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"

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 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 */

177 static 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

187 static 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

218 static 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

242 static 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 */

274 static 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

300 static 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 */

332 static 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 */

356 static 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 */

397 static 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_uintp2((i - quant_index_table[index] + bias) / 2, 3);

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_uintp2((bias - quant_index_table[i]) / 2, 3);

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 */

494 static 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 */

516 static 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 */

545 static 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 */

589 static 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

625 static 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 */

655 static 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 */

682 static 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 */

710 static 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 */

740 static 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 */

779 static 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 */

801 static 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 */

853 static 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 */

875 static 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 */

893 static 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 */

911 static 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

953 static 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

1020 static 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 */

1048 static 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 < 8) {

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 > FFMIN(MAX_SUBPACKETS, avctx->block_align)) {

1219 avpriv_request_sample(avctx, "subpackets > %d", FFMIN(MAX_SUBPACKETS, avctx->block_align));

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)

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

1278 AVCodec 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 };

Generated on Sun Mar 8 2015 02:34:49 for FFmpeg by doxygen 1.8.2