FFmpeg: libavcodec/cook.c Source File

FFmpeg

[フレーム]

libavcodec

cook.c

Go to the documentation of this file.

1 /*

2 * COOK compatible decoder

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 */

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 */

45 #include "libavutil/channel_layout.h"

46 #include "libavutil/lfg.h"

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"

57 #include "cookdata.h"

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

65 #define SUBBAND_SIZE 20

66 #define MAX_SUBPACKETS 5

68 typedef struct cook_gains {

69 int *now;

70 int *previous;

71 } cook_gains;

73 typedef struct COOKSubpacket {

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;

91 float mono_previous_buffer1[1024];

92 float mono_previous_buffer2[1024];

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

102 typedef 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

161 static float pow2tab[127];

162 static float rootpow2tab[127];

163

164 /*************** init functions ***************/

165

166 /* table generator */

167 static av_cold void init_pow2table(void)

168 {

169 /* fast way of computing 2^i and 2^(0.5*i) for -63 <= i < 64 */

170 int i;

171 static const float exp2_tab[2] = {1, M_SQRT2};

172 float exp2_val = powf(2, -63);

173 float root_val = powf(2, -32);

174 for (i = -63; i < 64; i++) {

175 if (!(i & 1))

176 root_val *= 2;

177 pow2tab[63 + i] = exp2_val;

178 rootpow2tab[63 + i] = root_val * exp2_tab[i & 1];

179 exp2_val *= 2;

180 }

181 }

182

183 /* table generator */

184 static av_cold void init_gain_table(COOKContext *q)

185 {

186 int i;

187 q->gain_size_factor = q->samples_per_channel / 8;

188 for (i = 0; i < 23; i++)

189 q->gain_table[i] = pow(pow2tab[i + 52],

190 (1.0 / (double) q->gain_size_factor));

191 }

192

193

194 static av_cold int init_cook_vlc_tables(COOKContext *q)

195 {

196 int i, result;

197

198 result = 0;

199 for (i = 0; i < 13; i++) {

200 result |= init_vlc(&q->envelope_quant_index[i], 9, 24,

201 envelope_quant_index_huffbits[i], 1, 1,

202 envelope_quant_index_huffcodes[i], 2, 2, 0);

203 }

204 av_log(q->avctx, AV_LOG_DEBUG, "sqvh VLC init\n");

205 for (i = 0; i < 7; i++) {

206 result |= init_vlc(&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i],

207 cvh_huffbits[i], 1, 1,

208 cvh_huffcodes[i], 2, 2, 0);

209 }

210

211 for (i = 0; i < q->num_subpackets; i++) {

212 if (q->subpacket[i].joint_stereo == 1) {

213 result |= init_vlc(&q->subpacket[i].channel_coupling, 6,

214 (1 << q->subpacket[i].js_vlc_bits) - 1,

215 ccpl_huffbits[q->subpacket[i].js_vlc_bits - 2], 1, 1,

216 ccpl_huffcodes[q->subpacket[i].js_vlc_bits - 2], 2, 2, 0);

217 av_log(q->avctx, AV_LOG_DEBUG, "subpacket %i Joint-stereo VLC used.\n", i);

218 }

219 }

220

221 av_log(q->avctx, AV_LOG_DEBUG, "VLC tables initialized.\n");

222 return result;

223 }

224

225 static av_cold int init_cook_mlt(COOKContext *q)

226 {

227 int j, ret;

228 int mlt_size = q->samples_per_channel;

229

230 if ((q->mlt_window = av_malloc_array(mlt_size, sizeof(*q->mlt_window))) == 0)

231 return AVERROR(ENOMEM);

232

233 /* Initialize the MLT window: simple sine window. */

234 ff_sine_window_init(q->mlt_window, mlt_size);

235 for (j = 0; j < mlt_size; j++)

236 q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);

237

238 /* Initialize the MDCT. */

239 if ((ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size) + 1, 1, 1.0 / 32768.0))) {

240 av_freep(&q->mlt_window);

241 return ret;

242 }

243 av_log(q->avctx, AV_LOG_DEBUG, "MDCT initialized, order = %d.\n",

244 av_log2(mlt_size) + 1);

245

246 return 0;

247 }

248

249 static av_cold void init_cplscales_table(COOKContext *q)

250 {

251 int i;

252 for (i = 0; i < 5; i++)

253 q->cplscales[i] = cplscales[i];

254 }

255

256 /*************** init functions end ***********/

257

258 #define DECODE_BYTES_PAD1(bytes) (3 - ((bytes) + 3) % 4)

259 #define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))

260

261 /**

262 * Cook indata decoding, every 32 bits are XORed with 0x37c511f2.

263 * Why? No idea, some checksum/error detection method maybe.

264 *

265 * Out buffer size: extra bytes are needed to cope with

266 * padding/misalignment.

267 * Subpackets passed to the decoder can contain two, consecutive

268 * half-subpackets, of identical but arbitrary size.

269 * 1234 1234 1234 1234 extraA extraB

270 * Case 1: AAAA BBBB 0 0

271 * Case 2: AAAA ABBB BB-- 3 3

272 * Case 3: AAAA AABB BBBB 2 2

273 * Case 4: AAAA AAAB BBBB BB-- 1 5

274 *

275 * Nice way to waste CPU cycles.

276 *

277 * @param inbuffer pointer to byte array of indata

278 * @param out pointer to byte array of outdata

279 * @param bytes number of bytes

280 */

281 static inline int decode_bytes(const uint8_t *inbuffer, uint8_t *out, int bytes)

282 {

283 static const uint32_t tab[4] = {

284 AV_BE2NE32C(0x37c511f2u), AV_BE2NE32C(0xf237c511u),

285 AV_BE2NE32C(0x11f237c5u), AV_BE2NE32C(0xc511f237u),

286 };

287 int i, off;

288 uint32_t c;

289 const uint32_t *buf;

290 uint32_t *obuf = (uint32_t *) out;

291 /* FIXME: 64 bit platforms would be able to do 64 bits at a time.

292 * I'm too lazy though, should be something like

293 * for (i = 0; i < bitamount / 64; i++)

294 * (int64_t) out[i] = 0x37c511f237c511f2 ^ av_be2ne64(int64_t) in[i]);

295 * Buffer alignment needs to be checked. */

296

297 off = (intptr_t) inbuffer & 3;

298 buf = (const uint32_t *) (inbuffer - off);

299 c = tab[off];

300 bytes += 3 + off;

301 for (i = 0; i < bytes / 4; i++)

302 obuf[i] = c ^ buf[i];

303

304 return off;

305 }

306

307 static av_cold int cook_decode_close(AVCodecContext *avctx)

308 {

309 int i;

310 COOKContext *q = avctx->priv_data;

311 av_log(avctx, AV_LOG_DEBUG, "Deallocating memory.\n");

312

313 /* Free allocated memory buffers. */

314 av_freep(&q->mlt_window);

315 av_freep(&q->decoded_bytes_buffer);

316

317 /* Free the transform. */

318 ff_mdct_end(&q->mdct_ctx);

319

320 /* Free the VLC tables. */

321 for (i = 0; i < 13; i++)

322 ff_free_vlc(&q->envelope_quant_index[i]);

323 for (i = 0; i < 7; i++)

324 ff_free_vlc(&q->sqvh[i]);

325 for (i = 0; i < q->num_subpackets; i++)

326 ff_free_vlc(&q->subpacket[i].channel_coupling);

327

328 av_log(avctx, AV_LOG_DEBUG, "Memory deallocated.\n");

329

330 return 0;

331 }

332

333 /**

334 * Fill the gain array for the timedomain quantization.

335 *

336 * @param gb pointer to the GetBitContext

337 * @param gaininfo array[9] of gain indexes

338 */

339 static void decode_gain_info(GetBitContext *gb, int *gaininfo)

340 {

341 int i, n;

342

343 n = get_unary(gb, 0, get_bits_left(gb)); // amount of elements*2 to update

344

345 i = 0;

346 while (n--) {

347 int index = get_bits(gb, 3);

348 int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1;

349

350 while (i <= index)

351 gaininfo[i++] = gain;

352 }

353 while (i <= 8)

354 gaininfo[i++] = 0;

355 }

356

357 /**

358 * Create the quant index table needed for the envelope.

359 *

360 * @param q pointer to the COOKContext

361 * @param quant_index_table pointer to the array

362 */

363 static int decode_envelope(COOKContext *q, COOKSubpacket *p,

364 int *quant_index_table)

365 {

366 int i, j, vlc_index;

367

368 quant_index_table[0] = get_bits(&q->gb, 6) - 6; // This is used later in categorize

369

370 for (i = 1; i < p->total_subbands; i++) {

371 vlc_index = i;

372 if (i >= p->js_subband_start * 2) {

373 vlc_index -= p->js_subband_start;

374 } else {

375 vlc_index /= 2;

376 if (vlc_index < 1)

377 vlc_index = 1;

378 }

379 if (vlc_index > 13)

380 vlc_index = 13; // the VLC tables >13 are identical to No. 13

381

382 j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index - 1].table,

383 q->envelope_quant_index[vlc_index - 1].bits, 2);

384 quant_index_table[i] = quant_index_table[i - 1] + j - 12; // differential encoding

385 if (quant_index_table[i] > 63 || quant_index_table[i] < -63) {

386 av_log(q->avctx, AV_LOG_ERROR,

387 "Invalid quantizer %d at position %d, outside [-63, 63] range\n",

388 quant_index_table[i], i);

389 return AVERROR_INVALIDDATA;

390 }

391 }

392

393 return 0;

394 }

395

396 /**

397 * Calculate the category and category_index vector.

398 *

399 * @param q pointer to the COOKContext

400 * @param quant_index_table pointer to the array

401 * @param category pointer to the category array

402 * @param category_index pointer to the category_index array

403 */

404 static void categorize(COOKContext *q, COOKSubpacket *p, const int *quant_index_table,

405 int *category, int *category_index)

406 {

407 int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;

408 int exp_index2[102] = { 0 };

409 int exp_index1[102] = { 0 };

410

411 int tmp_categorize_array[128 * 2] = { 0 };

412 int tmp_categorize_array1_idx = p->numvector_size;

413 int tmp_categorize_array2_idx = p->numvector_size;

414

415 bits_left = p->bits_per_subpacket - get_bits_count(&q->gb);

416

417 if (bits_left > q->samples_per_channel)

418 bits_left = q->samples_per_channel +

419 ((bits_left - q->samples_per_channel) * 5) / 8;

420

421 bias = -32;

422

423 /* Estimate bias. */

424 for (i = 32; i > 0; i = i / 2) {

425 num_bits = 0;

426 index = 0;

427 for (j = p->total_subbands; j > 0; j--) {

428 exp_idx = av_clip_uintp2((i - quant_index_table[index] + bias) / 2, 3);

429 index++;

430 num_bits += expbits_tab[exp_idx];

431 }

432 if (num_bits >= bits_left - 32)

433 bias += i;

434 }

435

436 /* Calculate total number of bits. */

437 num_bits = 0;

438 for (i = 0; i < p->total_subbands; i++) {

439 exp_idx = av_clip_uintp2((bias - quant_index_table[i]) / 2, 3);

440 num_bits += expbits_tab[exp_idx];

441 exp_index1[i] = exp_idx;

442 exp_index2[i] = exp_idx;

443 }

444 tmpbias1 = tmpbias2 = num_bits;

445

446 for (j = 1; j < p->numvector_size; j++) {

447 if (tmpbias1 + tmpbias2 > 2 * bits_left) { /* ---> */

448 int max = -999999;

449 index = -1;

450 for (i = 0; i < p->total_subbands; i++) {

451 if (exp_index1[i] < 7) {

452 v = (-2 * exp_index1[i]) - quant_index_table[i] + bias;

453 if (v >= max) {

454 max = v;

455 index = i;

456 }

457 }

458 }

459 if (index == -1)

460 break;

461 tmp_categorize_array[tmp_categorize_array1_idx++] = index;

462 tmpbias1 -= expbits_tab[exp_index1[index]] -

463 expbits_tab[exp_index1[index] + 1];

464 ++exp_index1[index];

465 } else { /* <--- */

466 int min = 999999;

467 index = -1;

468 for (i = 0; i < p->total_subbands; i++) {

469 if (exp_index2[i] > 0) {

470 v = (-2 * exp_index2[i]) - quant_index_table[i] + bias;

471 if (v < min) {

472 min = v;

473 index = i;

474 }

475 }

476 }

477 if (index == -1)

478 break;

479 tmp_categorize_array[--tmp_categorize_array2_idx] = index;

480 tmpbias2 -= expbits_tab[exp_index2[index]] -

481 expbits_tab[exp_index2[index] - 1];

482 --exp_index2[index];

483 }

484 }

485

486 for (i = 0; i < p->total_subbands; i++)

487 category[i] = exp_index2[i];

488

489 for (i = 0; i < p->numvector_size - 1; i++)

490 category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];

491 }

492

493

494 /**

495 * Expand the category vector.

496 *

497 * @param q pointer to the COOKContext

498 * @param category pointer to the category array

499 * @param category_index pointer to the category_index array

500 */

501 static inline void expand_category(COOKContext *q, int *category,

502 int *category_index)

503 {

504 int i;

505 for (i = 0; i < q->num_vectors; i++)

506 {

507 int idx = category_index[i];

508 if (++category[idx] >= FF_ARRAY_ELEMS(dither_tab))

509 --category[idx];

510 }

511 }

512

513 /**

514 * The real requantization of the mltcoefs

515 *

516 * @param q pointer to the COOKContext

517 * @param index index

518 * @param quant_index quantisation index

519 * @param subband_coef_index array of indexes to quant_centroid_tab

520 * @param subband_coef_sign signs of coefficients

521 * @param mlt_p pointer into the mlt buffer

522 */

523 static void scalar_dequant_float(COOKContext *q, int index, int quant_index,

524 int *subband_coef_index, int *subband_coef_sign,

525 float *mlt_p)

526 {

527 int i;

528 float f1;

529

530 for (i = 0; i < SUBBAND_SIZE; i++) {

531 if (subband_coef_index[i]) {

532 f1 = quant_centroid_tab[index][subband_coef_index[i]];

533 if (subband_coef_sign[i])

534 f1 = -f1;

535 } else {

536 /* noise coding if subband_coef_index[i] == 0 */

537 f1 = dither_tab[index];

538 if (av_lfg_get(&q->random_state) < 0x80000000)

539 f1 = -f1;

540 }

541 mlt_p[i] = f1 * rootpow2tab[quant_index + 63];

542 }

543 }

544 /**

545 * Unpack the subband_coef_index and subband_coef_sign vectors.

546 *

547 * @param q pointer to the COOKContext

548 * @param category pointer to the category array

549 * @param subband_coef_index array of indexes to quant_centroid_tab

550 * @param subband_coef_sign signs of coefficients

551 */

552 static int unpack_SQVH(COOKContext *q, COOKSubpacket *p, int category,

553 int *subband_coef_index, int *subband_coef_sign)

554 {

555 int i, j;

556 int vlc, vd, tmp, result;

557

558 vd = vd_tab[category];

559 result = 0;

560 for (i = 0; i < vpr_tab[category]; i++) {

561 vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);

562 if (p->bits_per_subpacket < get_bits_count(&q->gb)) {

563 vlc = 0;

564 result = 1;

565 }

566 for (j = vd - 1; j >= 0; j--) {

567 tmp = (vlc * invradix_tab[category]) / 0x100000;

568 subband_coef_index[vd * i + j] = vlc - tmp * (kmax_tab[category] + 1);

569 vlc = tmp;

570 }

571 for (j = 0; j < vd; j++) {

572 if (subband_coef_index[i * vd + j]) {

573 if (get_bits_count(&q->gb) < p->bits_per_subpacket) {

574 subband_coef_sign[i * vd + j] = get_bits1(&q->gb);

575 } else {

576 result = 1;

577 subband_coef_sign[i * vd + j] = 0;

578 }

579 } else {

580 subband_coef_sign[i * vd + j] = 0;

581 }

582 }

583 }

584 return result;

585 }

586

587

588 /**

589 * Fill the mlt_buffer with mlt coefficients.

590 *

591 * @param q pointer to the COOKContext

592 * @param category pointer to the category array

593 * @param quant_index_table pointer to the array

594 * @param mlt_buffer pointer to mlt coefficients

595 */

596 static void decode_vectors(COOKContext *q, COOKSubpacket *p, int *category,

597 int *quant_index_table, float *mlt_buffer)

598 {

599 /* A zero in this table means that the subband coefficient is

600 random noise coded. */

601 int subband_coef_index[SUBBAND_SIZE];

602 /* A zero in this table means that the subband coefficient is a

603 positive multiplicator. */

604 int subband_coef_sign[SUBBAND_SIZE];

605 int band, j;

606 int index = 0;

607

608 for (band = 0; band < p->total_subbands; band++) {

609 index = category[band];

610 if (category[band] < 7) {

611 if (unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)) {

612 index = 7;

613 for (j = 0; j < p->total_subbands; j++)

614 category[band + j] = 7;

615 }

616 }

617 if (index >= 7) {

618 memset(subband_coef_index, 0, sizeof(subband_coef_index));

619 memset(subband_coef_sign, 0, sizeof(subband_coef_sign));

620 }

621 q->scalar_dequant(q, index, quant_index_table[band],

622 subband_coef_index, subband_coef_sign,

623 &mlt_buffer[band * SUBBAND_SIZE]);

624 }

625

626 /* FIXME: should this be removed, or moved into loop above? */

627 if (p->total_subbands * SUBBAND_SIZE >= q->samples_per_channel)

628 return;

629 }

630

631

632 static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)

633 {

634 int category_index[128] = { 0 };

635 int category[128] = { 0 };

636 int quant_index_table[102];

637 int res, i;

638

639 if ((res = decode_envelope(q, p, quant_index_table)) < 0)

640 return res;

641 q->num_vectors = get_bits(&q->gb, p->log2_numvector_size);

642 categorize(q, p, quant_index_table, category, category_index);

643 expand_category(q, category, category_index);

644 for (i=0; i<p->total_subbands; i++) {

645 if (category[i] > 7)

646 return AVERROR_INVALIDDATA;

647 }

648 decode_vectors(q, p, category, quant_index_table, mlt_buffer);

649

650 return 0;

651 }

652

653

654 /**

655 * the actual requantization of the timedomain samples

656 *

657 * @param q pointer to the COOKContext

658 * @param buffer pointer to the timedomain buffer

659 * @param gain_index index for the block multiplier

660 * @param gain_index_next index for the next block multiplier

661 */

662 static void interpolate_float(COOKContext *q, float *buffer,

663 int gain_index, int gain_index_next)

664 {

665 int i;

666 float fc1, fc2;

667 fc1 = pow2tab[gain_index + 63];

668

669 if (gain_index == gain_index_next) { // static gain

670 for (i = 0; i < q->gain_size_factor; i++)

671 buffer[i] *= fc1;

672 } else { // smooth gain

673 fc2 = q->gain_table[11 + (gain_index_next - gain_index)];

674 for (i = 0; i < q->gain_size_factor; i++) {

675 buffer[i] *= fc1;

676 fc1 *= fc2;

677 }

678 }

679 }

680

681 /**

682 * Apply transform window, overlap buffers.

683 *

684 * @param q pointer to the COOKContext

685 * @param inbuffer pointer to the mltcoefficients

686 * @param gains_ptr current and previous gains

687 * @param previous_buffer pointer to the previous buffer to be used for overlapping

688 */

689 static void imlt_window_float(COOKContext *q, float *inbuffer,

690 cook_gains *gains_ptr, float *previous_buffer)

691 {

692 const float fc = pow2tab[gains_ptr->previous[0] + 63];

693 int i;

694 /* The weird thing here, is that the two halves of the time domain

695 * buffer are swapped. Also, the newest data, that we save away for

696 * next frame, has the wrong sign. Hence the subtraction below.

697 * Almost sounds like a complex conjugate/reverse data/FFT effect.

698 */

699

700 /* Apply window and overlap */

701 for (i = 0; i < q->samples_per_channel; i++)

702 inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] -

703 previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];

704 }

705

706 /**

707 * The modulated lapped transform, this takes transform coefficients

708 * and transforms them into timedomain samples.

709 * Apply transform window, overlap buffers, apply gain profile

710 * and buffer management.

711 *

712 * @param q pointer to the COOKContext

713 * @param inbuffer pointer to the mltcoefficients

714 * @param gains_ptr current and previous gains

715 * @param previous_buffer pointer to the previous buffer to be used for overlapping

716 */

717 static void imlt_gain(COOKContext *q, float *inbuffer,

718 cook_gains *gains_ptr, float *previous_buffer)

719 {

720 float *buffer0 = q->mono_mdct_output;

721 float *buffer1 = q->mono_mdct_output + q->samples_per_channel;

722 int i;

723

724 /* Inverse modified discrete cosine transform */

725 q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);

726

727 q->imlt_window(q, buffer1, gains_ptr, previous_buffer);

728

729 /* Apply gain profile */

730 for (i = 0; i < 8; i++)

731 if (gains_ptr->now[i] || gains_ptr->now[i + 1])

732 q->interpolate(q, &buffer1[q->gain_size_factor * i],

733 gains_ptr->now[i], gains_ptr->now[i + 1]);

734

735 /* Save away the current to be previous block. */

736 memcpy(previous_buffer, buffer0,

737 q->samples_per_channel * sizeof(*previous_buffer));

738 }

739

740

741 /**

742 * function for getting the jointstereo coupling information

743 *

744 * @param q pointer to the COOKContext

745 * @param decouple_tab decoupling array

746 */

747 static int decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)

748 {

749 int i;

750 int vlc = get_bits1(&q->gb);

751 int start = cplband[p->js_subband_start];

752 int end = cplband[p->subbands - 1];

753 int length = end - start + 1;

754

755 if (start > end)

756 return 0;

757

758 if (vlc)

759 for (i = 0; i < length; i++)

760 decouple_tab[start + i] = get_vlc2(&q->gb,

761 p->channel_coupling.table,

762 p->channel_coupling.bits, 2);

763 else

764 for (i = 0; i < length; i++) {

765 int v = get_bits(&q->gb, p->js_vlc_bits);

766 if (v == (1<<p->js_vlc_bits)-1) {

767 av_log(q->avctx, AV_LOG_ERROR, "decouple value too large\n");

768 return AVERROR_INVALIDDATA;

769 }

770 decouple_tab[start + i] = v;

771 }

772 return 0;

773 }

774

775 /**

776 * function decouples a pair of signals from a single signal via multiplication.

777 *

778 * @param q pointer to the COOKContext

779 * @param subband index of the current subband

780 * @param f1 multiplier for channel 1 extraction

781 * @param f2 multiplier for channel 2 extraction

782 * @param decode_buffer input buffer

783 * @param mlt_buffer1 pointer to left channel mlt coefficients

784 * @param mlt_buffer2 pointer to right channel mlt coefficients

785 */

786 static void decouple_float(COOKContext *q,

787 COOKSubpacket *p,

788 int subband,

789 float f1, float f2,

790 float *decode_buffer,

791 float *mlt_buffer1, float *mlt_buffer2)

792 {

793 int j, tmp_idx;

794 for (j = 0; j < SUBBAND_SIZE; j++) {

795 tmp_idx = ((p->js_subband_start + subband) * SUBBAND_SIZE) + j;

796 mlt_buffer1[SUBBAND_SIZE * subband + j] = f1 * decode_buffer[tmp_idx];

797 mlt_buffer2[SUBBAND_SIZE * subband + j] = f2 * decode_buffer[tmp_idx];

798 }

799 }

800

801 /**

802 * function for decoding joint stereo data

803 *

804 * @param q pointer to the COOKContext

805 * @param mlt_buffer1 pointer to left channel mlt coefficients

806 * @param mlt_buffer2 pointer to right channel mlt coefficients

807 */

808 static int joint_decode(COOKContext *q, COOKSubpacket *p,

809 float *mlt_buffer_left, float *mlt_buffer_right)

810 {

811 int i, j, res;

812 int decouple_tab[SUBBAND_SIZE] = { 0 };

813 float *decode_buffer = q->decode_buffer_0;

814 int idx, cpl_tmp;

815 float f1, f2;

816 const float *cplscale;

817

818 memset(decode_buffer, 0, sizeof(q->decode_buffer_0));

819

820 /* Make sure the buffers are zeroed out. */

821 memset(mlt_buffer_left, 0, 1024 * sizeof(*mlt_buffer_left));

822 memset(mlt_buffer_right, 0, 1024 * sizeof(*mlt_buffer_right));

823 if ((res = decouple_info(q, p, decouple_tab)) < 0)

824 return res;

825 if ((res = mono_decode(q, p, decode_buffer)) < 0)

826 return res;

827 /* The two channels are stored interleaved in decode_buffer. */

828 for (i = 0; i < p->js_subband_start; i++) {

829 for (j = 0; j < SUBBAND_SIZE; j++) {

830 mlt_buffer_left[i * 20 + j] = decode_buffer[i * 40 + j];

831 mlt_buffer_right[i * 20 + j] = decode_buffer[i * 40 + 20 + j];

832 }

833 }

834

835 /* When we reach js_subband_start (the higher frequencies)

836 the coefficients are stored in a coupling scheme. */

837 idx = (1 << p->js_vlc_bits) - 1;

838 for (i = p->js_subband_start; i < p->subbands; i++) {

839 cpl_tmp = cplband[i];

840 idx -= decouple_tab[cpl_tmp];

841 cplscale = q->cplscales[p->js_vlc_bits - 2]; // choose decoupler table

842 f1 = cplscale[decouple_tab[cpl_tmp] + 1];

843 f2 = cplscale[idx];

844 q->decouple(q, p, i, f1, f2, decode_buffer,

845 mlt_buffer_left, mlt_buffer_right);

846 idx = (1 << p->js_vlc_bits) - 1;

847 }

848

849 return 0;

850 }

851

852 /**

853 * First part of subpacket decoding:

854 * decode raw stream bytes and read gain info.

855 *

856 * @param q pointer to the COOKContext

857 * @param inbuffer pointer to raw stream data

858 * @param gains_ptr array of current/prev gain pointers

859 */

860 static inline void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p,

861 const uint8_t *inbuffer,

862 cook_gains *gains_ptr)

863 {

864 int offset;

865

866 offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,

867 p->bits_per_subpacket / 8);

868 init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,

869 p->bits_per_subpacket);

870 decode_gain_info(&q->gb, gains_ptr->now);

871

872 /* Swap current and previous gains */

873 FFSWAP(int *, gains_ptr->now, gains_ptr->previous);

874 }

875

876 /**

877 * Saturate the output signal and interleave.

878 *

879 * @param q pointer to the COOKContext

880 * @param out pointer to the output vector

881 */

882 static void saturate_output_float(COOKContext *q, float *out)

883 {

884 q->adsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel,

885 -1.0f, 1.0f, FFALIGN(q->samples_per_channel, 8));

886 }

887

888

889 /**

890 * Final part of subpacket decoding:

891 * Apply modulated lapped transform, gain compensation,

892 * clip and convert to integer.

893 *

894 * @param q pointer to the COOKContext

895 * @param decode_buffer pointer to the mlt coefficients

896 * @param gains_ptr array of current/prev gain pointers

897 * @param previous_buffer pointer to the previous buffer to be used for overlapping

898 * @param out pointer to the output buffer

899 */

900 static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,

901 cook_gains *gains_ptr, float *previous_buffer,

902 float *out)

903 {

904 imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);

905 if (out)

906 q->saturate_output(q, out);

907 }

908

909

910 /**

911 * Cook subpacket decoding. This function returns one decoded subpacket,

912 * usually 1024 samples per channel.

913 *

914 * @param q pointer to the COOKContext

915 * @param inbuffer pointer to the inbuffer

916 * @param outbuffer pointer to the outbuffer

917 */

918 static int decode_subpacket(COOKContext *q, COOKSubpacket *p,

919 const uint8_t *inbuffer, float **outbuffer)

920 {

921 int sub_packet_size = p->size;

922 int res;

923

924 memset(q->decode_buffer_1, 0, sizeof(q->decode_buffer_1));

925 decode_bytes_and_gain(q, p, inbuffer, &p->gains1);

926

927 if (p->joint_stereo) {

928 if ((res = joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2)) < 0)

929 return res;

930 } else {

931 if ((res = mono_decode(q, p, q->decode_buffer_1)) < 0)

932 return res;

933

934 if (p->num_channels == 2) {

935 decode_bytes_and_gain(q, p, inbuffer + sub_packet_size / 2, &p->gains2);

936 if ((res = mono_decode(q, p, q->decode_buffer_2)) < 0)

937 return res;

938 }

939 }

940

941 mlt_compensate_output(q, q->decode_buffer_1, &p->gains1,

942 p->mono_previous_buffer1,

943 outbuffer ? outbuffer[p->ch_idx] : NULL);

944

945 if (p->num_channels == 2) {

946 if (p->joint_stereo)

947 mlt_compensate_output(q, q->decode_buffer_2, &p->gains1,

948 p->mono_previous_buffer2,

949 outbuffer ? outbuffer[p->ch_idx + 1] : NULL);

950 else

951 mlt_compensate_output(q, q->decode_buffer_2, &p->gains2,

952 p->mono_previous_buffer2,

953 outbuffer ? outbuffer[p->ch_idx + 1] : NULL);

954 }

955

956 return 0;

957 }

958

959

960 static int cook_decode_frame(AVCodecContext *avctx, void *data,

961 int *got_frame_ptr, AVPacket *avpkt)

962 {

963 AVFrame *frame = data;

964 const uint8_t *buf = avpkt->data;

965 int buf_size = avpkt->size;

966 COOKContext *q = avctx->priv_data;

967 float **samples = NULL;

968 int i, ret;

969 int offset = 0;

970 int chidx = 0;

971

972 if (buf_size < avctx->block_align)

973 return buf_size;

974

975 /* get output buffer */

976 if (q->discarded_packets >= 2) {

977 frame->nb_samples = q->samples_per_channel;

978 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)

979 return ret;

980 samples = (float **)frame->extended_data;

981 }

982

983 /* estimate subpacket sizes */

984 q->subpacket[0].size = avctx->block_align;

985

986 for (i = 1; i < q->num_subpackets; i++) {

987 q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];

988 q->subpacket[0].size -= q->subpacket[i].size + 1;

989 if (q->subpacket[0].size < 0) {

990 av_log(avctx, AV_LOG_DEBUG,

991 "frame subpacket size total > avctx->block_align!\n");

992 return AVERROR_INVALIDDATA;

993 }

994 }

995

996 /* decode supbackets */

997 for (i = 0; i < q->num_subpackets; i++) {

998 q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size * 8) >>

999 q->subpacket[i].bits_per_subpdiv;

1000 q->subpacket[i].ch_idx = chidx;

1001 av_log(avctx, AV_LOG_DEBUG,

1002 "subpacket[%i] size %i js %i %i block_align %i\n",

1003 i, q->subpacket[i].size, q->subpacket[i].joint_stereo, offset,

1004 avctx->block_align);

1005

1006 if ((ret = decode_subpacket(q, &q->subpacket[i], buf + offset, samples)) < 0)

1007 return ret;

1008 offset += q->subpacket[i].size;

1009 chidx += q->subpacket[i].num_channels;

1010 av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] %i %i\n",

1011 i, q->subpacket[i].size * 8, get_bits_count(&q->gb));

1012 }

1013

1014 /* Discard the first two frames: no valid audio. */

1015 if (q->discarded_packets < 2) {

1016 q->discarded_packets++;

1017 *got_frame_ptr = 0;

1018 return avctx->block_align;

1019 }

1020

1021 *got_frame_ptr = 1;

1022

1023 return avctx->block_align;

1024 }

1025

1026 static void dump_cook_context(COOKContext *q)

1027 {

1028 //int i=0;

1029 #define PRINT(a, b) ff_dlog(q->avctx, " %s = %d\n", a, b);

1030 ff_dlog(q->avctx, "COOKextradata\n");

1031 ff_dlog(q->avctx, "cookversion=%x\n", q->subpacket[0].cookversion);

1032 if (q->subpacket[0].cookversion > STEREO) {

1033 PRINT("js_subband_start", q->subpacket[0].js_subband_start);

1034 PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits);

1035 }

1036 ff_dlog(q->avctx, "COOKContext\n");

1037 PRINT("nb_channels", q->avctx->channels);

1038 PRINT("bit_rate", (int)q->avctx->bit_rate);

1039 PRINT("sample_rate", q->avctx->sample_rate);

1040 PRINT("samples_per_channel", q->subpacket[0].samples_per_channel);

1041 PRINT("subbands", q->subpacket[0].subbands);

1042 PRINT("js_subband_start", q->subpacket[0].js_subband_start);

1043 PRINT("log2_numvector_size", q->subpacket[0].log2_numvector_size);

1044 PRINT("numvector_size", q->subpacket[0].numvector_size);

1045 PRINT("total_subbands", q->subpacket[0].total_subbands);

1046 }

1047

1048 /**

1049 * Cook initialization

1050 *

1051 * @param avctx pointer to the AVCodecContext

1052 */

1053 static av_cold int cook_decode_init(AVCodecContext *avctx)

1054 {

1055 COOKContext *q = avctx->priv_data;

1056 const uint8_t *edata_ptr = avctx->extradata;

1057 const uint8_t *edata_ptr_end = edata_ptr + avctx->extradata_size;

1058 int extradata_size = avctx->extradata_size;

1059 int s = 0;

1060 unsigned int channel_mask = 0;

1061 int samples_per_frame = 0;

1062 int ret;

1063 q->avctx = avctx;

1064

1065 /* Take care of the codec specific extradata. */

1066 if (extradata_size < 8) {

1067 av_log(avctx, AV_LOG_ERROR, "Necessary extradata missing!\n");

1068 return AVERROR_INVALIDDATA;

1069 }

1070 av_log(avctx, AV_LOG_DEBUG, "codecdata_length=%d\n", avctx->extradata_size);

1071

1072 /* Take data from the AVCodecContext (RM container). */

1073 if (!avctx->channels) {

1074 av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");

1075 return AVERROR_INVALIDDATA;

1076 }

1077

1078 /* Initialize RNG. */

1079 av_lfg_init(&q->random_state, 0);

1080

1081 ff_audiodsp_init(&q->adsp);

1082

1083 while (edata_ptr < edata_ptr_end) {

1084 /* 8 for mono, 16 for stereo, ? for multichannel

1085 Swap to right endianness so we don't need to care later on. */

1086 if (extradata_size >= 8) {

1087 q->subpacket[s].cookversion = bytestream_get_be32(&edata_ptr);

1088 samples_per_frame = bytestream_get_be16(&edata_ptr);

1089 q->subpacket[s].subbands = bytestream_get_be16(&edata_ptr);

1090 extradata_size -= 8;

1091 }

1092 if (extradata_size >= 8) {

1093 bytestream_get_be32(&edata_ptr); // Unknown unused

1094 q->subpacket[s].js_subband_start = bytestream_get_be16(&edata_ptr);

1095 if (q->subpacket[s].js_subband_start >= 51) {

1096 av_log(avctx, AV_LOG_ERROR, "js_subband_start %d is too large\n", q->subpacket[s].js_subband_start);

1097 return AVERROR_INVALIDDATA;

1098 }

1099

1100 q->subpacket[s].js_vlc_bits = bytestream_get_be16(&edata_ptr);

1101 extradata_size -= 8;

1102 }

1103

1104 /* Initialize extradata related variables. */

1105 q->subpacket[s].samples_per_channel = samples_per_frame / avctx->channels;

1106 q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;

1107

1108 /* Initialize default data states. */

1109 q->subpacket[s].log2_numvector_size = 5;

1110 q->subpacket[s].total_subbands = q->subpacket[s].subbands;

1111 q->subpacket[s].num_channels = 1;

1112

1113 /* Initialize version-dependent variables */

1114

1115 av_log(avctx, AV_LOG_DEBUG, "subpacket[%i].cookversion=%x\n", s,

1116 q->subpacket[s].cookversion);

1117 q->subpacket[s].joint_stereo = 0;

1118 switch (q->subpacket[s].cookversion) {

1119 case MONO:

1120 if (avctx->channels != 1) {

1121 avpriv_request_sample(avctx, "Container channels != 1");

1122 return AVERROR_PATCHWELCOME;

1123 }

1124 av_log(avctx, AV_LOG_DEBUG, "MONO\n");

1125 break;

1126 case STEREO:

1127 if (avctx->channels != 1) {

1128 q->subpacket[s].bits_per_subpdiv = 1;

1129 q->subpacket[s].num_channels = 2;

1130 }

1131 av_log(avctx, AV_LOG_DEBUG, "STEREO\n");

1132 break;

1133 case JOINT_STEREO:

1134 if (avctx->channels != 2) {

1135 avpriv_request_sample(avctx, "Container channels != 2");

1136 return AVERROR_PATCHWELCOME;

1137 }

1138 av_log(avctx, AV_LOG_DEBUG, "JOINT_STEREO\n");

1139 if (avctx->extradata_size >= 16) {

1140 q->subpacket[s].total_subbands = q->subpacket[s].subbands +

1141 q->subpacket[s].js_subband_start;

1142 q->subpacket[s].joint_stereo = 1;

1143 q->subpacket[s].num_channels = 2;

1144 }

1145 if (q->subpacket[s].samples_per_channel > 256) {

1146 q->subpacket[s].log2_numvector_size = 6;

1147 }

1148 if (q->subpacket[s].samples_per_channel > 512) {

1149 q->subpacket[s].log2_numvector_size = 7;

1150 }

1151 break;

1152 case MC_COOK:

1153 av_log(avctx, AV_LOG_DEBUG, "MULTI_CHANNEL\n");

1154 if (extradata_size >= 4)

1155 channel_mask |= q->subpacket[s].channel_mask = bytestream_get_be32(&edata_ptr);

1156

1157 if (av_get_channel_layout_nb_channels(q->subpacket[s].channel_mask) > 1) {

1158 q->subpacket[s].total_subbands = q->subpacket[s].subbands +

1159 q->subpacket[s].js_subband_start;

1160 q->subpacket[s].joint_stereo = 1;

1161 q->subpacket[s].num_channels = 2;

1162 q->subpacket[s].samples_per_channel = samples_per_frame >> 1;

1163

1164 if (q->subpacket[s].samples_per_channel > 256) {

1165 q->subpacket[s].log2_numvector_size = 6;

1166 }

1167 if (q->subpacket[s].samples_per_channel > 512) {

1168 q->subpacket[s].log2_numvector_size = 7;

1169 }

1170 } else

1171 q->subpacket[s].samples_per_channel = samples_per_frame;

1172

1173 break;

1174 default:

1175 avpriv_request_sample(avctx, "Cook version %d",

1176 q->subpacket[s].cookversion);

1177 return AVERROR_PATCHWELCOME;

1178 }

1179

1180 if (s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {

1181 av_log(avctx, AV_LOG_ERROR, "different number of samples per channel!\n");

1182 return AVERROR_INVALIDDATA;

1183 } else

1184 q->samples_per_channel = q->subpacket[0].samples_per_channel;

1185

1186

1187 /* Initialize variable relations */

1188 q->subpacket[s].numvector_size = (1 << q->subpacket[s].log2_numvector_size);

1189

1190 /* Try to catch some obviously faulty streams, othervise it might be exploitable */

1191 if (q->subpacket[s].total_subbands > 53) {

1192 avpriv_request_sample(avctx, "total_subbands > 53");

1193 return AVERROR_PATCHWELCOME;

1194 }

1195

1196 if ((q->subpacket[s].js_vlc_bits > 6) ||

1197 (q->subpacket[s].js_vlc_bits < 2 * q->subpacket[s].joint_stereo)) {

1198 av_log(avctx, AV_LOG_ERROR, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",

1199 q->subpacket[s].js_vlc_bits, 2 * q->subpacket[s].joint_stereo);

1200 return AVERROR_INVALIDDATA;

1201 }

1202

1203 if (q->subpacket[s].subbands > 50) {

1204 avpriv_request_sample(avctx, "subbands > 50");

1205 return AVERROR_PATCHWELCOME;

1206 }

1207 if (q->subpacket[s].subbands == 0) {

1208 avpriv_request_sample(avctx, "subbands = 0");

1209 return AVERROR_PATCHWELCOME;

1210 }

1211 q->subpacket[s].gains1.now = q->subpacket[s].gain_1;

1212 q->subpacket[s].gains1.previous = q->subpacket[s].gain_2;

1213 q->subpacket[s].gains2.now = q->subpacket[s].gain_3;

1214 q->subpacket[s].gains2.previous = q->subpacket[s].gain_4;

1215

1216 if (q->num_subpackets + q->subpacket[s].num_channels > q->avctx->channels) {

1217 av_log(avctx, AV_LOG_ERROR, "Too many subpackets %d for channels %d\n", q->num_subpackets, q->avctx->channels);

1218 return AVERROR_INVALIDDATA;

1219 }

1220

1221 q->num_subpackets++;

1222 s++;

1223 if (s > FFMIN(MAX_SUBPACKETS, avctx->block_align)) {

1224 avpriv_request_sample(avctx, "subpackets > %d", FFMIN(MAX_SUBPACKETS, avctx->block_align));

1225 return AVERROR_PATCHWELCOME;

1226 }

1227 }

1228 /* Generate tables */

1229 init_pow2table();

1230 init_gain_table(q);

1231 init_cplscales_table(q);

1232

1233 if ((ret = init_cook_vlc_tables(q)))

1234 return ret;

1235

1236

1237 if (avctx->block_align >= UINT_MAX / 2)

1238 return AVERROR(EINVAL);

1239

1240 /* Pad the databuffer with:

1241 DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),

1242 AV_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */

1243 q->decoded_bytes_buffer =

1244 av_mallocz(avctx->block_align

1245 + DECODE_BYTES_PAD1(avctx->block_align)

1246 + AV_INPUT_BUFFER_PADDING_SIZE);

1247 if (!q->decoded_bytes_buffer)

1248 return AVERROR(ENOMEM);

1249

1250 /* Initialize transform. */

1251 if ((ret = init_cook_mlt(q)))

1252 return ret;

1253

1254 /* Initialize COOK signal arithmetic handling */

1255 if (1) {

1256 q->scalar_dequant = scalar_dequant_float;

1257 q->decouple = decouple_float;

1258 q->imlt_window = imlt_window_float;

1259 q->interpolate = interpolate_float;

1260 q->saturate_output = saturate_output_float;

1261 }

1262

1263 /* Try to catch some obviously faulty streams, othervise it might be exploitable */

1264 if (q->samples_per_channel != 256 && q->samples_per_channel != 512 &&

1265 q->samples_per_channel != 1024) {

1266 avpriv_request_sample(avctx, "samples_per_channel = %d",

1267 q->samples_per_channel);

1268 return AVERROR_PATCHWELCOME;

1269 }

1270

1271 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

1272 if (channel_mask)

1273 avctx->channel_layout = channel_mask;

1274 else

1275 avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;

1276

1277

1278 dump_cook_context(q);

1279

1280 return 0;

1281 }

1282

1283 AVCodec ff_cook_decoder = {

1284 .name = "cook",

1285 .long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),

1286 .type = AVMEDIA_TYPE_AUDIO,

1287 .id = AV_CODEC_ID_COOK,

1288 .priv_data_size = sizeof(COOKContext),

1289 .init = cook_decode_init,

1290 .close = cook_decode_close,

1291 .decode = cook_decode_frame,

1292 .capabilities = AV_CODEC_CAP_DR1,

1293 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,

1294 AV_SAMPLE_FMT_NONE },

1295 };