1/* 2 * COOK compatible decoder 3 * Copyright (c) 2003 Sascha Sommer 4 * Copyright (c) 2005 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 * Cook compatible decoder. Bastardization of the G.722.1 standard. 26 * This decoder handles RealNetworks, RealAudio G2 data. 27 * Cook is identified by the codec name cook in RM files. 28 * 29 * To use this decoder, a calling application must supply the extradata 30 * bytes provided from the RM container; 8+ bytes for mono streams and 31 * 16+ for stereo streams (maybe more). 32 * 33 * Codec technicalities (all this assume a buffer length of 1024): 34 * Cook works with several different techniques to achieve its compression. 35 * In the timedomain the buffer is divided into 8 pieces and quantized. If 36 * two neighboring pieces have different quantization index a smooth 37 * quantization curve is used to get a smooth overlap between the different 38 * pieces. 39 * To get to the transformdomain Cook uses a modulated lapped transform. 40 * The transform domain has 50 subbands with 20 elements each. This 41 * means only a maximum of 50*20=1000 coefficients are used out of the 1024 42 * available. 43 */ 44 45#include "libavutil/channel_layout.h" 46#include "libavutil/lfg.h" 47 48#include "audiodsp.h" 49#include "avcodec.h" 50#include "get_bits.h" 51#include "bytestream.h" 52#include "fft.h" 53#include "internal.h" 54#include "sinewin.h" 55#include "unary.h" 56 57#include "cookdata.h" 58 59/* the different Cook versions */ 60#define MONO 0x1000001 61#define STEREO 0x1000002 62#define JOINT_STEREO 0x1000003 63#define MC_COOK 0x2000000 // multichannel Cook, not supported 64 65#define SUBBAND_SIZE 20 66#define MAX_SUBPACKETS 5 67 68typedef struct { 69 int *now; 70 int *previous; 71} cook_gains; 72 73typedef struct { 74 int ch_idx; 75 int size; 76 int num_channels; 77 int cookversion; 78 int subbands; 79 int js_subband_start; 80 int js_vlc_bits; 81 int samples_per_channel; 82 int log2_numvector_size; 83 unsigned int channel_mask; 84 VLC channel_coupling; 85 int joint_stereo; 86 int bits_per_subpacket; 87 int bits_per_subpdiv; 88 int total_subbands; 89 int numvector_size; // 1 << log2_numvector_size; 90 91 float mono_previous_buffer1[1024]; 92 float mono_previous_buffer2[1024]; 93 94 cook_gains gains1; 95 cook_gains gains2; 96 int gain_1[9]; 97 int gain_2[9]; 98 int gain_3[9]; 99 int gain_4[9]; 100} COOKSubpacket; 101 102typedef struct cook { 103 /* 104 * The following 5 functions provide the lowlevel arithmetic on 105 * the internal audio buffers. 106 */ 107 void (*scalar_dequant)(struct cook *q, int index, int quant_index, 108 int *subband_coef_index, int *subband_coef_sign, 109 float *mlt_p); 110 111 void (*decouple)(struct cook *q, 112 COOKSubpacket *p, 113 int subband, 114 float f1, float f2, 115 float *decode_buffer, 116 float *mlt_buffer1, float *mlt_buffer2); 117 118 void (*imlt_window)(struct cook *q, float *buffer1, 119 cook_gains *gains_ptr, float *previous_buffer); 120 121 void (*interpolate)(struct cook *q, float *buffer, 122 int gain_index, int gain_index_next); 123 124 void (*saturate_output)(struct cook *q, float *out); 125 126 AVCodecContext* avctx; 127 AudioDSPContext adsp; 128 GetBitContext gb; 129 /* stream data */ 130 int num_vectors; 131 int samples_per_channel; 132 /* states */ 133 AVLFG random_state; 134 int discarded_packets; 135 136 /* transform data */ 137 FFTContext mdct_ctx; 138 float* mlt_window; 139 140 /* VLC data */ 141 VLC envelope_quant_index[13]; 142 VLC sqvh[7]; // scalar quantization 143 144 /* generatable tables and related variables */ 145 int gain_size_factor; 146 float gain_table[23]; 147 148 /* data buffers */ 149 150 uint8_t* decoded_bytes_buffer; 151 DECLARE_ALIGNED(32, float, mono_mdct_output)[2048]; 152 float decode_buffer_1[1024]; 153 float decode_buffer_2[1024]; 154 float decode_buffer_0[1060]; /* static allocation for joint decode */ 155 156 const float *cplscales[5]; 157 int num_subpackets; 158 COOKSubpacket subpacket[MAX_SUBPACKETS]; 159} COOKContext; 160 161static float pow2tab[127]; 162static float rootpow2tab[127]; 163 164/*************** init functions ***************/ 165 166/* table generator */ 167static av_cold void init_pow2table(void) 168{ 169 int i; 170 for (i = -63; i < 64; i++) { 171 pow2tab[63 + i] = pow(2, i); 172 rootpow2tab[63 + i] = sqrt(pow(2, i)); 173 } 174} 175 176/* table generator */ 177static av_cold void init_gain_table(COOKContext *q) 178{ 179 int i; 180 q->gain_size_factor = q->samples_per_channel / 8; 181 for (i = 0; i < 23; i++) 182 q->gain_table[i] = pow(pow2tab[i + 52], 183 (1.0 / (double) q->gain_size_factor)); 184} 185 186 187static av_cold int init_cook_vlc_tables(COOKContext *q) 188{ 189 int i, result; 190 191 result = 0; 192 for (i = 0; i < 13; i++) { 193 result |= init_vlc(&q->envelope_quant_index[i], 9, 24, 194 envelope_quant_index_huffbits[i], 1, 1, 195 envelope_quant_index_huffcodes[i], 2, 2, 0); 196 } 197 av_log(q->avctx, AV_LOG_DEBUG, "sqvh VLC init\n"); 198 for (i = 0; i < 7; i++) { 199 result |= init_vlc(&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i], 200 cvh_huffbits[i], 1, 1, 201 cvh_huffcodes[i], 2, 2, 0); 202 } 203 204 for (i = 0; i < q->num_subpackets; i++) { 205 if (q->subpacket[i].joint_stereo == 1) { 206 result |= init_vlc(&q->subpacket[i].channel_coupling, 6, 207 (1 << q->subpacket[i].js_vlc_bits) - 1, 208 ccpl_huffbits[q->subpacket[i].js_vlc_bits - 2], 1, 1, 209 ccpl_huffcodes[q->subpacket[i].js_vlc_bits - 2], 2, 2, 0); 210 av_log(q->avctx, AV_LOG_DEBUG, "subpacket %i Joint-stereo VLC used.\n", i); 211 } 212 } 213 214 av_log(q->avctx, AV_LOG_DEBUG, "VLC tables initialized.\n"); 215 return result; 216} 217 218static av_cold int init_cook_mlt(COOKContext *q) 219{ 220 int j, ret; 221 int mlt_size = q->samples_per_channel; 222 223 if ((q->mlt_window = av_malloc_array(mlt_size, sizeof(*q->mlt_window))) == 0) 224 return AVERROR(ENOMEM); 225 226 /* Initialize the MLT window: simple sine window. */ 227 ff_sine_window_init(q->mlt_window, mlt_size); 228 for (j = 0; j < mlt_size; j++) 229 q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel); 230 231 /* Initialize the MDCT. */ 232 if ((ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size) + 1, 1, 1.0 / 32768.0))) { 233 av_freep(&q->mlt_window); 234 return ret; 235 } 236 av_log(q->avctx, AV_LOG_DEBUG, "MDCT initialized, order = %d.\n", 237 av_log2(mlt_size) + 1); 238 239 return 0; 240} 241 242static av_cold void init_cplscales_table(COOKContext *q) 243{ 244 int i; 245 for (i = 0; i < 5; i++) 246 q->cplscales[i] = cplscales[i]; 247} 248 249/*************** init functions end ***********/ 250 251#define DECODE_BYTES_PAD1(bytes) (3 - ((bytes) + 3) % 4) 252#define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes))) 253 254/** 255 * Cook indata decoding, every 32 bits are XORed with 0x37c511f2. 256 * Why? No idea, some checksum/error detection method maybe. 257 * 258 * Out buffer size: extra bytes are needed to cope with 259 * padding/misalignment. 260 * Subpackets passed to the decoder can contain two, consecutive 261 * half-subpackets, of identical but arbitrary size. 262 * 1234 1234 1234 1234 extraA extraB 263 * Case 1: AAAA BBBB 0 0 264 * Case 2: AAAA ABBB BB-- 3 3 265 * Case 3: AAAA AABB BBBB 2 2 266 * Case 4: AAAA AAAB BBBB BB-- 1 5 267 * 268 * Nice way to waste CPU cycles. 269 * 270 * @param inbuffer pointer to byte array of indata 271 * @param out pointer to byte array of outdata 272 * @param bytes number of bytes 273 */ 274static inline int decode_bytes(const uint8_t *inbuffer, uint8_t *out, int bytes) 275{ 276 static const uint32_t tab[4] = { 277 AV_BE2NE32C(0x37c511f2u), AV_BE2NE32C(0xf237c511u), 278 AV_BE2NE32C(0x11f237c5u), AV_BE2NE32C(0xc511f237u), 279 }; 280 int i, off; 281 uint32_t c; 282 const uint32_t *buf; 283 uint32_t *obuf = (uint32_t *) out; 284 /* FIXME: 64 bit platforms would be able to do 64 bits at a time. 285 * I'm too lazy though, should be something like 286 * for (i = 0; i < bitamount / 64; i++) 287 * (int64_t) out[i] = 0x37c511f237c511f2 ^ av_be2ne64(int64_t) in[i]); 288 * Buffer alignment needs to be checked. */ 289 290 off = (intptr_t) inbuffer & 3; 291 buf = (const uint32_t *) (inbuffer - off); 292 c = tab[off]; 293 bytes += 3 + off; 294 for (i = 0; i < bytes / 4; i++) 295 obuf[i] = c ^ buf[i]; 296 297 return off; 298} 299 300static av_cold int cook_decode_close(AVCodecContext *avctx) 301{ 302 int i; 303 COOKContext *q = avctx->priv_data; 304 av_log(avctx, AV_LOG_DEBUG, "Deallocating memory.\n"); 305 306 /* Free allocated memory buffers. */ 307 av_freep(&q->mlt_window); 308 av_freep(&q->decoded_bytes_buffer); 309 310 /* Free the transform. */ 311 ff_mdct_end(&q->mdct_ctx); 312 313 /* Free the VLC tables. */ 314 for (i = 0; i < 13; i++) 315 ff_free_vlc(&q->envelope_quant_index[i]); 316 for (i = 0; i < 7; i++) 317 ff_free_vlc(&q->sqvh[i]); 318 for (i = 0; i < q->num_subpackets; i++) 319 ff_free_vlc(&q->subpacket[i].channel_coupling); 320 321 av_log(avctx, AV_LOG_DEBUG, "Memory deallocated.\n"); 322 323 return 0; 324} 325 326/** 327 * Fill the gain array for the timedomain quantization. 328 * 329 * @param gb pointer to the GetBitContext 330 * @param gaininfo array[9] of gain indexes 331 */ 332static void decode_gain_info(GetBitContext *gb, int *gaininfo) 333{ 334 int i, n; 335 336 n = get_unary(gb, 0, get_bits_left(gb)); // amount of elements*2 to update 337 338 i = 0; 339 while (n--) { 340 int index = get_bits(gb, 3); 341 int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1; 342 343 while (i <= index) 344 gaininfo[i++] = gain; 345 } 346 while (i <= 8) 347 gaininfo[i++] = 0; 348} 349 350/** 351 * Create the quant index table needed for the envelope. 352 * 353 * @param q pointer to the COOKContext 354 * @param quant_index_table pointer to the array 355 */ 356static int decode_envelope(COOKContext *q, COOKSubpacket *p, 357 int *quant_index_table) 358{ 359 int i, j, vlc_index; 360 361 quant_index_table[0] = get_bits(&q->gb, 6) - 6; // This is used later in categorize 362 363 for (i = 1; i < p->total_subbands; i++) { 364 vlc_index = i; 365 if (i >= p->js_subband_start * 2) { 366 vlc_index -= p->js_subband_start; 367 } else { 368 vlc_index /= 2; 369 if (vlc_index < 1) 370 vlc_index = 1; 371 } 372 if (vlc_index > 13) 373 vlc_index = 13; // the VLC tables >13 are identical to No. 13 374 375 j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index - 1].table, 376 q->envelope_quant_index[vlc_index - 1].bits, 2); 377 quant_index_table[i] = quant_index_table[i - 1] + j - 12; // differential encoding 378 if (quant_index_table[i] > 63 || quant_index_table[i] < -63) { 379 av_log(q->avctx, AV_LOG_ERROR, 380 "Invalid quantizer %d at position %d, outside [-63, 63] range\n", 381 quant_index_table[i], i); 382 return AVERROR_INVALIDDATA; 383 } 384 } 385 386 return 0; 387} 388 389/** 390 * Calculate the category and category_index vector. 391 * 392 * @param q pointer to the COOKContext 393 * @param quant_index_table pointer to the array 394 * @param category pointer to the category array 395 * @param category_index pointer to the category_index array 396 */ 397static void categorize(COOKContext *q, COOKSubpacket *p, const int *quant_index_table, 398 int *category, int *category_index) 399{ 400 int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j; 401 int exp_index2[102] = { 0 }; 402 int exp_index1[102] = { 0 }; 403 404 int tmp_categorize_array[128 * 2] = { 0 }; 405 int tmp_categorize_array1_idx = p->numvector_size; 406 int tmp_categorize_array2_idx = p->numvector_size; 407 408 bits_left = p->bits_per_subpacket - get_bits_count(&q->gb); 409 410 if (bits_left > q->samples_per_channel) 411 bits_left = q->samples_per_channel + 412 ((bits_left - q->samples_per_channel) * 5) / 8; 413 414 bias = -32; 415 416 /* Estimate bias. */ 417 for (i = 32; i > 0; i = i / 2) { 418 num_bits = 0; 419 index = 0; 420 for (j = p->total_subbands; j > 0; j--) { 421 exp_idx = av_clip((i - quant_index_table[index] + bias) / 2, 0, 7); 422 index++; 423 num_bits += expbits_tab[exp_idx]; 424 } 425 if (num_bits >= bits_left - 32) 426 bias += i; 427 } 428 429 /* Calculate total number of bits. */ 430 num_bits = 0; 431 for (i = 0; i < p->total_subbands; i++) { 432 exp_idx = av_clip((bias - quant_index_table[i]) / 2, 0, 7); 433 num_bits += expbits_tab[exp_idx]; 434 exp_index1[i] = exp_idx; 435 exp_index2[i] = exp_idx; 436 } 437 tmpbias1 = tmpbias2 = num_bits; 438 439 for (j = 1; j < p->numvector_size; j++) { 440 if (tmpbias1 + tmpbias2 > 2 * bits_left) { /* ---> */ 441 int max = -999999; 442 index = -1; 443 for (i = 0; i < p->total_subbands; i++) { 444 if (exp_index1[i] < 7) { 445 v = (-2 * exp_index1[i]) - quant_index_table[i] + bias; 446 if (v >= max) { 447 max = v; 448 index = i; 449 } 450 } 451 } 452 if (index == -1) 453 break; 454 tmp_categorize_array[tmp_categorize_array1_idx++] = index; 455 tmpbias1 -= expbits_tab[exp_index1[index]] - 456 expbits_tab[exp_index1[index] + 1]; 457 ++exp_index1[index]; 458 } else { /* <--- */ 459 int min = 999999; 460 index = -1; 461 for (i = 0; i < p->total_subbands; i++) { 462 if (exp_index2[i] > 0) { 463 v = (-2 * exp_index2[i]) - quant_index_table[i] + bias; 464 if (v < min) { 465 min = v; 466 index = i; 467 } 468 } 469 } 470 if (index == -1) 471 break; 472 tmp_categorize_array[--tmp_categorize_array2_idx] = index; 473 tmpbias2 -= expbits_tab[exp_index2[index]] - 474 expbits_tab[exp_index2[index] - 1]; 475 --exp_index2[index]; 476 } 477 } 478 479 for (i = 0; i < p->total_subbands; i++) 480 category[i] = exp_index2[i]; 481 482 for (i = 0; i < p->numvector_size - 1; i++) 483 category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++]; 484} 485 486 487/** 488 * Expand the category vector. 489 * 490 * @param q pointer to the COOKContext 491 * @param category pointer to the category array 492 * @param category_index pointer to the category_index array 493 */ 494static inline void expand_category(COOKContext *q, int *category, 495 int *category_index) 496{ 497 int i; 498 for (i = 0; i < q->num_vectors; i++) 499 { 500 int idx = category_index[i]; 501 if (++category[idx] >= FF_ARRAY_ELEMS(dither_tab)) 502 --category[idx]; 503 } 504} 505 506/** 507 * The real requantization of the mltcoefs 508 * 509 * @param q pointer to the COOKContext 510 * @param index index 511 * @param quant_index quantisation index 512 * @param subband_coef_index array of indexes to quant_centroid_tab 513 * @param subband_coef_sign signs of coefficients 514 * @param mlt_p pointer into the mlt buffer 515 */ 516static void scalar_dequant_float(COOKContext *q, int index, int quant_index, 517 int *subband_coef_index, int *subband_coef_sign, 518 float *mlt_p) 519{ 520 int i; 521 float f1; 522 523 for (i = 0; i < SUBBAND_SIZE; i++) { 524 if (subband_coef_index[i]) { 525 f1 = quant_centroid_tab[index][subband_coef_index[i]]; 526 if (subband_coef_sign[i]) 527 f1 = -f1; 528 } else { 529 /* noise coding if subband_coef_index[i] == 0 */ 530 f1 = dither_tab[index]; 531 if (av_lfg_get(&q->random_state) < 0x80000000) 532 f1 = -f1; 533 } 534 mlt_p[i] = f1 * rootpow2tab[quant_index + 63]; 535 } 536} 537/** 538 * Unpack the subband_coef_index and subband_coef_sign vectors. 539 * 540 * @param q pointer to the COOKContext 541 * @param category pointer to the category array 542 * @param subband_coef_index array of indexes to quant_centroid_tab 543 * @param subband_coef_sign signs of coefficients 544 */ 545static int unpack_SQVH(COOKContext *q, COOKSubpacket *p, int category, 546 int *subband_coef_index, int *subband_coef_sign) 547{ 548 int i, j; 549 int vlc, vd, tmp, result; 550 551 vd = vd_tab[category]; 552 result = 0; 553 for (i = 0; i < vpr_tab[category]; i++) { 554 vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3); 555 if (p->bits_per_subpacket < get_bits_count(&q->gb)) { 556 vlc = 0; 557 result = 1; 558 } 559 for (j = vd - 1; j >= 0; j--) { 560 tmp = (vlc * invradix_tab[category]) / 0x100000; 561 subband_coef_index[vd * i + j] = vlc - tmp * (kmax_tab[category] + 1); 562 vlc = tmp; 563 } 564 for (j = 0; j < vd; j++) { 565 if (subband_coef_index[i * vd + j]) { 566 if (get_bits_count(&q->gb) < p->bits_per_subpacket) { 567 subband_coef_sign[i * vd + j] = get_bits1(&q->gb); 568 } else { 569 result = 1; 570 subband_coef_sign[i * vd + j] = 0; 571 } 572 } else { 573 subband_coef_sign[i * vd + j] = 0; 574 } 575 } 576 } 577 return result; 578} 579 580 581/** 582 * Fill the mlt_buffer with mlt coefficients. 583 * 584 * @param q pointer to the COOKContext 585 * @param category pointer to the category array 586 * @param quant_index_table pointer to the array 587 * @param mlt_buffer pointer to mlt coefficients 588 */ 589static void decode_vectors(COOKContext *q, COOKSubpacket *p, int *category, 590 int *quant_index_table, float *mlt_buffer) 591{ 592 /* A zero in this table means that the subband coefficient is 593 random noise coded. */ 594 int subband_coef_index[SUBBAND_SIZE]; 595 /* A zero in this table means that the subband coefficient is a 596 positive multiplicator. */ 597 int subband_coef_sign[SUBBAND_SIZE]; 598 int band, j; 599 int index = 0; 600 601 for (band = 0; band < p->total_subbands; band++) { 602 index = category[band]; 603 if (category[band] < 7) { 604 if (unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)) { 605 index = 7; 606 for (j = 0; j < p->total_subbands; j++) 607 category[band + j] = 7; 608 } 609 } 610 if (index >= 7) { 611 memset(subband_coef_index, 0, sizeof(subband_coef_index)); 612 memset(subband_coef_sign, 0, sizeof(subband_coef_sign)); 613 } 614 q->scalar_dequant(q, index, quant_index_table[band], 615 subband_coef_index, subband_coef_sign, 616 &mlt_buffer[band * SUBBAND_SIZE]); 617 } 618 619 /* FIXME: should this be removed, or moved into loop above? */ 620 if (p->total_subbands * SUBBAND_SIZE >= q->samples_per_channel) 621 return; 622} 623 624 625static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer) 626{ 627 int category_index[128] = { 0 }; 628 int category[128] = { 0 }; 629 int quant_index_table[102]; 630 int res, i; 631 632 if ((res = decode_envelope(q, p, quant_index_table)) < 0) 633 return res; 634 q->num_vectors = get_bits(&q->gb, p->log2_numvector_size); 635 categorize(q, p, quant_index_table, category, category_index); 636 expand_category(q, category, category_index); 637 for (i=0; i<p->total_subbands; i++) { 638 if (category[i] > 7) 639 return AVERROR_INVALIDDATA; 640 } 641 decode_vectors(q, p, category, quant_index_table, mlt_buffer); 642 643 return 0; 644} 645 646 647/** 648 * the actual requantization of the timedomain samples 649 * 650 * @param q pointer to the COOKContext 651 * @param buffer pointer to the timedomain buffer 652 * @param gain_index index for the block multiplier 653 * @param gain_index_next index for the next block multiplier 654 */ 655static void interpolate_float(COOKContext *q, float *buffer, 656 int gain_index, int gain_index_next) 657{ 658 int i; 659 float fc1, fc2; 660 fc1 = pow2tab[gain_index + 63]; 661 662 if (gain_index == gain_index_next) { // static gain 663 for (i = 0; i < q->gain_size_factor; i++) 664 buffer[i] *= fc1; 665 } else { // smooth gain 666 fc2 = q->gain_table[11 + (gain_index_next - gain_index)]; 667 for (i = 0; i < q->gain_size_factor; i++) { 668 buffer[i] *= fc1; 669 fc1 *= fc2; 670 } 671 } 672} 673 674/** 675 * Apply transform window, overlap buffers. 676 * 677 * @param q pointer to the COOKContext 678 * @param inbuffer pointer to the mltcoefficients 679 * @param gains_ptr current and previous gains 680 * @param previous_buffer pointer to the previous buffer to be used for overlapping 681 */ 682static void imlt_window_float(COOKContext *q, float *inbuffer, 683 cook_gains *gains_ptr, float *previous_buffer) 684{ 685 const float fc = pow2tab[gains_ptr->previous[0] + 63]; 686 int i; 687 /* The weird thing here, is that the two halves of the time domain 688 * buffer are swapped. Also, the newest data, that we save away for 689 * next frame, has the wrong sign. Hence the subtraction below. 690 * Almost sounds like a complex conjugate/reverse data/FFT effect. 691 */ 692 693 /* Apply window and overlap */ 694 for (i = 0; i < q->samples_per_channel; i++) 695 inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] - 696 previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i]; 697} 698 699/** 700 * The modulated lapped transform, this takes transform coefficients 701 * and transforms them into timedomain samples. 702 * Apply transform window, overlap buffers, apply gain profile 703 * and buffer management. 704 * 705 * @param q pointer to the COOKContext 706 * @param inbuffer pointer to the mltcoefficients 707 * @param gains_ptr current and previous gains 708 * @param previous_buffer pointer to the previous buffer to be used for overlapping 709 */ 710static void imlt_gain(COOKContext *q, float *inbuffer, 711 cook_gains *gains_ptr, float *previous_buffer) 712{ 713 float *buffer0 = q->mono_mdct_output; 714 float *buffer1 = q->mono_mdct_output + q->samples_per_channel; 715 int i; 716 717 /* Inverse modified discrete cosine transform */ 718 q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer); 719 720 q->imlt_window(q, buffer1, gains_ptr, previous_buffer); 721 722 /* Apply gain profile */ 723 for (i = 0; i < 8; i++) 724 if (gains_ptr->now[i] || gains_ptr->now[i + 1]) 725 q->interpolate(q, &buffer1[q->gain_size_factor * i], 726 gains_ptr->now[i], gains_ptr->now[i + 1]); 727 728 /* Save away the current to be previous block. */ 729 memcpy(previous_buffer, buffer0, 730 q->samples_per_channel * sizeof(*previous_buffer)); 731} 732 733 734/** 735 * function for getting the jointstereo coupling information 736 * 737 * @param q pointer to the COOKContext 738 * @param decouple_tab decoupling array 739 */ 740static int decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab) 741{ 742 int i; 743 int vlc = get_bits1(&q->gb); 744 int start = cplband[p->js_subband_start]; 745 int end = cplband[p->subbands - 1]; 746 int length = end - start + 1; 747 748 if (start > end) 749 return 0; 750 751 if (vlc) 752 for (i = 0; i < length; i++) 753 decouple_tab[start + i] = get_vlc2(&q->gb, 754 p->channel_coupling.table, 755 p->channel_coupling.bits, 2); 756 else 757 for (i = 0; i < length; i++) { 758 int v = get_bits(&q->gb, p->js_vlc_bits); 759 if (v == (1<<p->js_vlc_bits)-1) { 760 av_log(q->avctx, AV_LOG_ERROR, "decouple value too large\n"); 761 return AVERROR_INVALIDDATA; 762 } 763 decouple_tab[start + i] = v; 764 } 765 return 0; 766} 767 768/** 769 * function decouples a pair of signals from a single signal via multiplication. 770 * 771 * @param q pointer to the COOKContext 772 * @param subband index of the current subband 773 * @param f1 multiplier for channel 1 extraction 774 * @param f2 multiplier for channel 2 extraction 775 * @param decode_buffer input buffer 776 * @param mlt_buffer1 pointer to left channel mlt coefficients 777 * @param mlt_buffer2 pointer to right channel mlt coefficients 778 */ 779static void decouple_float(COOKContext *q, 780 COOKSubpacket *p, 781 int subband, 782 float f1, float f2, 783 float *decode_buffer, 784 float *mlt_buffer1, float *mlt_buffer2) 785{ 786 int j, tmp_idx; 787 for (j = 0; j < SUBBAND_SIZE; j++) { 788 tmp_idx = ((p->js_subband_start + subband) * SUBBAND_SIZE) + j; 789 mlt_buffer1[SUBBAND_SIZE * subband + j] = f1 * decode_buffer[tmp_idx]; 790 mlt_buffer2[SUBBAND_SIZE * subband + j] = f2 * decode_buffer[tmp_idx]; 791 } 792} 793 794/** 795 * function for decoding joint stereo data 796 * 797 * @param q pointer to the COOKContext 798 * @param mlt_buffer1 pointer to left channel mlt coefficients 799 * @param mlt_buffer2 pointer to right channel mlt coefficients 800 */ 801static int joint_decode(COOKContext *q, COOKSubpacket *p, 802 float *mlt_buffer_left, float *mlt_buffer_right) 803{ 804 int i, j, res; 805 int decouple_tab[SUBBAND_SIZE] = { 0 }; 806 float *decode_buffer = q->decode_buffer_0; 807 int idx, cpl_tmp; 808 float f1, f2; 809 const float *cplscale; 810 811 memset(decode_buffer, 0, sizeof(q->decode_buffer_0)); 812 813 /* Make sure the buffers are zeroed out. */ 814 memset(mlt_buffer_left, 0, 1024 * sizeof(*mlt_buffer_left)); 815 memset(mlt_buffer_right, 0, 1024 * sizeof(*mlt_buffer_right)); 816 if ((res = decouple_info(q, p, decouple_tab)) < 0) 817 return res; 818 if ((res = mono_decode(q, p, decode_buffer)) < 0) 819 return res; 820 /* The two channels are stored interleaved in decode_buffer. */ 821 for (i = 0; i < p->js_subband_start; i++) { 822 for (j = 0; j < SUBBAND_SIZE; j++) { 823 mlt_buffer_left[i * 20 + j] = decode_buffer[i * 40 + j]; 824 mlt_buffer_right[i * 20 + j] = decode_buffer[i * 40 + 20 + j]; 825 } 826 } 827 828 /* When we reach js_subband_start (the higher frequencies) 829 the coefficients are stored in a coupling scheme. */ 830 idx = (1 << p->js_vlc_bits) - 1; 831 for (i = p->js_subband_start; i < p->subbands; i++) { 832 cpl_tmp = cplband[i]; 833 idx -= decouple_tab[cpl_tmp]; 834 cplscale = q->cplscales[p->js_vlc_bits - 2]; // choose decoupler table 835 f1 = cplscale[decouple_tab[cpl_tmp] + 1]; 836 f2 = cplscale[idx]; 837 q->decouple(q, p, i, f1, f2, decode_buffer, 838 mlt_buffer_left, mlt_buffer_right); 839 idx = (1 << p->js_vlc_bits) - 1; 840 } 841 842 return 0; 843} 844 845/** 846 * First part of subpacket decoding: 847 * decode raw stream bytes and read gain info. 848 * 849 * @param q pointer to the COOKContext 850 * @param inbuffer pointer to raw stream data 851 * @param gains_ptr array of current/prev gain pointers 852 */ 853static inline void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p, 854 const uint8_t *inbuffer, 855 cook_gains *gains_ptr) 856{ 857 int offset; 858 859 offset = decode_bytes(inbuffer, q->decoded_bytes_buffer, 860 p->bits_per_subpacket / 8); 861 init_get_bits(&q->gb, q->decoded_bytes_buffer + offset, 862 p->bits_per_subpacket); 863 decode_gain_info(&q->gb, gains_ptr->now); 864 865 /* Swap current and previous gains */ 866 FFSWAP(int *, gains_ptr->now, gains_ptr->previous); 867} 868 869/** 870 * Saturate the output signal and interleave. 871 * 872 * @param q pointer to the COOKContext 873 * @param out pointer to the output vector 874 */ 875static void saturate_output_float(COOKContext *q, float *out) 876{ 877 q->adsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel, 878 -1.0f, 1.0f, FFALIGN(q->samples_per_channel, 8)); 879} 880 881 882/** 883 * Final part of subpacket decoding: 884 * Apply modulated lapped transform, gain compensation, 885 * clip and convert to integer. 886 * 887 * @param q pointer to the COOKContext 888 * @param decode_buffer pointer to the mlt coefficients 889 * @param gains_ptr array of current/prev gain pointers 890 * @param previous_buffer pointer to the previous buffer to be used for overlapping 891 * @param out pointer to the output buffer 892 */ 893static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer, 894 cook_gains *gains_ptr, float *previous_buffer, 895 float *out) 896{ 897 imlt_gain(q, decode_buffer, gains_ptr, previous_buffer); 898 if (out) 899 q->saturate_output(q, out); 900} 901 902 903/** 904 * Cook subpacket decoding. This function returns one decoded subpacket, 905 * usually 1024 samples per channel. 906 * 907 * @param q pointer to the COOKContext 908 * @param inbuffer pointer to the inbuffer 909 * @param outbuffer pointer to the outbuffer 910 */ 911static int decode_subpacket(COOKContext *q, COOKSubpacket *p, 912 const uint8_t *inbuffer, float **outbuffer) 913{ 914 int sub_packet_size = p->size; 915 int res; 916 917 memset(q->decode_buffer_1, 0, sizeof(q->decode_buffer_1)); 918 decode_bytes_and_gain(q, p, inbuffer, &p->gains1); 919 920 if (p->joint_stereo) { 921 if ((res = joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2)) < 0) 922 return res; 923 } else { 924 if ((res = mono_decode(q, p, q->decode_buffer_1)) < 0) 925 return res; 926 927 if (p->num_channels == 2) { 928 decode_bytes_and_gain(q, p, inbuffer + sub_packet_size / 2, &p->gains2); 929 if ((res = mono_decode(q, p, q->decode_buffer_2)) < 0) 930 return res; 931 } 932 } 933 934 mlt_compensate_output(q, q->decode_buffer_1, &p->gains1, 935 p->mono_previous_buffer1, 936 outbuffer ? outbuffer[p->ch_idx] : NULL); 937 938 if (p->num_channels == 2) { 939 if (p->joint_stereo) 940 mlt_compensate_output(q, q->decode_buffer_2, &p->gains1, 941 p->mono_previous_buffer2, 942 outbuffer ? outbuffer[p->ch_idx + 1] : NULL); 943 else 944 mlt_compensate_output(q, q->decode_buffer_2, &p->gains2, 945 p->mono_previous_buffer2, 946 outbuffer ? outbuffer[p->ch_idx + 1] : NULL); 947 } 948 949 return 0; 950} 951 952 953static int cook_decode_frame(AVCodecContext *avctx, void *data, 954 int *got_frame_ptr, AVPacket *avpkt) 955{ 956 AVFrame *frame = data; 957 const uint8_t *buf = avpkt->data; 958 int buf_size = avpkt->size; 959 COOKContext *q = avctx->priv_data; 960 float **samples = NULL; 961 int i, ret; 962 int offset = 0; 963 int chidx = 0; 964 965 if (buf_size < avctx->block_align) 966 return buf_size; 967 968 /* get output buffer */ 969 if (q->discarded_packets >= 2) { 970 frame->nb_samples = q->samples_per_channel; 971 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) 972 return ret; 973 samples = (float **)frame->extended_data; 974 } 975 976 /* estimate subpacket sizes */ 977 q->subpacket[0].size = avctx->block_align; 978 979 for (i = 1; i < q->num_subpackets; i++) { 980 q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i]; 981 q->subpacket[0].size -= q->subpacket[i].size + 1; 982 if (q->subpacket[0].size < 0) { 983 av_log(avctx, AV_LOG_DEBUG, 984 "frame subpacket size total > avctx->block_align!\n"); 985 return AVERROR_INVALIDDATA; 986 } 987 } 988 989 /* decode supbackets */ 990 for (i = 0; i < q->num_subpackets; i++) { 991 q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size * 8) >> 992 q->subpacket[i].bits_per_subpdiv; 993 q->subpacket[i].ch_idx = chidx; 994 av_log(avctx, AV_LOG_DEBUG, 995 "subpacket[%i] size %i js %i %i block_align %i\n", 996 i, q->subpacket[i].size, q->subpacket[i].joint_stereo, offset, 997 avctx->block_align); 998 999 if ((ret = decode_subpacket(q, &q->subpacket[i], buf + offset, samples)) < 0) 1000 return ret; 1001 offset += q->subpacket[i].size; 1002 chidx += q->subpacket[i].num_channels; 1003 av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] %i %i\n", 1004 i, q->subpacket[i].size * 8, get_bits_count(&q->gb)); 1005 } 1006 1007 /* Discard the first two frames: no valid audio. */ 1008 if (q->discarded_packets < 2) { 1009 q->discarded_packets++; 1010 *got_frame_ptr = 0; 1011 return avctx->block_align; 1012 } 1013 1014 *got_frame_ptr = 1; 1015 1016 return avctx->block_align; 1017} 1018 1019#ifdef DEBUG 1020static void dump_cook_context(COOKContext *q) 1021{ 1022 //int i=0; 1023#define PRINT(a, b) av_dlog(q->avctx, " %s = %d\n", a, b); 1024 av_dlog(q->avctx, "COOKextradata\n"); 1025 av_dlog(q->avctx, "cookversion=%x\n", q->subpacket[0].cookversion); 1026 if (q->subpacket[0].cookversion > STEREO) { 1027 PRINT("js_subband_start", q->subpacket[0].js_subband_start); 1028 PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits); 1029 } 1030 av_dlog(q->avctx, "COOKContext\n"); 1031 PRINT("nb_channels", q->avctx->channels); 1032 PRINT("bit_rate", q->avctx->bit_rate); 1033 PRINT("sample_rate", q->avctx->sample_rate); 1034 PRINT("samples_per_channel", q->subpacket[0].samples_per_channel); 1035 PRINT("subbands", q->subpacket[0].subbands); 1036 PRINT("js_subband_start", q->subpacket[0].js_subband_start); 1037 PRINT("log2_numvector_size", q->subpacket[0].log2_numvector_size); 1038 PRINT("numvector_size", q->subpacket[0].numvector_size); 1039 PRINT("total_subbands", q->subpacket[0].total_subbands); 1040} 1041#endif 1042 1043/** 1044 * Cook initialization 1045 * 1046 * @param avctx pointer to the AVCodecContext 1047 */ 1048static av_cold int cook_decode_init(AVCodecContext *avctx) 1049{ 1050 COOKContext *q = avctx->priv_data; 1051 const uint8_t *edata_ptr = avctx->extradata; 1052 const uint8_t *edata_ptr_end = edata_ptr + avctx->extradata_size; 1053 int extradata_size = avctx->extradata_size; 1054 int s = 0; 1055 unsigned int channel_mask = 0; 1056 int samples_per_frame = 0; 1057 int ret; 1058 q->avctx = avctx; 1059 1060 /* Take care of the codec specific extradata. */ 1061 if (extradata_size <= 0) { 1062 av_log(avctx, AV_LOG_ERROR, "Necessary extradata missing!\n"); 1063 return AVERROR_INVALIDDATA; 1064 } 1065 av_log(avctx, AV_LOG_DEBUG, "codecdata_length=%d\n", avctx->extradata_size); 1066 1067 /* Take data from the AVCodecContext (RM container). */ 1068 if (!avctx->channels) { 1069 av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n"); 1070 return AVERROR_INVALIDDATA; 1071 } 1072 1073 /* Initialize RNG. */ 1074 av_lfg_init(&q->random_state, 0); 1075 1076 ff_audiodsp_init(&q->adsp); 1077 1078 while (edata_ptr < edata_ptr_end) { 1079 /* 8 for mono, 16 for stereo, ? for multichannel 1080 Swap to right endianness so we don't need to care later on. */ 1081 if (extradata_size >= 8) { 1082 q->subpacket[s].cookversion = bytestream_get_be32(&edata_ptr); 1083 samples_per_frame = bytestream_get_be16(&edata_ptr); 1084 q->subpacket[s].subbands = bytestream_get_be16(&edata_ptr); 1085 extradata_size -= 8; 1086 } 1087 if (extradata_size >= 8) { 1088 bytestream_get_be32(&edata_ptr); // Unknown unused 1089 q->subpacket[s].js_subband_start = bytestream_get_be16(&edata_ptr); 1090 if (q->subpacket[s].js_subband_start >= 51) { 1091 av_log(avctx, AV_LOG_ERROR, "js_subband_start %d is too large\n", q->subpacket[s].js_subband_start); 1092 return AVERROR_INVALIDDATA; 1093 } 1094 1095 q->subpacket[s].js_vlc_bits = bytestream_get_be16(&edata_ptr); 1096 extradata_size -= 8; 1097 } 1098 1099 /* Initialize extradata related variables. */ 1100 q->subpacket[s].samples_per_channel = samples_per_frame / avctx->channels; 1101 q->subpacket[s].bits_per_subpacket = avctx->block_align * 8; 1102 1103 /* Initialize default data states. */ 1104 q->subpacket[s].log2_numvector_size = 5; 1105 q->subpacket[s].total_subbands = q->subpacket[s].subbands; 1106 q->subpacket[s].num_channels = 1; 1107 1108 /* Initialize version-dependent variables */ 1109 1110 av_log(avctx, AV_LOG_DEBUG, "subpacket[%i].cookversion=%x\n", s, 1111 q->subpacket[s].cookversion); 1112 q->subpacket[s].joint_stereo = 0; 1113 switch (q->subpacket[s].cookversion) { 1114 case MONO: 1115 if (avctx->channels != 1) { 1116 avpriv_request_sample(avctx, "Container channels != 1"); 1117 return AVERROR_PATCHWELCOME; 1118 } 1119 av_log(avctx, AV_LOG_DEBUG, "MONO\n"); 1120 break; 1121 case STEREO: 1122 if (avctx->channels != 1) { 1123 q->subpacket[s].bits_per_subpdiv = 1; 1124 q->subpacket[s].num_channels = 2; 1125 } 1126 av_log(avctx, AV_LOG_DEBUG, "STEREO\n"); 1127 break; 1128 case JOINT_STEREO: 1129 if (avctx->channels != 2) { 1130 avpriv_request_sample(avctx, "Container channels != 2"); 1131 return AVERROR_PATCHWELCOME; 1132 } 1133 av_log(avctx, AV_LOG_DEBUG, "JOINT_STEREO\n"); 1134 if (avctx->extradata_size >= 16) { 1135 q->subpacket[s].total_subbands = q->subpacket[s].subbands + 1136 q->subpacket[s].js_subband_start; 1137 q->subpacket[s].joint_stereo = 1; 1138 q->subpacket[s].num_channels = 2; 1139 } 1140 if (q->subpacket[s].samples_per_channel > 256) { 1141 q->subpacket[s].log2_numvector_size = 6; 1142 } 1143 if (q->subpacket[s].samples_per_channel > 512) { 1144 q->subpacket[s].log2_numvector_size = 7; 1145 } 1146 break; 1147 case MC_COOK: 1148 av_log(avctx, AV_LOG_DEBUG, "MULTI_CHANNEL\n"); 1149 if (extradata_size >= 4) 1150 channel_mask |= q->subpacket[s].channel_mask = bytestream_get_be32(&edata_ptr); 1151 1152 if (av_get_channel_layout_nb_channels(q->subpacket[s].channel_mask) > 1) { 1153 q->subpacket[s].total_subbands = q->subpacket[s].subbands + 1154 q->subpacket[s].js_subband_start; 1155 q->subpacket[s].joint_stereo = 1; 1156 q->subpacket[s].num_channels = 2; 1157 q->subpacket[s].samples_per_channel = samples_per_frame >> 1; 1158 1159 if (q->subpacket[s].samples_per_channel > 256) { 1160 q->subpacket[s].log2_numvector_size = 6; 1161 } 1162 if (q->subpacket[s].samples_per_channel > 512) { 1163 q->subpacket[s].log2_numvector_size = 7; 1164 } 1165 } else 1166 q->subpacket[s].samples_per_channel = samples_per_frame; 1167 1168 break; 1169 default: 1170 avpriv_request_sample(avctx, "Cook version %d", 1171 q->subpacket[s].cookversion); 1172 return AVERROR_PATCHWELCOME; 1173 } 1174 1175 if (s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) { 1176 av_log(avctx, AV_LOG_ERROR, "different number of samples per channel!\n"); 1177 return AVERROR_INVALIDDATA; 1178 } else 1179 q->samples_per_channel = q->subpacket[0].samples_per_channel; 1180 1181 1182 /* Initialize variable relations */ 1183 q->subpacket[s].numvector_size = (1 << q->subpacket[s].log2_numvector_size); 1184 1185 /* Try to catch some obviously faulty streams, othervise it might be exploitable */ 1186 if (q->subpacket[s].total_subbands > 53) { 1187 avpriv_request_sample(avctx, "total_subbands > 53"); 1188 return AVERROR_PATCHWELCOME; 1189 } 1190 1191 if ((q->subpacket[s].js_vlc_bits > 6) || 1192 (q->subpacket[s].js_vlc_bits < 2 * q->subpacket[s].joint_stereo)) { 1193 av_log(avctx, AV_LOG_ERROR, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n", 1194 q->subpacket[s].js_vlc_bits, 2 * q->subpacket[s].joint_stereo); 1195 return AVERROR_INVALIDDATA; 1196 } 1197 1198 if (q->subpacket[s].subbands > 50) { 1199 avpriv_request_sample(avctx, "subbands > 50"); 1200 return AVERROR_PATCHWELCOME; 1201 } 1202 if (q->subpacket[s].subbands == 0) { 1203 avpriv_request_sample(avctx, "subbands = 0"); 1204 return AVERROR_PATCHWELCOME; 1205 } 1206 q->subpacket[s].gains1.now = q->subpacket[s].gain_1; 1207 q->subpacket[s].gains1.previous = q->subpacket[s].gain_2; 1208 q->subpacket[s].gains2.now = q->subpacket[s].gain_3; 1209 q->subpacket[s].gains2.previous = q->subpacket[s].gain_4; 1210 1211 if (q->num_subpackets + q->subpacket[s].num_channels > q->avctx->channels) { 1212 av_log(avctx, AV_LOG_ERROR, "Too many subpackets %d for channels %d\n", q->num_subpackets, q->avctx->channels); 1213 return AVERROR_INVALIDDATA; 1214 } 1215 1216 q->num_subpackets++; 1217 s++; 1218 if (s > MAX_SUBPACKETS) { 1219 avpriv_request_sample(avctx, "subpackets > %d", MAX_SUBPACKETS); 1220 return AVERROR_PATCHWELCOME; 1221 } 1222 } 1223 /* Generate tables */ 1224 init_pow2table(); 1225 init_gain_table(q); 1226 init_cplscales_table(q); 1227 1228 if ((ret = init_cook_vlc_tables(q))) 1229 return ret; 1230 1231 1232 if (avctx->block_align >= UINT_MAX / 2) 1233 return AVERROR(EINVAL); 1234 1235 /* Pad the databuffer with: 1236 DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(), 1237 FF_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */ 1238 q->decoded_bytes_buffer = 1239 av_mallocz(avctx->block_align 1240 + DECODE_BYTES_PAD1(avctx->block_align) 1241 + FF_INPUT_BUFFER_PADDING_SIZE); 1242 if (q->decoded_bytes_buffer == NULL) 1243 return AVERROR(ENOMEM); 1244 1245 /* Initialize transform. */ 1246 if ((ret = init_cook_mlt(q))) 1247 return ret; 1248 1249 /* Initialize COOK signal arithmetic handling */ 1250 if (1) { 1251 q->scalar_dequant = scalar_dequant_float; 1252 q->decouple = decouple_float; 1253 q->imlt_window = imlt_window_float; 1254 q->interpolate = interpolate_float; 1255 q->saturate_output = saturate_output_float; 1256 } 1257 1258 /* Try to catch some obviously faulty streams, othervise it might be exploitable */ 1259 if (q->samples_per_channel != 256 && q->samples_per_channel != 512 && 1260 q->samples_per_channel != 1024) { 1261 avpriv_request_sample(avctx, "samples_per_channel = %d", 1262 q->samples_per_channel); 1263 return AVERROR_PATCHWELCOME; 1264 } 1265 1266 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; 1267 if (channel_mask) 1268 avctx->channel_layout = channel_mask; 1269 else 1270 avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO; 1271 1272#ifdef DEBUG 1273 dump_cook_context(q); 1274#endif 1275 return 0; 1276} 1277 1278AVCodec ff_cook_decoder = { 1279 .name = "cook", 1280 .long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"), 1281 .type = AVMEDIA_TYPE_AUDIO, 1282 .id = AV_CODEC_ID_COOK, 1283 .priv_data_size = sizeof(COOKContext), 1284 .init = cook_decode_init, 1285 .close = cook_decode_close, 1286 .decode = cook_decode_frame, 1287 .capabilities = CODEC_CAP_DR1, 1288 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, 1289 AV_SAMPLE_FMT_NONE }, 1290}; 1291