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 "libavutil/mem_internal.h" 43#include "libavutil/thread.h" 44 45#include "avcodec.h" 46#include "bytestream.h" 47#include "codec_internal.h" 48#include "fft.h" 49#include "get_bits.h" 50#include "internal.h" 51 52#include "atrac.h" 53#include "atrac3data.h" 54 55#define MIN_CHANNELS 1 56#define MAX_CHANNELS 8 57#define MAX_JS_PAIRS 8 / 2 58 59#define JOINT_STEREO 0x12 60#define SINGLE 0x2 61 62#define SAMPLES_PER_FRAME 1024 63#define MDCT_SIZE 512 64 65#define ATRAC3_VLC_BITS 8 66 67typedef struct GainBlock { 68 AtracGainInfo g_block[4]; 69} GainBlock; 70 71typedef struct TonalComponent { 72 int pos; 73 int num_coefs; 74 float coef[8]; 75} TonalComponent; 76 77typedef struct ChannelUnit { 78 int bands_coded; 79 int num_components; 80 float prev_frame[SAMPLES_PER_FRAME]; 81 int gc_blk_switch; 82 TonalComponent components[64]; 83 GainBlock gain_block[2]; 84 85 DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME]; 86 DECLARE_ALIGNED(32, float, imdct_buf)[SAMPLES_PER_FRAME]; 87 88 float delay_buf1[46]; ///<qmf delay buffers 89 float delay_buf2[46]; 90 float delay_buf3[46]; 91} ChannelUnit; 92 93typedef struct ATRAC3Context { 94 GetBitContext gb; 95 //@{ 96 /** stream data */ 97 int coding_mode; 98 99 ChannelUnit *units; 100 //@} 101 //@{ 102 /** joint-stereo related variables */ 103 int matrix_coeff_index_prev[MAX_JS_PAIRS][4]; 104 int matrix_coeff_index_now[MAX_JS_PAIRS][4]; 105 int matrix_coeff_index_next[MAX_JS_PAIRS][4]; 106 int weighting_delay[MAX_JS_PAIRS][6]; 107 //@} 108 //@{ 109 /** data buffers */ 110 uint8_t *decoded_bytes_buffer; 111 float temp_buf[1070]; 112 //@} 113 //@{ 114 /** extradata */ 115 int scrambled_stream; 116 //@} 117 118 AtracGCContext gainc_ctx; 119 FFTContext mdct_ctx; 120 void (*vector_fmul)(float *dst, const float *src0, const float *src1, 121 int len); 122} ATRAC3Context; 123 124static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE]; 125static VLCElem atrac3_vlc_table[7 * 1 << ATRAC3_VLC_BITS]; 126static VLC spectral_coeff_tab[7]; 127 128/** 129 * Regular 512 points IMDCT without overlapping, with the exception of the 130 * swapping of odd bands caused by the reverse spectra of the QMF. 131 * 132 * @param odd_band 1 if the band is an odd band 133 */ 134static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band) 135{ 136 int i; 137 138 if (odd_band) { 139 /** 140 * Reverse the odd bands before IMDCT, this is an effect of the QMF 141 * transform or it gives better compression to do it this way. 142 * FIXME: It should be possible to handle this in imdct_calc 143 * for that to happen a modification of the prerotation step of 144 * all SIMD code and C code is needed. 145 * Or fix the functions before so they generate a pre reversed spectrum. 146 */ 147 for (i = 0; i < 128; i++) 148 FFSWAP(float, input[i], input[255 - i]); 149 } 150 151 q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input); 152 153 /* Perform windowing on the output. */ 154 q->vector_fmul(output, output, mdct_window, MDCT_SIZE); 155} 156 157/* 158 * indata descrambling, only used for data coming from the rm container 159 */ 160static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes) 161{ 162 int i, off; 163 uint32_t c; 164 const uint32_t *buf; 165 uint32_t *output = (uint32_t *)out; 166 167 off = (intptr_t)input & 3; 168 buf = (const uint32_t *)(input - off); 169 if (off) 170 c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8)))); 171 else 172 c = av_be2ne32(0x537F6103U); 173 bytes += 3 + off; 174 for (i = 0; i < bytes / 4; i++) 175 output[i] = c ^ buf[i]; 176 177 if (off) 178 avpriv_request_sample(NULL, "Offset of %d", off); 179 180 return off; 181} 182 183static av_cold void init_imdct_window(void) 184{ 185 int i, j; 186 187 /* generate the mdct window, for details see 188 * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */ 189 for (i = 0, j = 255; i < 128; i++, j--) { 190 float wi = sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0; 191 float wj = sin(((j + 0.5) / 256.0 - 0.5) * M_PI) + 1.0; 192 float w = 0.5 * (wi * wi + wj * wj); 193 mdct_window[i] = mdct_window[511 - i] = wi / w; 194 mdct_window[j] = mdct_window[511 - j] = wj / w; 195 } 196} 197 198static av_cold int atrac3_decode_close(AVCodecContext *avctx) 199{ 200 ATRAC3Context *q = avctx->priv_data; 201 202 av_freep(&q->units); 203 av_freep(&q->decoded_bytes_buffer); 204 205 ff_mdct_end(&q->mdct_ctx); 206 207 return 0; 208} 209 210/** 211 * Mantissa decoding 212 * 213 * @param selector which table the output values are coded with 214 * @param coding_flag constant length coding or variable length coding 215 * @param mantissas mantissa output table 216 * @param num_codes number of values to get 217 */ 218static void read_quant_spectral_coeffs(GetBitContext *gb, int selector, 219 int coding_flag, int *mantissas, 220 int num_codes) 221{ 222 int i, code, huff_symb; 223 224 if (selector == 1) 225 num_codes /= 2; 226 227 if (coding_flag != 0) { 228 /* constant length coding (CLC) */ 229 int num_bits = clc_length_tab[selector]; 230 231 if (selector > 1) { 232 for (i = 0; i < num_codes; i++) { 233 if (num_bits) 234 code = get_sbits(gb, num_bits); 235 else 236 code = 0; 237 mantissas[i] = code; 238 } 239 } else { 240 for (i = 0; i < num_codes; i++) { 241 if (num_bits) 242 code = get_bits(gb, num_bits); // num_bits is always 4 in this case 243 else 244 code = 0; 245 mantissas[i * 2 ] = mantissa_clc_tab[code >> 2]; 246 mantissas[i * 2 + 1] = mantissa_clc_tab[code & 3]; 247 } 248 } 249 } else { 250 /* variable length coding (VLC) */ 251 if (selector != 1) { 252 for (i = 0; i < num_codes; i++) { 253 mantissas[i] = get_vlc2(gb, spectral_coeff_tab[selector-1].table, 254 ATRAC3_VLC_BITS, 1); 255 } 256 } else { 257 for (i = 0; i < num_codes; i++) { 258 huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table, 259 ATRAC3_VLC_BITS, 1); 260 mantissas[i * 2 ] = mantissa_vlc_tab[huff_symb * 2 ]; 261 mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1]; 262 } 263 } 264 } 265} 266 267/** 268 * Restore the quantized band spectrum coefficients 269 * 270 * @return subband count, fix for broken specification/files 271 */ 272static int decode_spectrum(GetBitContext *gb, float *output) 273{ 274 int num_subbands, coding_mode, i, j, first, last, subband_size; 275 int subband_vlc_index[32], sf_index[32]; 276 int mantissas[128]; 277 float scale_factor; 278 279 num_subbands = get_bits(gb, 5); // number of coded subbands 280 coding_mode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC 281 282 /* get the VLC selector table for the subbands, 0 means not coded */ 283 for (i = 0; i <= num_subbands; i++) 284 subband_vlc_index[i] = get_bits(gb, 3); 285 286 /* read the scale factor indexes from the stream */ 287 for (i = 0; i <= num_subbands; i++) { 288 if (subband_vlc_index[i] != 0) 289 sf_index[i] = get_bits(gb, 6); 290 } 291 292 for (i = 0; i <= num_subbands; i++) { 293 first = subband_tab[i ]; 294 last = subband_tab[i + 1]; 295 296 subband_size = last - first; 297 298 if (subband_vlc_index[i] != 0) { 299 /* decode spectral coefficients for this subband */ 300 /* TODO: This can be done faster is several blocks share the 301 * same VLC selector (subband_vlc_index) */ 302 read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode, 303 mantissas, subband_size); 304 305 /* decode the scale factor for this subband */ 306 scale_factor = ff_atrac_sf_table[sf_index[i]] * 307 inv_max_quant[subband_vlc_index[i]]; 308 309 /* inverse quantize the coefficients */ 310 for (j = 0; first < last; first++, j++) 311 output[first] = mantissas[j] * scale_factor; 312 } else { 313 /* this subband was not coded, so zero the entire subband */ 314 memset(output + first, 0, subband_size * sizeof(*output)); 315 } 316 } 317 318 /* clear the subbands that were not coded */ 319 first = subband_tab[i]; 320 memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(*output)); 321 return num_subbands; 322} 323 324/** 325 * Restore the quantized tonal components 326 * 327 * @param components tonal components 328 * @param num_bands number of coded bands 329 */ 330static int decode_tonal_components(GetBitContext *gb, 331 TonalComponent *components, int num_bands) 332{ 333 int i, b, c, m; 334 int nb_components, coding_mode_selector, coding_mode; 335 int band_flags[4], mantissa[8]; 336 int component_count = 0; 337 338 nb_components = get_bits(gb, 5); 339 340 /* no tonal components */ 341 if (nb_components == 0) 342 return 0; 343 344 coding_mode_selector = get_bits(gb, 2); 345 if (coding_mode_selector == 2) 346 return AVERROR_INVALIDDATA; 347 348 coding_mode = coding_mode_selector & 1; 349 350 for (i = 0; i < nb_components; i++) { 351 int coded_values_per_component, quant_step_index; 352 353 for (b = 0; b <= num_bands; b++) 354 band_flags[b] = get_bits1(gb); 355 356 coded_values_per_component = get_bits(gb, 3); 357 358 quant_step_index = get_bits(gb, 3); 359 if (quant_step_index <= 1) 360 return AVERROR_INVALIDDATA; 361 362 if (coding_mode_selector == 3) 363 coding_mode = get_bits1(gb); 364 365 for (b = 0; b < (num_bands + 1) * 4; b++) { 366 int coded_components; 367 368 if (band_flags[b >> 2] == 0) 369 continue; 370 371 coded_components = get_bits(gb, 3); 372 373 for (c = 0; c < coded_components; c++) { 374 TonalComponent *cmp = &components[component_count]; 375 int sf_index, coded_values, max_coded_values; 376 float scale_factor; 377 378 sf_index = get_bits(gb, 6); 379 if (component_count >= 64) 380 return AVERROR_INVALIDDATA; 381 382 cmp->pos = b * 64 + get_bits(gb, 6); 383 384 max_coded_values = SAMPLES_PER_FRAME - cmp->pos; 385 coded_values = coded_values_per_component + 1; 386 coded_values = FFMIN(max_coded_values, coded_values); 387 388 scale_factor = ff_atrac_sf_table[sf_index] * 389 inv_max_quant[quant_step_index]; 390 391 read_quant_spectral_coeffs(gb, quant_step_index, coding_mode, 392 mantissa, coded_values); 393 394 cmp->num_coefs = coded_values; 395 396 /* inverse quant */ 397 for (m = 0; m < coded_values; m++) 398 cmp->coef[m] = mantissa[m] * scale_factor; 399 400 component_count++; 401 } 402 } 403 } 404 405 return component_count; 406} 407 408/** 409 * Decode gain parameters for the coded bands 410 * 411 * @param block the gainblock for the current band 412 * @param num_bands amount of coded bands 413 */ 414static int decode_gain_control(GetBitContext *gb, GainBlock *block, 415 int num_bands) 416{ 417 int b, j; 418 int *level, *loc; 419 420 AtracGainInfo *gain = block->g_block; 421 422 for (b = 0; b <= num_bands; b++) { 423 gain[b].num_points = get_bits(gb, 3); 424 level = gain[b].lev_code; 425 loc = gain[b].loc_code; 426 427 for (j = 0; j < gain[b].num_points; j++) { 428 level[j] = get_bits(gb, 4); 429 loc[j] = get_bits(gb, 5); 430 if (j && loc[j] <= loc[j - 1]) 431 return AVERROR_INVALIDDATA; 432 } 433 } 434 435 /* Clear the unused blocks. */ 436 for (; b < 4 ; b++) 437 gain[b].num_points = 0; 438 439 return 0; 440} 441 442/** 443 * Combine the tonal band spectrum and regular band spectrum 444 * 445 * @param spectrum output spectrum buffer 446 * @param num_components number of tonal components 447 * @param components tonal components for this band 448 * @return position of the last tonal coefficient 449 */ 450static int add_tonal_components(float *spectrum, int num_components, 451 TonalComponent *components) 452{ 453 int i, j, last_pos = -1; 454 float *input, *output; 455 456 for (i = 0; i < num_components; i++) { 457 last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos); 458 input = components[i].coef; 459 output = &spectrum[components[i].pos]; 460 461 for (j = 0; j < components[i].num_coefs; j++) 462 output[j] += input[j]; 463 } 464 465 return last_pos; 466} 467 468#define INTERPOLATE(old, new, nsample) \ 469 ((old) + (nsample) * 0.125 * ((new) - (old))) 470 471static void reverse_matrixing(float *su1, float *su2, int *prev_code, 472 int *curr_code) 473{ 474 int i, nsample, band; 475 float mc1_l, mc1_r, mc2_l, mc2_r; 476 477 for (i = 0, band = 0; band < 4 * 256; band += 256, i++) { 478 int s1 = prev_code[i]; 479 int s2 = curr_code[i]; 480 nsample = band; 481 482 if (s1 != s2) { 483 /* Selector value changed, interpolation needed. */ 484 mc1_l = matrix_coeffs[s1 * 2 ]; 485 mc1_r = matrix_coeffs[s1 * 2 + 1]; 486 mc2_l = matrix_coeffs[s2 * 2 ]; 487 mc2_r = matrix_coeffs[s2 * 2 + 1]; 488 489 /* Interpolation is done over the first eight samples. */ 490 for (; nsample < band + 8; nsample++) { 491 float c1 = su1[nsample]; 492 float c2 = su2[nsample]; 493 c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample - band) + 494 c2 * INTERPOLATE(mc1_r, mc2_r, nsample - band); 495 su1[nsample] = c2; 496 su2[nsample] = c1 * 2.0 - c2; 497 } 498 } 499 500 /* Apply the matrix without interpolation. */ 501 switch (s2) { 502 case 0: /* M/S decoding */ 503 for (; nsample < band + 256; nsample++) { 504 float c1 = su1[nsample]; 505 float c2 = su2[nsample]; 506 su1[nsample] = c2 * 2.0; 507 su2[nsample] = (c1 - c2) * 2.0; 508 } 509 break; 510 case 1: 511 for (; nsample < band + 256; nsample++) { 512 float c1 = su1[nsample]; 513 float c2 = su2[nsample]; 514 su1[nsample] = (c1 + c2) * 2.0; 515 su2[nsample] = c2 * -2.0; 516 } 517 break; 518 case 2: 519 case 3: 520 for (; nsample < band + 256; nsample++) { 521 float c1 = su1[nsample]; 522 float c2 = su2[nsample]; 523 su1[nsample] = c1 + c2; 524 su2[nsample] = c1 - c2; 525 } 526 break; 527 default: 528 av_assert1(0); 529 } 530 } 531} 532 533static void get_channel_weights(int index, int flag, float ch[2]) 534{ 535 if (index == 7) { 536 ch[0] = 1.0; 537 ch[1] = 1.0; 538 } else { 539 ch[0] = (index & 7) / 7.0; 540 ch[1] = sqrt(2 - ch[0] * ch[0]); 541 if (flag) 542 FFSWAP(float, ch[0], ch[1]); 543 } 544} 545 546static void channel_weighting(float *su1, float *su2, int *p3) 547{ 548 int band, nsample; 549 /* w[x][y] y=0 is left y=1 is right */ 550 float w[2][2]; 551 552 if (p3[1] != 7 || p3[3] != 7) { 553 get_channel_weights(p3[1], p3[0], w[0]); 554 get_channel_weights(p3[3], p3[2], w[1]); 555 556 for (band = 256; band < 4 * 256; band += 256) { 557 for (nsample = band; nsample < band + 8; nsample++) { 558 su1[nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample - band); 559 su2[nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample - band); 560 } 561 for(; nsample < band + 256; nsample++) { 562 su1[nsample] *= w[1][0]; 563 su2[nsample] *= w[1][1]; 564 } 565 } 566 } 567} 568 569/** 570 * Decode a Sound Unit 571 * 572 * @param snd the channel unit to be used 573 * @param output the decoded samples before IQMF in float representation 574 * @param channel_num channel number 575 * @param coding_mode the coding mode (JOINT_STEREO or single channels) 576 */ 577static int decode_channel_sound_unit(ATRAC3Context *q, GetBitContext *gb, 578 ChannelUnit *snd, float *output, 579 int channel_num, int coding_mode) 580{ 581 int band, ret, num_subbands, last_tonal, num_bands; 582 GainBlock *gain1 = &snd->gain_block[ snd->gc_blk_switch]; 583 GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch]; 584 585 if (coding_mode == JOINT_STEREO && (channel_num % 2) == 1) { 586 if (get_bits(gb, 2) != 3) { 587 av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n"); 588 return AVERROR_INVALIDDATA; 589 } 590 } else { 591 if (get_bits(gb, 6) != 0x28) { 592 av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n"); 593 return AVERROR_INVALIDDATA; 594 } 595 } 596 597 /* number of coded QMF bands */ 598 snd->bands_coded = get_bits(gb, 2); 599 600 ret = decode_gain_control(gb, gain2, snd->bands_coded); 601 if (ret) 602 return ret; 603 604 snd->num_components = decode_tonal_components(gb, snd->components, 605 snd->bands_coded); 606 if (snd->num_components < 0) 607 return snd->num_components; 608 609 num_subbands = decode_spectrum(gb, snd->spectrum); 610 611 /* Merge the decoded spectrum and tonal components. */ 612 last_tonal = add_tonal_components(snd->spectrum, snd->num_components, 613 snd->components); 614 615 616 /* calculate number of used MLT/QMF bands according to the amount of coded 617 spectral lines */ 618 num_bands = (subband_tab[num_subbands] - 1) >> 8; 619 if (last_tonal >= 0) 620 num_bands = FFMAX((last_tonal + 256) >> 8, num_bands); 621 622 623 /* Reconstruct time domain samples. */ 624 for (band = 0; band < 4; band++) { 625 /* Perform the IMDCT step without overlapping. */ 626 if (band <= num_bands) 627 imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1); 628 else 629 memset(snd->imdct_buf, 0, 512 * sizeof(*snd->imdct_buf)); 630 631 /* gain compensation and overlapping */ 632 ff_atrac_gain_compensation(&q->gainc_ctx, snd->imdct_buf, 633 &snd->prev_frame[band * 256], 634 &gain1->g_block[band], &gain2->g_block[band], 635 256, &output[band * 256]); 636 } 637 638 /* Swap the gain control buffers for the next frame. */ 639 snd->gc_blk_switch ^= 1; 640 641 return 0; 642} 643 644static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf, 645 float **out_samples) 646{ 647 ATRAC3Context *q = avctx->priv_data; 648 int ret, i, ch; 649 uint8_t *ptr1; 650 int channels = avctx->ch_layout.nb_channels; 651 652 if (q->coding_mode == JOINT_STEREO) { 653 /* channel coupling mode */ 654 655 /* Decode sound unit pairs (channels are expected to be even). 656 * Multichannel joint stereo interleaves pairs (6ch: 2ch + 2ch + 2ch) */ 657 const uint8_t *js_databuf; 658 int js_pair, js_block_align; 659 660 js_block_align = (avctx->block_align / channels) * 2; /* block pair */ 661 662 for (ch = 0; ch < channels; ch = ch + 2) { 663 js_pair = ch/2; 664 js_databuf = databuf + js_pair * js_block_align; /* align to current pair */ 665 666 /* Set the bitstream reader at the start of first channel sound unit. */ 667 init_get_bits(&q->gb, 668 js_databuf, js_block_align * 8); 669 670 /* decode Sound Unit 1 */ 671 ret = decode_channel_sound_unit(q, &q->gb, &q->units[ch], 672 out_samples[ch], ch, JOINT_STEREO); 673 if (ret != 0) 674 return ret; 675 676 /* Framedata of the su2 in the joint-stereo mode is encoded in 677 * reverse byte order so we need to swap it first. */ 678 if (js_databuf == q->decoded_bytes_buffer) { 679 uint8_t *ptr2 = q->decoded_bytes_buffer + js_block_align - 1; 680 ptr1 = q->decoded_bytes_buffer; 681 for (i = 0; i < js_block_align / 2; i++, ptr1++, ptr2--) 682 FFSWAP(uint8_t, *ptr1, *ptr2); 683 } else { 684 const uint8_t *ptr2 = js_databuf + js_block_align - 1; 685 for (i = 0; i < js_block_align; i++) 686 q->decoded_bytes_buffer[i] = *ptr2--; 687 } 688 689 /* Skip the sync codes (0xF8). */ 690 ptr1 = q->decoded_bytes_buffer; 691 for (i = 4; *ptr1 == 0xF8; i++, ptr1++) { 692 if (i >= js_block_align) 693 return AVERROR_INVALIDDATA; 694 } 695 696 697 /* set the bitstream reader at the start of the second Sound Unit */ 698 ret = init_get_bits8(&q->gb, 699 ptr1, q->decoded_bytes_buffer + js_block_align - ptr1); 700 if (ret < 0) 701 return ret; 702 703 /* Fill the Weighting coeffs delay buffer */ 704 memmove(q->weighting_delay[js_pair], &q->weighting_delay[js_pair][2], 705 4 * sizeof(*q->weighting_delay[js_pair])); 706 q->weighting_delay[js_pair][4] = get_bits1(&q->gb); 707 q->weighting_delay[js_pair][5] = get_bits(&q->gb, 3); 708 709 for (i = 0; i < 4; i++) { 710 q->matrix_coeff_index_prev[js_pair][i] = q->matrix_coeff_index_now[js_pair][i]; 711 q->matrix_coeff_index_now[js_pair][i] = q->matrix_coeff_index_next[js_pair][i]; 712 q->matrix_coeff_index_next[js_pair][i] = get_bits(&q->gb, 2); 713 } 714 715 /* Decode Sound Unit 2. */ 716 ret = decode_channel_sound_unit(q, &q->gb, &q->units[ch+1], 717 out_samples[ch+1], ch+1, JOINT_STEREO); 718 if (ret != 0) 719 return ret; 720 721 /* Reconstruct the channel coefficients. */ 722 reverse_matrixing(out_samples[ch], out_samples[ch+1], 723 q->matrix_coeff_index_prev[js_pair], 724 q->matrix_coeff_index_now[js_pair]); 725 726 channel_weighting(out_samples[ch], out_samples[ch+1], q->weighting_delay[js_pair]); 727 } 728 } else { 729 /* single channels */ 730 /* Decode the channel sound units. */ 731 for (i = 0; i < channels; i++) { 732 /* Set the bitstream reader at the start of a channel sound unit. */ 733 init_get_bits(&q->gb, 734 databuf + i * avctx->block_align / channels, 735 avctx->block_align * 8 / channels); 736 737 ret = decode_channel_sound_unit(q, &q->gb, &q->units[i], 738 out_samples[i], i, q->coding_mode); 739 if (ret != 0) 740 return ret; 741 } 742 } 743 744 /* Apply the iQMF synthesis filter. */ 745 for (i = 0; i < channels; i++) { 746 float *p1 = out_samples[i]; 747 float *p2 = p1 + 256; 748 float *p3 = p2 + 256; 749 float *p4 = p3 + 256; 750 ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf); 751 ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf); 752 ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf); 753 } 754 755 return 0; 756} 757 758static int al_decode_frame(AVCodecContext *avctx, const uint8_t *databuf, 759 int size, float **out_samples) 760{ 761 ATRAC3Context *q = avctx->priv_data; 762 int channels = avctx->ch_layout.nb_channels; 763 int ret, i; 764 765 /* Set the bitstream reader at the start of a channel sound unit. */ 766 init_get_bits(&q->gb, databuf, size * 8); 767 /* single channels */ 768 /* Decode the channel sound units. */ 769 for (i = 0; i < channels; i++) { 770 ret = decode_channel_sound_unit(q, &q->gb, &q->units[i], 771 out_samples[i], i, q->coding_mode); 772 if (ret != 0) 773 return ret; 774 while (i < channels && get_bits_left(&q->gb) > 6 && show_bits(&q->gb, 6) != 0x28) { 775 skip_bits(&q->gb, 1); 776 } 777 } 778 779 /* Apply the iQMF synthesis filter. */ 780 for (i = 0; i < channels; i++) { 781 float *p1 = out_samples[i]; 782 float *p2 = p1 + 256; 783 float *p3 = p2 + 256; 784 float *p4 = p3 + 256; 785 ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf); 786 ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf); 787 ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf); 788 } 789 790 return 0; 791} 792 793static int atrac3_decode_frame(AVCodecContext *avctx, AVFrame *frame, 794 int *got_frame_ptr, AVPacket *avpkt) 795{ 796 const uint8_t *buf = avpkt->data; 797 int buf_size = avpkt->size; 798 ATRAC3Context *q = avctx->priv_data; 799 int ret; 800 const uint8_t *databuf; 801 802 if (buf_size < avctx->block_align) { 803 av_log(avctx, AV_LOG_ERROR, 804 "Frame too small (%d bytes). Truncated file?\n", buf_size); 805 return AVERROR_INVALIDDATA; 806 } 807 808 /* get output buffer */ 809 frame->nb_samples = SAMPLES_PER_FRAME; 810 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) 811 return ret; 812 813 /* Check if we need to descramble and what buffer to pass on. */ 814 if (q->scrambled_stream) { 815 decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align); 816 databuf = q->decoded_bytes_buffer; 817 } else { 818 databuf = buf; 819 } 820 821 ret = decode_frame(avctx, databuf, (float **)frame->extended_data); 822 if (ret) { 823 av_log(avctx, AV_LOG_ERROR, "Frame decoding error!\n"); 824 return ret; 825 } 826 827 *got_frame_ptr = 1; 828 829 return avctx->block_align; 830} 831 832static int atrac3al_decode_frame(AVCodecContext *avctx, AVFrame *frame, 833 int *got_frame_ptr, AVPacket *avpkt) 834{ 835 int ret; 836 837 frame->nb_samples = SAMPLES_PER_FRAME; 838 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) 839 return ret; 840 841 ret = al_decode_frame(avctx, avpkt->data, avpkt->size, 842 (float **)frame->extended_data); 843 if (ret) { 844 av_log(avctx, AV_LOG_ERROR, "Frame decoding error!\n"); 845 return ret; 846 } 847 848 *got_frame_ptr = 1; 849 850 return avpkt->size; 851} 852 853static av_cold void atrac3_init_static_data(void) 854{ 855 VLCElem *table = atrac3_vlc_table; 856 const uint8_t (*hufftabs)[2] = atrac3_hufftabs; 857 int i; 858 859 init_imdct_window(); 860 ff_atrac_generate_tables(); 861 862 /* Initialize the VLC tables. */ 863 for (i = 0; i < 7; i++) { 864 spectral_coeff_tab[i].table = table; 865 spectral_coeff_tab[i].table_allocated = 256; 866 ff_init_vlc_from_lengths(&spectral_coeff_tab[i], ATRAC3_VLC_BITS, huff_tab_sizes[i], 867 &hufftabs[0][1], 2, 868 &hufftabs[0][0], 2, 1, 869 -31, INIT_VLC_USE_NEW_STATIC, NULL); 870 hufftabs += huff_tab_sizes[i]; 871 table += 256; 872 } 873} 874 875static av_cold int atrac3_decode_init(AVCodecContext *avctx) 876{ 877 static AVOnce init_static_once = AV_ONCE_INIT; 878 int i, js_pair, ret; 879 int version, delay, samples_per_frame, frame_factor; 880 const uint8_t *edata_ptr = avctx->extradata; 881 ATRAC3Context *q = avctx->priv_data; 882 AVFloatDSPContext *fdsp; 883 int channels = avctx->ch_layout.nb_channels; 884 885 if (channels < MIN_CHANNELS || channels > MAX_CHANNELS) { 886 av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n"); 887 return AVERROR(EINVAL); 888 } 889 890 /* Take care of the codec-specific extradata. */ 891 if (avctx->codec_id == AV_CODEC_ID_ATRAC3AL) { 892 version = 4; 893 samples_per_frame = SAMPLES_PER_FRAME * channels; 894 delay = 0x88E; 895 q->coding_mode = SINGLE; 896 } else if (avctx->extradata_size == 14) { 897 /* Parse the extradata, WAV format */ 898 av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n", 899 bytestream_get_le16(&edata_ptr)); // Unknown value always 1 900 edata_ptr += 4; // samples per channel 901 q->coding_mode = bytestream_get_le16(&edata_ptr); 902 av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n", 903 bytestream_get_le16(&edata_ptr)); //Dupe of coding mode 904 frame_factor = bytestream_get_le16(&edata_ptr); // Unknown always 1 905 av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n", 906 bytestream_get_le16(&edata_ptr)); // Unknown always 0 907 908 /* setup */ 909 samples_per_frame = SAMPLES_PER_FRAME * channels; 910 version = 4; 911 delay = 0x88E; 912 q->coding_mode = q->coding_mode ? JOINT_STEREO : SINGLE; 913 q->scrambled_stream = 0; 914 915 if (avctx->block_align != 96 * channels * frame_factor && 916 avctx->block_align != 152 * channels * frame_factor && 917 avctx->block_align != 192 * channels * frame_factor) { 918 av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor " 919 "configuration %d/%d/%d\n", avctx->block_align, 920 channels, frame_factor); 921 return AVERROR_INVALIDDATA; 922 } 923 } else if (avctx->extradata_size == 12 || avctx->extradata_size == 10) { 924 /* Parse the extradata, RM format. */ 925 version = bytestream_get_be32(&edata_ptr); 926 samples_per_frame = bytestream_get_be16(&edata_ptr); 927 delay = bytestream_get_be16(&edata_ptr); 928 q->coding_mode = bytestream_get_be16(&edata_ptr); 929 q->scrambled_stream = 1; 930 931 } else { 932 av_log(avctx, AV_LOG_ERROR, "Unknown extradata size %d.\n", 933 avctx->extradata_size); 934 return AVERROR(EINVAL); 935 } 936 937 /* Check the extradata */ 938 939 if (version != 4) { 940 av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version); 941 return AVERROR_INVALIDDATA; 942 } 943 944 if (samples_per_frame != SAMPLES_PER_FRAME * channels) { 945 av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n", 946 samples_per_frame); 947 return AVERROR_INVALIDDATA; 948 } 949 950 if (delay != 0x88E) { 951 av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n", 952 delay); 953 return AVERROR_INVALIDDATA; 954 } 955 956 if (q->coding_mode == SINGLE) 957 av_log(avctx, AV_LOG_DEBUG, "Single channels detected.\n"); 958 else if (q->coding_mode == JOINT_STEREO) { 959 if (channels % 2 == 1) { /* Joint stereo channels must be even */ 960 av_log(avctx, AV_LOG_ERROR, "Invalid joint stereo channel configuration.\n"); 961 return AVERROR_INVALIDDATA; 962 } 963 av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n"); 964 } else { 965 av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n", 966 q->coding_mode); 967 return AVERROR_INVALIDDATA; 968 } 969 970 if (avctx->block_align > 4096 || avctx->block_align <= 0) 971 return AVERROR(EINVAL); 972 973 q->decoded_bytes_buffer = av_mallocz(FFALIGN(avctx->block_align, 4) + 974 AV_INPUT_BUFFER_PADDING_SIZE); 975 if (!q->decoded_bytes_buffer) 976 return AVERROR(ENOMEM); 977 978 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; 979 980 /* initialize the MDCT transform */ 981 if ((ret = ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768)) < 0) { 982 av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n"); 983 return ret; 984 } 985 986 /* init the joint-stereo decoding data */ 987 for (js_pair = 0; js_pair < MAX_JS_PAIRS; js_pair++) { 988 q->weighting_delay[js_pair][0] = 0; 989 q->weighting_delay[js_pair][1] = 7; 990 q->weighting_delay[js_pair][2] = 0; 991 q->weighting_delay[js_pair][3] = 7; 992 q->weighting_delay[js_pair][4] = 0; 993 q->weighting_delay[js_pair][5] = 7; 994 995 for (i = 0; i < 4; i++) { 996 q->matrix_coeff_index_prev[js_pair][i] = 3; 997 q->matrix_coeff_index_now[js_pair][i] = 3; 998 q->matrix_coeff_index_next[js_pair][i] = 3; 999 } 1000 } 1001 1002 ff_atrac_init_gain_compensation(&q->gainc_ctx, 4, 3); 1003 fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT); 1004 if (!fdsp) 1005 return AVERROR(ENOMEM); 1006 q->vector_fmul = fdsp->vector_fmul; 1007 av_free(fdsp); 1008 1009 q->units = av_calloc(channels, sizeof(*q->units)); 1010 if (!q->units) 1011 return AVERROR(ENOMEM); 1012 1013 ff_thread_once(&init_static_once, atrac3_init_static_data); 1014 1015 return 0; 1016} 1017 1018const FFCodec ff_atrac3_decoder = { 1019 .p.name = "atrac3", 1020 .p.long_name = NULL_IF_CONFIG_SMALL("ATRAC3 (Adaptive TRansform Acoustic Coding 3)"), 1021 .p.type = AVMEDIA_TYPE_AUDIO, 1022 .p.id = AV_CODEC_ID_ATRAC3, 1023 .priv_data_size = sizeof(ATRAC3Context), 1024 .init = atrac3_decode_init, 1025 .close = atrac3_decode_close, 1026 FF_CODEC_DECODE_CB(atrac3_decode_frame), 1027 .p.capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1, 1028 .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, 1029 AV_SAMPLE_FMT_NONE }, 1030 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP, 1031}; 1032 1033const FFCodec ff_atrac3al_decoder = { 1034 .p.name = "atrac3al", 1035 .p.long_name = NULL_IF_CONFIG_SMALL("ATRAC3 AL (Adaptive TRansform Acoustic Coding 3 Advanced Lossless)"), 1036 .p.type = AVMEDIA_TYPE_AUDIO, 1037 .p.id = AV_CODEC_ID_ATRAC3AL, 1038 .priv_data_size = sizeof(ATRAC3Context), 1039 .init = atrac3_decode_init, 1040 .close = atrac3_decode_close, 1041 FF_CODEC_DECODE_CB(atrac3al_decode_frame), 1042 .p.capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1, 1043 .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, 1044 AV_SAMPLE_FMT_NONE }, 1045 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP, 1046}; 1047