); 748 return AVERROR_INVALIDDATA; 749 } 750 751 /* get output buffer */ 752 frame->nb_samples = SAMPLES_PER_FRAME; 753 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) 754 return ret; 755 756 /* Check if we need to descramble and what buffer to pass on. */ 757 if (q->scrambled_stream) { 758 decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align); 759 databuf = q->decoded_bytes_buffer; 760 } else { 761 databuf = buf; 762 } 763 764 ret = decode_frame(avctx, databuf, (float **)frame->extended_data); 765 if (ret) { 766 av_log(NULL, AV_LOG_ERROR, "Frame decoding error!\n"); 767 return ret; 768 } 769 770 *got_frame_ptr = 1; 771 772 return avctx->block_align; 773} 774 775static av_cold void atrac3_init_static_data(void) 776{ 777 int i; 778 779 init_imdct_window(); 780 ff_atrac_generate_tables(); 781 782 /* Initialize the VLC tables. */ 783 for (i = 0; i < 7; i++) { 784 spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]]; 785 spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] - 786 atrac3_vlc_offs[i ]; 787 init_vlc(&spectral_coeff_tab[i], 9, huff_tab_sizes[i], 788 huff_bits[i], 1, 1, 789 huff_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC); 790 } 791} 792 793static av_cold int atrac3_decode_init(AVCodecContext *avctx) 794{ 795 static int static_init_done; 796 int i, ret; 797 int version, delay, samples_per_frame, frame_factor; 798 const uint8_t *edata_ptr = avctx->extradata; 799 ATRAC3Context *q = avctx->priv_data; 800 801 if (avctx->channels <= 0 || avctx->channels > 2) { 802 av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n"); 803 return AVERROR(EINVAL); 804 } 805 806 if (!static_init_done) 807 atrac3_init_static_data(); 808 static_init_done = 1; 809 810 /* Take care of the codec-specific extradata. */ 811 if (avctx->extradata_size == 14) { 812 /* Parse the extradata, WAV format */ 813 av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n", 814 bytestream_get_le16(&edata_ptr)); // Unknown value always 1 815 edata_ptr += 4; // samples per channel 816 q->coding_mode = bytestream_get_le16(&edata_ptr); 817 av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n", 818 bytestream_get_le16(&edata_ptr)); //Dupe of coding mode 819 frame_factor = bytestream_get_le16(&edata_ptr); // Unknown always 1 820 av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n", 821 bytestream_get_le16(&edata_ptr)); // Unknown always 0 822 823 /* setup */ 824 samples_per_frame = SAMPLES_PER_FRAME * avctx->channels; 825 version = 4; 826 delay = 0x88E; 827 q->coding_mode = q->coding_mode ? JOINT_STEREO : STEREO; 828 q->scrambled_stream = 0; 829 830 if (avctx->block_align != 96 * avctx->channels * frame_factor && 831 avctx->block_align != 152 * avctx->channels * frame_factor && 832 avctx->block_align != 192 * avctx->channels * frame_factor) { 833 av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor " 834 "configuration %d/%d/%d\n", avctx->block_align, 835 avctx->channels, frame_factor); 836 return AVERROR_INVALIDDATA; 837 } 838 } else if (avctx->extradata_size == 12 || avctx->extradata_size == 10) { 839 /* Parse the extradata, RM format. */ 840 version = bytestream_get_be32(&edata_ptr); 841 samples_per_frame = bytestream_get_be16(&edata_ptr); 842 delay = bytestream_get_be16(&edata_ptr); 843 q->coding_mode = bytestream_get_be16(&edata_ptr); 844 q->scrambled_stream = 1; 845 846 } else { 847 av_log(NULL, AV_LOG_ERROR, "Unknown extradata size %d.\n", 848 avctx->extradata_size); 849 return AVERROR(EINVAL); 850 } 851 852 /* Check the extradata */ 853 854 if (version != 4) { 855 av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version); 856 return AVERROR_INVALIDDATA; 857 } 858 859 if (samples_per_frame != SAMPLES_PER_FRAME && 860 samples_per_frame != SAMPLES_PER_FRAME * 2) { 861 av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n", 862 samples_per_frame); 863 return AVERROR_INVALIDDATA; 864 } 865 866 if (delay != 0x88E) { 867 av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n", 868 delay); 869 return AVERROR_INVALIDDATA; 870 } 871 872 if (q->coding_mode == STEREO) 873 av_log(avctx, AV_LOG_DEBUG, "Normal stereo detected.\n"); 874 else if (q->coding_mode == JOINT_STEREO) { 875 if (avctx->channels != 2) { 876 av_log(avctx, AV_LOG_ERROR, "Invalid coding mode\n"); 877 return AVERROR_INVALIDDATA; 878 } 879 av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n"); 880 } else { 881 av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n", 882 q->coding_mode); 883 return AVERROR_INVALIDDATA; 884 } 885 886 if (avctx->block_align >= UINT_MAX / 2) 887 return AVERROR(EINVAL); 888 889 q->decoded_bytes_buffer = av_mallocz(FFALIGN(avctx->block_align, 4) + 890 FF_INPUT_BUFFER_PADDING_SIZE); 891 if (q->decoded_bytes_buffer == NULL) 892 return AVERROR(ENOMEM); 893 894 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; 895 896 /* initialize the MDCT transform */ 897 if ((ret = ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768)) < 0) { 898 av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n"); 899 av_freep(&q->decoded_bytes_buffer); 900 return ret; 901 } 902 903 /* init the joint-stereo decoding data */ 904 q->weighting_delay[0] = 0; 905 q->weighting_delay[1] = 7; 906 q->weighting_delay[2] = 0; 907 q->weighting_delay[3] = 7; 908 q->weighting_delay[4] = 0; 909 q->weighting_delay[5] = 7; 910 911 for (i = 0; i < 4; i++) { 912 q->matrix_coeff_index_prev[i] = 3; 913 q->matrix_coeff_index_now[i] = 3; 914 q->matrix_coeff_index_next[i] = 3; 915 } 916 917 ff_atrac_init_gain_compensation(&q->gainc_ctx, 4, 3); 918 avpriv_float_dsp_init(&q->fdsp, avctx->flags & CODEC_FLAG_BITEXACT); 919 ff_fmt_convert_init(&q->fmt_conv, avctx); 920 921 q->units = av_mallocz_array(avctx->channels, sizeof(*q->units)); 922 if (!q->units) { 923 atrac3_decode_close(avctx); 924 return AVERROR(ENOMEM); 925 } 926 927 return 0; 928} 929 930AVCodec ff_atrac3_decoder = { 931 .name = "atrac3", 932 .long_name = NULL_IF_CONFIG_SMALL("ATRAC3 (Adaptive TRansform Acoustic Coding 3)"), 933 .type = AVMEDIA_TYPE_AUDIO, 934 .id = AV_CODEC_ID_ATRAC3, 935 .priv_data_size = sizeof(ATRAC3Context), 936 .init = atrac3_decode_init, 937 .close = atrac3_decode_close, 938 .decode = atrac3_decode_frame, 939 .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1, 940 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, 941 AV_SAMPLE_FMT_NONE }, 942}; 943