FFmpeg: libavcodec/cook.c Source File

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cook.c

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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"

47 #include "avcodec.h"

48 #include "get_bits.h"

49 #include "dsputil.h"

50 #include "bytestream.h"

51 #include "fft.h"

52 #include "internal.h"

53 #include "sinewin.h"

55 #include "cookdata.h"

57 /* the different Cook versions */

58 #define MONO 0x1000001

59 #define STEREO 0x1000002

60 #define JOINT_STEREO 0x1000003

61 #define MC_COOK 0x2000000 // multichannel Cook, not supported

63 #define SUBBAND_SIZE 20

64 #define MAX_SUBPACKETS 5

66 typedef struct {

67 int *now;

68 int *previous;

69 } cook_gains;

71 typedef struct {

72 int ch_idx;

73 int size;

74 int num_channels;

75 int cookversion;

76 int subbands;

77 int js_subband_start;

78 int js_vlc_bits;

79 int samples_per_channel;

80 int log2_numvector_size;

81 unsigned int channel_mask;

82 VLC channel_coupling;

83 int joint_stereo;

84 int bits_per_subpacket;

85 int bits_per_subpdiv;

86 int total_subbands;

87 int numvector_size; // 1 << log2_numvector_size;

89 float mono_previous_buffer1[1024];

90 float mono_previous_buffer2[1024];

92 cook_gains gains1;

93 cook_gains gains2;

94 int gain_1[9];

95 int gain_2[9];

96 int gain_3[9];

97 int gain_4[9];

98 } COOKSubpacket;

100 typedef struct cook {

101 /*

102 * The following 5 functions provide the lowlevel arithmetic on

103 * the internal audio buffers.

104 */

105 void (*scalar_dequant)(struct cook *q, int index, int quant_index,

106 int *subband_coef_index, int *subband_coef_sign,

107 float *mlt_p);

108

109 void (*decouple)(struct cook *q,

110 COOKSubpacket *p,

111 int subband,

112 float f1, float f2,

113 float *decode_buffer,

114 float *mlt_buffer1, float *mlt_buffer2);

115

116 void (*imlt_window)(struct cook *q, float *buffer1,

117 cook_gains *gains_ptr, float *previous_buffer);

118

119 void (*interpolate)(struct cook *q, float *buffer,

120 int gain_index, int gain_index_next);

121

122 void (*saturate_output)(struct cook *q, float *out);

123

124 AVCodecContext* avctx;

125 DSPContext dsp;

126 GetBitContext gb;

127 /* stream data */

128 int num_vectors;

129 int samples_per_channel;

130 /* states */

131 AVLFG random_state;

132 int discarded_packets;

133

134 /* transform data */

135 FFTContext mdct_ctx;

136 float* mlt_window;

137

138 /* VLC data */

139 VLC envelope_quant_index[13];

140 VLC sqvh[7]; // scalar quantization

141

142 /* generatable tables and related variables */

143 int gain_size_factor;

144 float gain_table[23];

145

146 /* data buffers */

147

148 uint8_t* decoded_bytes_buffer;

149 DECLARE_ALIGNED(32, float, mono_mdct_output)[2048];

150 float decode_buffer_1[1024];

151 float decode_buffer_2[1024];

152 float decode_buffer_0[1060]; /* static allocation for joint decode */

153

154 const float *cplscales[5];

155 int num_subpackets;

156 COOKSubpacket subpacket[MAX_SUBPACKETS];

157 } COOKContext;

158

159 static float pow2tab[127];

160 static float rootpow2tab[127];

161

162 /*************** init functions ***************/

163

164 /* table generator */

165 static av_cold void init_pow2table(void)

166 {

167 int i;

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

169 pow2tab[63 + i] = pow(2, i);

170 rootpow2tab[63 + i] = sqrt(pow(2, i));

171 }

172 }

173

174 /* table generator */

175 static av_cold void init_gain_table(COOKContext *q)

176 {

177 int i;

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

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

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

181 (1.0 / (double) q->gain_size_factor));

182 }

183

184

185 static av_cold int init_cook_vlc_tables(COOKContext *q)

186 {

187 int i, result;

188

189 result = 0;

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

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

192 envelope_quant_index_huffbits[i], 1, 1,

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

194 }

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

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

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

198 cvh_huffbits[i], 1, 1,

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

200 }

201

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

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

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

205 (1 << q->subpacket[i].js_vlc_bits) - 1,

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

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

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

209 }

210 }

211

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

213 return result;

214 }

215

216 static av_cold int init_cook_mlt(COOKContext *q)

217 {

218 int j, ret;

219 int mlt_size = q->samples_per_channel;

220

221 if ((q->mlt_window = av_malloc(mlt_size * sizeof(*q->mlt_window))) == 0)

222 return AVERROR(ENOMEM);

223

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

225 ff_sine_window_init(q->mlt_window, mlt_size);

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

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

228

229 /* Initialize the MDCT. */

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

231 av_free(q->mlt_window);

232 return ret;

233 }

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

235 av_log2(mlt_size) + 1);

236

237 return 0;

238 }

239

240 static av_cold void init_cplscales_table(COOKContext *q)

241 {

242 int i;

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

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

245 }

246

247 /*************** init functions end ***********/

248

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

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

251

252 /**

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

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

255 *

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

257 * padding/misalignment.

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

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

260 * 1234 1234 1234 1234 extraA extraB

261 * Case 1: AAAA BBBB 0 0

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

263 * Case 3: AAAA AABB BBBB 2 2

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

265 *

266 * Nice way to waste CPU cycles.

267 *

268 * @param inbuffer pointer to byte array of indata

269 * @param out pointer to byte array of outdata

270 * @param bytes number of bytes

271 */

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

273 {

274 static const uint32_t tab[4] = {

275 AV_BE2NE32C(0x37c511f2u), AV_BE2NE32C(0xf237c511u),

276 AV_BE2NE32C(0x11f237c5u), AV_BE2NE32C(0xc511f237u),

277 };

278 int i, off;

279 uint32_t c;

280 const uint32_t *buf;

281 uint32_t *obuf = (uint32_t *) out;

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

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

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

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

286 * Buffer alignment needs to be checked. */

287

288 off = (intptr_t) inbuffer & 3;

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

290 c = tab[off];

291 bytes += 3 + off;

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

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

294

295 return off;

296 }

297

298 static av_cold int cook_decode_close(AVCodecContext *avctx)

299 {

300 int i;

301 COOKContext *q = avctx->priv_data;

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

303

304 /* Free allocated memory buffers. */

305 av_free(q->mlt_window);

306 av_free(q->decoded_bytes_buffer);

307

308 /* Free the transform. */

309 ff_mdct_end(&q->mdct_ctx);

310

311 /* Free the VLC tables. */

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

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

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

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

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

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

318

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

320

321 return 0;

322 }

323

324 /**

325 * Fill the gain array for the timedomain quantization.

326 *

327 * @param gb pointer to the GetBitContext

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

329 */

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

331 {

332 int i, n;

333

334 while (get_bits1(gb)) {

335 /* NOTHING */

336 }

337

338 n = get_bits_count(gb) - 1; // amount of elements*2 to update

339

340 i = 0;

341 while (n--) {

342 int index = get_bits(gb, 3);

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

344

345 while (i <= index)

346 gaininfo[i++] = gain;

347 }

348 while (i <= 8)

349 gaininfo[i++] = 0;

350 }

351

352 /**

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

354 *

355 * @param q pointer to the COOKContext

356 * @param quant_index_table pointer to the array

357 */

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

359 int *quant_index_table)

360 {

361 int i, j, vlc_index;

362

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

364

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

366 vlc_index = i;

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

368 vlc_index -= p->js_subband_start;

369 } else {

370 vlc_index /= 2;

371 if (vlc_index < 1)

372 vlc_index = 1;

373 }

374 if (vlc_index > 13)

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

376

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

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

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

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

381 av_log(q->avctx, AV_LOG_ERROR,

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

383 quant_index_table[i], i);

384 return AVERROR_INVALIDDATA;

385 }

386 }

387

388 return 0;

389 }

390

391 /**

392 * Calculate the category and category_index vector.

393 *

394 * @param q pointer to the COOKContext

395 * @param quant_index_table pointer to the array

396 * @param category pointer to the category array

397 * @param category_index pointer to the category_index array

398 */

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

400 int *category, int *category_index)

401 {

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

403 int exp_index2[102] = { 0 };

404 int exp_index1[102] = { 0 };

405

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

407 int tmp_categorize_array1_idx = p->numvector_size;

408 int tmp_categorize_array2_idx = p->numvector_size;

409

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

411

412 if (bits_left > q->samples_per_channel)

413 bits_left = q->samples_per_channel +

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

415

416 bias = -32;

417

418 /* Estimate bias. */

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

420 num_bits = 0;

421 index = 0;

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

423 exp_idx = av_clip((i - quant_index_table[index] + bias) / 2, 0, 7);

424 index++;

425 num_bits += expbits_tab[exp_idx];

426 }

427 if (num_bits >= bits_left - 32)

428 bias += i;

429 }

430

431 /* Calculate total number of bits. */

432 num_bits = 0;

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

434 exp_idx = av_clip((bias - quant_index_table[i]) / 2, 0, 7);

435 num_bits += expbits_tab[exp_idx];

436 exp_index1[i] = exp_idx;

437 exp_index2[i] = exp_idx;

438 }

439 tmpbias1 = tmpbias2 = num_bits;

440

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

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

443 int max = -999999;

444 index = -1;

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

446 if (exp_index1[i] < 7) {

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

448 if (v >= max) {

449 max = v;

450 index = i;

451 }

452 }

453 }

454 if (index == -1)

455 break;

456 tmp_categorize_array[tmp_categorize_array1_idx++] = index;

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

458 expbits_tab[exp_index1[index] + 1];

459 ++exp_index1[index];

460 } else { /* <--- */

461 int min = 999999;

462 index = -1;

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

464 if (exp_index2[i] > 0) {

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

466 if (v < min) {

467 min = v;

468 index = i;

469 }

470 }

471 }

472 if (index == -1)

473 break;

474 tmp_categorize_array[--tmp_categorize_array2_idx] = index;

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

476 expbits_tab[exp_index2[index] - 1];

477 --exp_index2[index];

478 }

479 }

480

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

482 category[i] = exp_index2[i];

483

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

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

486 }

487

488

489 /**

490 * Expand the category vector.

491 *

492 * @param q pointer to the COOKContext

493 * @param category pointer to the category array

494 * @param category_index pointer to the category_index array

495 */

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

497 int *category_index)

498 {

499 int i;

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

501 {

502 int idx = category_index[i];

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

504 --category[idx];

505 }

506 }

507

508 /**

509 * The real requantization of the mltcoefs

510 *

511 * @param q pointer to the COOKContext

512 * @param index index

513 * @param quant_index quantisation index

514 * @param subband_coef_index array of indexes to quant_centroid_tab

515 * @param subband_coef_sign signs of coefficients

516 * @param mlt_p pointer into the mlt buffer

517 */

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

519 int *subband_coef_index, int *subband_coef_sign,

520 float *mlt_p)

521 {

522 int i;

523 float f1;

524

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

526 if (subband_coef_index[i]) {

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

528 if (subband_coef_sign[i])

529 f1 = -f1;

530 } else {

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

532 f1 = dither_tab[index];

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

534 f1 = -f1;

535 }

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

537 }

538 }

539 /**

540 * Unpack the subband_coef_index and subband_coef_sign vectors.

541 *

542 * @param q pointer to the COOKContext

543 * @param category pointer to the category array

544 * @param subband_coef_index array of indexes to quant_centroid_tab

545 * @param subband_coef_sign signs of coefficients

546 */

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

548 int *subband_coef_index, int *subband_coef_sign)

549 {

550 int i, j;

551 int vlc, vd, tmp, result;

552

553 vd = vd_tab[category];

554 result = 0;

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

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

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

558 vlc = 0;

559 result = 1;

560 }

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

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

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

564 vlc = tmp;

565 }

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

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

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

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

570 } else {

571 result = 1;

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

573 }

574 } else {

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

576 }

577 }

578 }

579 return result;

580 }

581

582

583 /**

584 * Fill the mlt_buffer with mlt coefficients.

585 *

586 * @param q pointer to the COOKContext

587 * @param category pointer to the category array

588 * @param quant_index_table pointer to the array

589 * @param mlt_buffer pointer to mlt coefficients

590 */

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

592 int *quant_index_table, float *mlt_buffer)

593 {

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

595 random noise coded. */

596 int subband_coef_index[SUBBAND_SIZE];

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

598 positive multiplicator. */

599 int subband_coef_sign[SUBBAND_SIZE];

600 int band, j;

601 int index = 0;

602

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

604 index = category[band];

605 if (category[band] < 7) {

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

607 index = 7;

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

609 category[band + j] = 7;

610 }

611 }

612 if (index >= 7) {

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

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

615 }

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

617 subband_coef_index, subband_coef_sign,

618 &mlt_buffer[band * SUBBAND_SIZE]);

619 }

620

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

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

623 return;

624 }

625

626

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

628 {

629 int category_index[128] = { 0 };

630 int category[128] = { 0 };

631 int quant_index_table[102];

632 int res, i;

633

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

635 return res;

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

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

638 expand_category(q, category, category_index);

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

640 if (category[i] > 7)

641 return AVERROR_INVALIDDATA;

642 }

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

644

645 return 0;

646 }

647

648

649 /**

650 * the actual requantization of the timedomain samples

651 *

652 * @param q pointer to the COOKContext

653 * @param buffer pointer to the timedomain buffer

654 * @param gain_index index for the block multiplier

655 * @param gain_index_next index for the next block multiplier

656 */

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

658 int gain_index, int gain_index_next)

659 {

660 int i;

661 float fc1, fc2;

662 fc1 = pow2tab[gain_index + 63];

663

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

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

666 buffer[i] *= fc1;

667 } else { // smooth gain

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

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

670 buffer[i] *= fc1;

671 fc1 *= fc2;

672 }

673 }

674 }

675

676 /**

677 * Apply transform window, overlap buffers.

678 *

679 * @param q pointer to the COOKContext

680 * @param inbuffer pointer to the mltcoefficients

681 * @param gains_ptr current and previous gains

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

683 */

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

685 cook_gains *gains_ptr, float *previous_buffer)

686 {

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

688 int i;

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

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

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

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

693 */

694

695 /* Apply window and overlap */

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

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

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

699 }

700

701 /**

702 * The modulated lapped transform, this takes transform coefficients

703 * and transforms them into timedomain samples.

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

705 * and buffer management.

706 *

707 * @param q pointer to the COOKContext

708 * @param inbuffer pointer to the mltcoefficients

709 * @param gains_ptr current and previous gains

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

711 */

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

713 cook_gains *gains_ptr, float *previous_buffer)

714 {

715 float *buffer0 = q->mono_mdct_output;

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

717 int i;

718

719 /* Inverse modified discrete cosine transform */

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

721

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

723

724 /* Apply gain profile */

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

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

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

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

729

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

731 memcpy(previous_buffer, buffer0,

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

733 }

734

735

736 /**

737 * function for getting the jointstereo coupling information

738 *

739 * @param q pointer to the COOKContext

740 * @param decouple_tab decoupling array

741 */

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

743 {

744 int i;

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

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

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

748 int length = end - start + 1;

749

750 if (start > end)

751 return 0;

752

753 if (vlc)

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

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

756 p->channel_coupling.table,

757 p->channel_coupling.bits, 2);

758 else

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

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

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

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

763 return AVERROR_INVALIDDATA;

764 }

765 decouple_tab[start + i] = v;

766 }

767 return 0;

768 }

769

770 /**

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

772 *

773 * @param q pointer to the COOKContext

774 * @param subband index of the current subband

775 * @param f1 multiplier for channel 1 extraction

776 * @param f2 multiplier for channel 2 extraction

777 * @param decode_buffer input buffer

778 * @param mlt_buffer1 pointer to left channel mlt coefficients

779 * @param mlt_buffer2 pointer to right channel mlt coefficients

780 */

781 static void decouple_float(COOKContext *q,

782 COOKSubpacket *p,

783 int subband,

784 float f1, float f2,

785 float *decode_buffer,

786 float *mlt_buffer1, float *mlt_buffer2)

787 {

788 int j, tmp_idx;

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

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

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

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

793 }

794 }

795

796 /**

797 * function for decoding joint stereo data

798 *

799 * @param q pointer to the COOKContext

800 * @param mlt_buffer1 pointer to left channel mlt coefficients

801 * @param mlt_buffer2 pointer to right channel mlt coefficients

802 */

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

804 float *mlt_buffer_left, float *mlt_buffer_right)

805 {

806 int i, j, res;

807 int decouple_tab[SUBBAND_SIZE] = { 0 };

808 float *decode_buffer = q->decode_buffer_0;

809 int idx, cpl_tmp;

810 float f1, f2;

811 const float *cplscale;

812

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

814

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

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

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

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

819 return res;

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

821 return res;

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

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

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

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

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

827 }

828 }

829

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

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

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

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

834 cpl_tmp = cplband[i];

835 idx -= decouple_tab[cpl_tmp];

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

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

838 f2 = cplscale[idx];

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

840 mlt_buffer_left, mlt_buffer_right);

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

842 }

843

844 return 0;

845 }

846

847 /**

848 * First part of subpacket decoding:

849 * decode raw stream bytes and read gain info.

850 *

851 * @param q pointer to the COOKContext

852 * @param inbuffer pointer to raw stream data

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

854 */

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

856 const uint8_t *inbuffer,

857 cook_gains *gains_ptr)

858 {

859 int offset;

860

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

862 p->bits_per_subpacket / 8);

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

864 p->bits_per_subpacket);

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

866

867 /* Swap current and previous gains */

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

869 }

870

871 /**

872 * Saturate the output signal and interleave.

873 *

874 * @param q pointer to the COOKContext

875 * @param out pointer to the output vector

876 */

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

878 {

879 q->dsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel,

880 -1.0f, 1.0f, FFALIGN(q->samples_per_channel, 8));

881 }

882

883

884 /**

885 * Final part of subpacket decoding:

886 * Apply modulated lapped transform, gain compensation,

887 * clip and convert to integer.

888 *

889 * @param q pointer to the COOKContext

890 * @param decode_buffer pointer to the mlt coefficients

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

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

893 * @param out pointer to the output buffer

894 */

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

896 cook_gains *gains_ptr, float *previous_buffer,

897 float *out)

898 {

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

900 if (out)

901 q->saturate_output(q, out);

902 }

903

904

905 /**

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

907 * usually 1024 samples per channel.

908 *

909 * @param q pointer to the COOKContext

910 * @param inbuffer pointer to the inbuffer

911 * @param outbuffer pointer to the outbuffer

912 */

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

914 const uint8_t *inbuffer, float **outbuffer)

915 {

916 int sub_packet_size = p->size;

917 int res;

918

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

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

921

922 if (p->joint_stereo) {

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

924 return res;

925 } else {

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

927 return res;

928

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

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

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

932 return res;

933 }

934 }

935

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

937 p->mono_previous_buffer1,

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

939

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

941 if (p->joint_stereo)

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

943 p->mono_previous_buffer2,

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

945 else

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

947 p->mono_previous_buffer2,

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

949 }

950

951 return 0;

952 }

953

954

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

956 int *got_frame_ptr, AVPacket *avpkt)

957 {

958 AVFrame *frame = data;

959 const uint8_t *buf = avpkt->data;

960 int buf_size = avpkt->size;

961 COOKContext *q = avctx->priv_data;

962 float **samples = NULL;

963 int i, ret;

964 int offset = 0;

965 int chidx = 0;

966

967 if (buf_size < avctx->block_align)

968 return buf_size;

969

970 /* get output buffer */

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

972 frame->nb_samples = q->samples_per_channel;

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

974 return ret;

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

976 }

977

978 /* estimate subpacket sizes */

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

980

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

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

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

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

985 av_log(avctx, AV_LOG_DEBUG,

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

987 return AVERROR_INVALIDDATA;

988 }

989 }

990

991 /* decode supbackets */

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

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

994 q->subpacket[i].bits_per_subpdiv;

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

996 av_log(avctx, AV_LOG_DEBUG,

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

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

999 avctx->block_align);

1000

1001 if ((ret = decode_subpacket(q, &q->subpacket[i], buf + offset, samples)) < 0)

1002 return ret;

1003 offset += q->subpacket[i].size;

1004 chidx += q->subpacket[i].num_channels;

1005 av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] %i %i\n",

1006 i, q->subpacket[i].size * 8, get_bits_count(&q->gb));

1007 }

1008

1009 /* Discard the first two frames: no valid audio. */

1010 if (q->discarded_packets < 2) {

1011 q->discarded_packets++;

1012 *got_frame_ptr = 0;

1013 return avctx->block_align;

1014 }

1015

1016 *got_frame_ptr = 1;

1017

1018 return avctx->block_align;

1019 }

1020

1021 #ifdef DEBUG

1022 static void dump_cook_context(COOKContext *q)

1023 {

1024 //int i=0;

1025 #define PRINT(a, b) av_dlog(q->avctx, " %s = %d\n", a, b);

1026 av_dlog(q->avctx, "COOKextradata\n");

1027 av_dlog(q->avctx, "cookversion=%x\n", q->subpacket[0].cookversion);

1028 if (q->subpacket[0].cookversion > STEREO) {

1029 PRINT("js_subband_start", q->subpacket[0].js_subband_start);

1030 PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits);

1031 }

1032 av_dlog(q->avctx, "COOKContext\n");

1033 PRINT("nb_channels", q->avctx->channels);

1034 PRINT("bit_rate", q->avctx->bit_rate);

1035 PRINT("sample_rate", q->avctx->sample_rate);

1036 PRINT("samples_per_channel", q->subpacket[0].samples_per_channel);

1037 PRINT("subbands", q->subpacket[0].subbands);

1038 PRINT("js_subband_start", q->subpacket[0].js_subband_start);

1039 PRINT("log2_numvector_size", q->subpacket[0].log2_numvector_size);

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

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

1042 }

1043 #endif

1044

1045 /**

1046 * Cook initialization

1047 *

1048 * @param avctx pointer to the AVCodecContext

1049 */

1050 static av_cold int cook_decode_init(AVCodecContext *avctx)

1051 {

1052 COOKContext *q = avctx->priv_data;

1053 const uint8_t *edata_ptr = avctx->extradata;

1054 const uint8_t *edata_ptr_end = edata_ptr + avctx->extradata_size;

1055 int extradata_size = avctx->extradata_size;

1056 int s = 0;

1057 unsigned int channel_mask = 0;

1058 int samples_per_frame = 0;

1059 int ret;

1060 q->avctx = avctx;

1061

1062 /* Take care of the codec specific extradata. */

1063 if (extradata_size <= 0) {

1064 av_log(avctx, AV_LOG_ERROR, "Necessary extradata missing!\n");

1065 return AVERROR_INVALIDDATA;

1066 }

1067 av_log(avctx, AV_LOG_DEBUG, "codecdata_length=%d\n", avctx->extradata_size);

1068

1069 /* Take data from the AVCodecContext (RM container). */

1070 if (!avctx->channels) {

1071 av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");

1072 return AVERROR_INVALIDDATA;

1073 }

1074

1075 /* Initialize RNG. */

1076 av_lfg_init(&q->random_state, 0);

1077

1078 ff_dsputil_init(&q->dsp, avctx);

1079

1080 while (edata_ptr < edata_ptr_end) {

1081 /* 8 for mono, 16 for stereo, ? for multichannel

1082 Swap to right endianness so we don't need to care later on. */

1083 if (extradata_size >= 8) {

1084 q->subpacket[s].cookversion = bytestream_get_be32(&edata_ptr);

1085 samples_per_frame = bytestream_get_be16(&edata_ptr);

1086 q->subpacket[s].subbands = bytestream_get_be16(&edata_ptr);

1087 extradata_size -= 8;

1088 }

1089 if (extradata_size >= 8) {

1090 bytestream_get_be32(&edata_ptr); // Unknown unused

1091 q->subpacket[s].js_subband_start = bytestream_get_be16(&edata_ptr);

1092 if (q->subpacket[s].js_subband_start >= 51) {

1093 av_log(avctx, AV_LOG_ERROR, "js_subband_start %d is too large\n", q->subpacket[s].js_subband_start);

1094 return AVERROR_INVALIDDATA;

1095 }

1096

1097 q->subpacket[s].js_vlc_bits = bytestream_get_be16(&edata_ptr);

1098 extradata_size -= 8;

1099 }

1100

1101 /* Initialize extradata related variables. */

1102 q->subpacket[s].samples_per_channel = samples_per_frame / avctx->channels;

1103 q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;

1104

1105 /* Initialize default data states. */

1106 q->subpacket[s].log2_numvector_size = 5;

1107 q->subpacket[s].total_subbands = q->subpacket[s].subbands;

1108 q->subpacket[s].num_channels = 1;

1109

1110 /* Initialize version-dependent variables */

1111

1112 av_log(avctx, AV_LOG_DEBUG, "subpacket[%i].cookversion=%x\n", s,

1113 q->subpacket[s].cookversion);

1114 q->subpacket[s].joint_stereo = 0;

1115 switch (q->subpacket[s].cookversion) {

1116 case MONO:

1117 if (avctx->channels != 1) {

1118 avpriv_request_sample(avctx, "Container channels != 1");

1119 return AVERROR_PATCHWELCOME;

1120 }

1121 av_log(avctx, AV_LOG_DEBUG, "MONO\n");

1122 break;

1123 case STEREO:

1124 if (avctx->channels != 1) {

1125 q->subpacket[s].bits_per_subpdiv = 1;

1126 q->subpacket[s].num_channels = 2;

1127 }

1128 av_log(avctx, AV_LOG_DEBUG, "STEREO\n");

1129 break;

1130 case JOINT_STEREO:

1131 if (avctx->channels != 2) {

1132 avpriv_request_sample(avctx, "Container channels != 2");

1133 return AVERROR_PATCHWELCOME;

1134 }

1135 av_log(avctx, AV_LOG_DEBUG, "JOINT_STEREO\n");

1136 if (avctx->extradata_size >= 16) {

1137 q->subpacket[s].total_subbands = q->subpacket[s].subbands +

1138 q->subpacket[s].js_subband_start;

1139 q->subpacket[s].joint_stereo = 1;

1140 q->subpacket[s].num_channels = 2;

1141 }

1142 if (q->subpacket[s].samples_per_channel > 256) {

1143 q->subpacket[s].log2_numvector_size = 6;

1144 }

1145 if (q->subpacket[s].samples_per_channel > 512) {

1146 q->subpacket[s].log2_numvector_size = 7;

1147 }

1148 break;

1149 case MC_COOK:

1150 av_log(avctx, AV_LOG_DEBUG, "MULTI_CHANNEL\n");

1151 if (extradata_size >= 4)

1152 channel_mask |= q->subpacket[s].channel_mask = bytestream_get_be32(&edata_ptr);

1153

1154 if (av_get_channel_layout_nb_channels(q->subpacket[s].channel_mask) > 1) {

1155 q->subpacket[s].total_subbands = q->subpacket[s].subbands +

1156 q->subpacket[s].js_subband_start;

1157 q->subpacket[s].joint_stereo = 1;

1158 q->subpacket[s].num_channels = 2;

1159 q->subpacket[s].samples_per_channel = samples_per_frame >> 1;

1160

1161 if (q->subpacket[s].samples_per_channel > 256) {

1162 q->subpacket[s].log2_numvector_size = 6;

1163 }

1164 if (q->subpacket[s].samples_per_channel > 512) {

1165 q->subpacket[s].log2_numvector_size = 7;

1166 }

1167 } else

1168 q->subpacket[s].samples_per_channel = samples_per_frame;

1169

1170 break;

1171 default:

1172 avpriv_request_sample(avctx, "Cook version %d",

1173 q->subpacket[s].cookversion);

1174 return AVERROR_PATCHWELCOME;

1175 }

1176

1177 if (s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {

1178 av_log(avctx, AV_LOG_ERROR, "different number of samples per channel!\n");

1179 return AVERROR_INVALIDDATA;

1180 } else

1181 q->samples_per_channel = q->subpacket[0].samples_per_channel;

1182

1183

1184 /* Initialize variable relations */

1185 q->subpacket[s].numvector_size = (1 << q->subpacket[s].log2_numvector_size);

1186

1187 /* Try to catch some obviously faulty streams, othervise it might be exploitable */

1188 if (q->subpacket[s].total_subbands > 53) {

1189 avpriv_request_sample(avctx, "total_subbands > 53");

1190 return AVERROR_PATCHWELCOME;

1191 }

1192

1193 if ((q->subpacket[s].js_vlc_bits > 6) ||

1194 (q->subpacket[s].js_vlc_bits < 2 * q->subpacket[s].joint_stereo)) {

1195 av_log(avctx, AV_LOG_ERROR, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",

1196 q->subpacket[s].js_vlc_bits, 2 * q->subpacket[s].joint_stereo);

1197 return AVERROR_INVALIDDATA;

1198 }

1199

1200 if (q->subpacket[s].subbands > 50) {

1201 avpriv_request_sample(avctx, "subbands > 50");

1202 return AVERROR_PATCHWELCOME;

1203 }

1204 if (q->subpacket[s].subbands == 0) {

1205 avpriv_request_sample(avctx, "subbands = 0");

1206 return AVERROR_PATCHWELCOME;

1207 }

1208 q->subpacket[s].gains1.now = q->subpacket[s].gain_1;

1209 q->subpacket[s].gains1.previous = q->subpacket[s].gain_2;

1210 q->subpacket[s].gains2.now = q->subpacket[s].gain_3;

1211 q->subpacket[s].gains2.previous = q->subpacket[s].gain_4;

1212

1213 if (q->num_subpackets + q->subpacket[s].num_channels > q->avctx->channels) {

1214 av_log(avctx, AV_LOG_ERROR, "Too many subpackets %d for channels %d\n", q->num_subpackets, q->avctx->channels);

1215 return AVERROR_INVALIDDATA;

1216 }

1217

1218 q->num_subpackets++;

1219 s++;

1220 if (s > MAX_SUBPACKETS) {

1221 avpriv_request_sample(avctx, "subpackets > %d", MAX_SUBPACKETS);

1222 return AVERROR_PATCHWELCOME;

1223 }

1224 }

1225 /* Generate tables */

1226 init_pow2table();

1227 init_gain_table(q);

1228 init_cplscales_table(q);

1229

1230 if ((ret = init_cook_vlc_tables(q)))

1231 return ret;

1232

1233

1234 if (avctx->block_align >= UINT_MAX / 2)

1235 return AVERROR(EINVAL);

1236

1237 /* Pad the databuffer with:

1238 DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),

1239 FF_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */

1240 q->decoded_bytes_buffer =

1241 av_mallocz(avctx->block_align

1242 + DECODE_BYTES_PAD1(avctx->block_align)

1243 + FF_INPUT_BUFFER_PADDING_SIZE);

1244 if (q->decoded_bytes_buffer == NULL)

1245 return AVERROR(ENOMEM);

1246

1247 /* Initialize transform. */

1248 if ((ret = init_cook_mlt(q)))

1249 return ret;

1250

1251 /* Initialize COOK signal arithmetic handling */

1252 if (1) {

1253 q->scalar_dequant = scalar_dequant_float;

1254 q->decouple = decouple_float;

1255 q->imlt_window = imlt_window_float;

1256 q->interpolate = interpolate_float;

1257 q->saturate_output = saturate_output_float;

1258 }

1259

1260 /* Try to catch some obviously faulty streams, othervise it might be exploitable */

1261 if (q->samples_per_channel != 256 && q->samples_per_channel != 512 &&

1262 q->samples_per_channel != 1024) {

1263 avpriv_request_sample(avctx, "samples_per_channel = %d",

1264 q->samples_per_channel);

1265 return AVERROR_PATCHWELCOME;

1266 }

1267

1268 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

1269 if (channel_mask)

1270 avctx->channel_layout = channel_mask;

1271 else

1272 avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;

1273

1274 #ifdef DEBUG

1275 dump_cook_context(q);

1276 #endif

1277 return 0;

1278 }

1279

1280 AVCodec ff_cook_decoder = {

1281 .name = "cook",

1282 .long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),

1283 .type = AVMEDIA_TYPE_AUDIO,

1284 .id = AV_CODEC_ID_COOK,

1285 .priv_data_size = sizeof(COOKContext),

1286 .init = cook_decode_init,

1287 .close = cook_decode_close,

1288 .decode = cook_decode_frame,

1289 .capabilities = CODEC_CAP_DR1,

1290 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,

1291 AV_SAMPLE_FMT_NONE },

1292 };

Generated on Sat Jan 25 2014 19:51:45 for FFmpeg by doxygen 1.8.2