FFmpeg: libavcodec/takdec.c Source File

FFmpeg

libavcodec

takdec.c

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1 /*

2 * TAK decoder

3 * Copyright (c) 2012 Paul B Mahol

4 *

5 * This file is part of FFmpeg.

6 *

7 * FFmpeg is free software; you can redistribute it and/or

8 * modify it under the terms of the GNU Lesser General Public

9 * License as published by the Free Software Foundation; either

10 * version 2.1 of the License, or (at your option) any later version.

11 *

12 * FFmpeg is distributed in the hope that it will be useful,

13 * but WITHOUT ANY WARRANTY; without even the implied warranty of

14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU

15 * Lesser General Public License for more details.

16 *

17 * You should have received a copy of the GNU Lesser General Public

18 * License along with FFmpeg; if not, write to the Free Software

19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

20 */

21

22 /**

23 * @file

24 * TAK (Tom's lossless Audio Kompressor) decoder

25 * @author Paul B Mahol

26 */

27

28 #include "libavutil/internal.h"

29 #include "libavutil/samplefmt.h"

30 #include "tak.h"

31 #include "audiodsp.h"

32 #include "thread.h"

33 #include "avcodec.h"

34 #include "internal.h"

35 #include "unary.h"

36

37 #define MAX_SUBFRAMES 8 ///< max number of subframes per channel

38 #define MAX_PREDICTORS 256

39

40 typedef struct MCDParam {

41 int8_t present; ///< decorrelation parameter availability for this channel

42 int8_t index; ///< index into array of decorrelation types

43 int8_t chan1;

44 int8_t chan2;

45 } MCDParam;

46

47 typedef struct TAKDecContext {

48 AVCodecContext *avctx; ///< parent AVCodecContext

49 AudioDSPContext adsp;

50 TAKStreamInfo ti;

51 GetBitContext gb; ///< bitstream reader initialized to start at the current frame

52

53 int uval;

54 int nb_samples; ///< number of samples in the current frame

55 uint8_t *decode_buffer;

56 unsigned int decode_buffer_size;

57 int32_t *decoded[TAK_MAX_CHANNELS]; ///< decoded samples for each channel

58

59 int8_t lpc_mode[TAK_MAX_CHANNELS];

60 int8_t sample_shift[TAK_MAX_CHANNELS]; ///< shift applied to every sample in the channel

61 int16_t predictors[MAX_PREDICTORS];

62 int nb_subframes; ///< number of subframes in the current frame

63 int16_t subframe_len[MAX_SUBFRAMES]; ///< subframe length in samples

64 int subframe_scale;

65

66 int8_t dmode; ///< channel decorrelation type in the current frame

67

68 MCDParam mcdparams[TAK_MAX_CHANNELS]; ///< multichannel decorrelation parameters

69

70 int8_t coding_mode[128];

71 DECLARE_ALIGNED(16, int16_t, filter)[MAX_PREDICTORS];

72 DECLARE_ALIGNED(16, int16_t, residues)[544];

73 } TAKDecContext;

74

75 static const int8_t mc_dmodes[] = { 1, 3, 4, 6, };

76

77 static const uint16_t predictor_sizes[] = {

78 4, 8, 12, 16, 24, 32, 48, 64, 80, 96, 128, 160, 192, 224, 256, 0,

79 };

80

81 static const struct CParam {

82 int init;

83 int escape;

84 int scale;

85 int aescape;

86 int bias;

87 } xcodes[50] = {

88 { 0x01, 0x0000001, 0x0000001, 0x0000003, 0x0000008 },

89 { 0x02, 0x0000003, 0x0000001, 0x0000007, 0x0000006 },

90 { 0x03, 0x0000005, 0x0000002, 0x000000E, 0x000000D },

91 { 0x03, 0x0000003, 0x0000003, 0x000000D, 0x0000018 },

92 { 0x04, 0x000000B, 0x0000004, 0x000001C, 0x0000019 },

93 { 0x04, 0x0000006, 0x0000006, 0x000001A, 0x0000030 },

94 { 0x05, 0x0000016, 0x0000008, 0x0000038, 0x0000032 },

95 { 0x05, 0x000000C, 0x000000C, 0x0000034, 0x0000060 },

96 { 0x06, 0x000002C, 0x0000010, 0x0000070, 0x0000064 },

97 { 0x06, 0x0000018, 0x0000018, 0x0000068, 0x00000C0 },

98 { 0x07, 0x0000058, 0x0000020, 0x00000E0, 0x00000C8 },

99 { 0x07, 0x0000030, 0x0000030, 0x00000D0, 0x0000180 },

100 { 0x08, 0x00000B0, 0x0000040, 0x00001C0, 0x0000190 },

101 { 0x08, 0x0000060, 0x0000060, 0x00001A0, 0x0000300 },

102 { 0x09, 0x0000160, 0x0000080, 0x0000380, 0x0000320 },

103 { 0x09, 0x00000C0, 0x00000C0, 0x0000340, 0x0000600 },

104 { 0x0A, 0x00002C0, 0x0000100, 0x0000700, 0x0000640 },

105 { 0x0A, 0x0000180, 0x0000180, 0x0000680, 0x0000C00 },

106 { 0x0B, 0x0000580, 0x0000200, 0x0000E00, 0x0000C80 },

107 { 0x0B, 0x0000300, 0x0000300, 0x0000D00, 0x0001800 },

108 { 0x0C, 0x0000B00, 0x0000400, 0x0001C00, 0x0001900 },

109 { 0x0C, 0x0000600, 0x0000600, 0x0001A00, 0x0003000 },

110 { 0x0D, 0x0001600, 0x0000800, 0x0003800, 0x0003200 },

111 { 0x0D, 0x0000C00, 0x0000C00, 0x0003400, 0x0006000 },

112 { 0x0E, 0x0002C00, 0x0001000, 0x0007000, 0x0006400 },

113 { 0x0E, 0x0001800, 0x0001800, 0x0006800, 0x000C000 },

114 { 0x0F, 0x0005800, 0x0002000, 0x000E000, 0x000C800 },

115 { 0x0F, 0x0003000, 0x0003000, 0x000D000, 0x0018000 },

116 { 0x10, 0x000B000, 0x0004000, 0x001C000, 0x0019000 },

117 { 0x10, 0x0006000, 0x0006000, 0x001A000, 0x0030000 },

118 { 0x11, 0x0016000, 0x0008000, 0x0038000, 0x0032000 },

119 { 0x11, 0x000C000, 0x000C000, 0x0034000, 0x0060000 },

120 { 0x12, 0x002C000, 0x0010000, 0x0070000, 0x0064000 },

121 { 0x12, 0x0018000, 0x0018000, 0x0068000, 0x00C0000 },

122 { 0x13, 0x0058000, 0x0020000, 0x00E0000, 0x00C8000 },

123 { 0x13, 0x0030000, 0x0030000, 0x00D0000, 0x0180000 },

124 { 0x14, 0x00B0000, 0x0040000, 0x01C0000, 0x0190000 },

125 { 0x14, 0x0060000, 0x0060000, 0x01A0000, 0x0300000 },

126 { 0x15, 0x0160000, 0x0080000, 0x0380000, 0x0320000 },

127 { 0x15, 0x00C0000, 0x00C0000, 0x0340000, 0x0600000 },

128 { 0x16, 0x02C0000, 0x0100000, 0x0700000, 0x0640000 },

129 { 0x16, 0x0180000, 0x0180000, 0x0680000, 0x0C00000 },

130 { 0x17, 0x0580000, 0x0200000, 0x0E00000, 0x0C80000 },

131 { 0x17, 0x0300000, 0x0300000, 0x0D00000, 0x1800000 },

132 { 0x18, 0x0B00000, 0x0400000, 0x1C00000, 0x1900000 },

133 { 0x18, 0x0600000, 0x0600000, 0x1A00000, 0x3000000 },

134 { 0x19, 0x1600000, 0x0800000, 0x3800000, 0x3200000 },

135 { 0x19, 0x0C00000, 0x0C00000, 0x3400000, 0x6000000 },

136 { 0x1A, 0x2C00000, 0x1000000, 0x7000000, 0x6400000 },

137 { 0x1A, 0x1800000, 0x1800000, 0x6800000, 0xC000000 },

138 };

139

140 static int set_bps_params(AVCodecContext *avctx)

141 {

142 switch (avctx->bits_per_raw_sample) {

143 case 8:

144 avctx->sample_fmt = AV_SAMPLE_FMT_U8P;

145 break;

146 case 16:

147 avctx->sample_fmt = AV_SAMPLE_FMT_S16P;

148 break;

149 case 24:

150 avctx->sample_fmt = AV_SAMPLE_FMT_S32P;

151 break;

152 default:

153 av_log(avctx, AV_LOG_ERROR, "invalid/unsupported bits per sample: %d\n",

154 avctx->bits_per_raw_sample);

155 return AVERROR_INVALIDDATA;

156 }

157

158 return 0;

159 }

160

161 static void set_sample_rate_params(AVCodecContext *avctx)

162 {

163 TAKDecContext *s = avctx->priv_data;

164 int shift = 3 - (avctx->sample_rate / 11025);

165 shift = FFMAX(0, shift);

166 s->uval = FFALIGN(avctx->sample_rate + 511 >> 9, 4) << shift;

167 s->subframe_scale = FFALIGN(avctx->sample_rate + 511 >> 9, 4) << 1;

168 }

169

170 static av_cold int tak_decode_init(AVCodecContext *avctx)

171 {

172 TAKDecContext *s = avctx->priv_data;

173

174 ff_audiodsp_init(&s->adsp);

175

176 s->avctx = avctx;

177 avctx->bits_per_raw_sample = avctx->bits_per_coded_sample;

178

179 set_sample_rate_params(avctx);

180

181 return set_bps_params(avctx);

182 }

183

184 static void decode_lpc(int32_t *coeffs, int mode, int length)

185 {

186 int i;

187

188 if (length < 2)

189 return;

190

191 if (mode == 1) {

192 int a1 = *coeffs++;

193 for (i = 0; i < length - 1 >> 1; i++) {

194 *coeffs += a1;

195 coeffs[1] += *coeffs;

196 a1 = coeffs[1];

197 coeffs += 2;

198 }

199 if (length - 1 & 1)

200 *coeffs += a1;

201 } else if (mode == 2) {

202 int a1 = coeffs[1];

203 int a2 = a1 + *coeffs;

204 coeffs[1] = a2;

205 if (length > 2) {

206 coeffs += 2;

207 for (i = 0; i < length - 2 >> 1; i++) {

208 int a3 = *coeffs + a1;

209 int a4 = a3 + a2;

210 *coeffs = a4;

211 a1 = coeffs[1] + a3;

212 a2 = a1 + a4;

213 coeffs[1] = a2;

214 coeffs += 2;

215 }

216 if (length & 1)

217 *coeffs += a1 + a2;

218 }

219 } else if (mode == 3) {

220 int a1 = coeffs[1];

221 int a2 = a1 + *coeffs;

222 coeffs[1] = a2;

223 if (length > 2) {

224 int a3 = coeffs[2];

225 int a4 = a3 + a1;

226 int a5 = a4 + a2;

227 coeffs += 3;

228 for (i = 0; i < length - 3; i++) {

229 a3 += *coeffs;

230 a4 += a3;

231 a5 += a4;

232 *coeffs = a5;

233 coeffs++;

234 }

235 }

236 }

237 }

238

239 static int decode_segment(TAKDecContext *s, int8_t mode, int32_t *decoded, int len)

240 {

241 struct CParam code;

242 GetBitContext *gb = &s->gb;

243 int i;

244

245 if (!mode) {

246 memset(decoded, 0, len * sizeof(*decoded));

247 return 0;

248 }

249

250 if (mode > FF_ARRAY_ELEMS(xcodes))

251 return AVERROR_INVALIDDATA;

252 code = xcodes[mode - 1];

253

254 for (i = 0; i < len; i++) {

255 int x = get_bits_long(gb, code.init);

256 if (x >= code.escape && get_bits1(gb)) {

257 x |= 1 << code.init;

258 if (x >= code.aescape) {

259 int scale = get_unary(gb, 1, 9);

260 if (scale == 9) {

261 int scale_bits = get_bits(gb, 3);

262 if (scale_bits > 0) {

263 if (scale_bits == 7) {

264 scale_bits += get_bits(gb, 5);

265 if (scale_bits > 29)

266 return AVERROR_INVALIDDATA;

267 }

268 scale = get_bits_long(gb, scale_bits) + 1;

269 x += code.scale * scale;

270 }

271 x += code.bias;

272 } else

273 x += code.scale * scale - code.escape;

274 } else

275 x -= code.escape;

276 }

277 decoded[i] = (x >> 1) ^ -(x & 1);

278 }

279

280 return 0;

281 }

282

283 static int decode_residues(TAKDecContext *s, int32_t *decoded, int length)

284 {

285 GetBitContext *gb = &s->gb;

286 int i, mode, ret;

287

288 if (length > s->nb_samples)

289 return AVERROR_INVALIDDATA;

290

291 if (get_bits1(gb)) {

292 int wlength, rval;

293

294 wlength = length / s->uval;

295

296 rval = length - (wlength * s->uval);

297

298 if (rval < s->uval / 2)

299 rval += s->uval;

300 else

301 wlength++;

302

303 if (wlength <= 1 || wlength > 128)

304 return AVERROR_INVALIDDATA;

305

306 s->coding_mode[0] = mode = get_bits(gb, 6);

307

308 for (i = 1; i < wlength; i++) {

309 int c = get_unary(gb, 1, 6);

310

311 switch (c) {

312 case 6:

313 mode = get_bits(gb, 6);

314 break;

315 case 5:

316 case 4:

317 case 3: {

318 /* mode += sign ? (1 - c) : (c - 1) */

319 int sign = get_bits1(gb);

320 mode += (-sign ^ (c - 1)) + sign;

321 break;

322 }

323 case 2:

324 mode++;

325 break;

326 case 1:

327 mode--;

328 break;

329 }

330 s->coding_mode[i] = mode;

331 }

332

333 i = 0;

334 while (i < wlength) {

335 int len = 0;

336

337 mode = s->coding_mode[i];

338 do {

339 if (i >= wlength - 1)

340 len += rval;

341 else

342 len += s->uval;

343 i++;

344

345 if (i == wlength)

346 break;

347 } while (s->coding_mode[i] == mode);

348

349 if ((ret = decode_segment(s, mode, decoded, len)) < 0)

350 return ret;

351 decoded += len;

352 }

353 } else {

354 mode = get_bits(gb, 6);

355 if ((ret = decode_segment(s, mode, decoded, length)) < 0)

356 return ret;

357 }

358

359 return 0;

360 }

361

362 static int get_bits_esc4(GetBitContext *gb)

363 {

364 if (get_bits1(gb))

365 return get_bits(gb, 4) + 1;

366 else

367 return 0;

368 }

369

370 static int decode_subframe(TAKDecContext *s, int32_t *decoded,

371 int subframe_size, int prev_subframe_size)

372 {

373 GetBitContext *gb = &s->gb;

374 int x, y, i, j, ret = 0;

375 int dshift, size, filter_quant, filter_order;

376 int tfilter[MAX_PREDICTORS];

377

378 if (!get_bits1(gb))

379 return decode_residues(s, decoded, subframe_size);

380

381 filter_order = predictor_sizes[get_bits(gb, 4)];

382

383 if (prev_subframe_size > 0 && get_bits1(gb)) {

384 if (filter_order > prev_subframe_size)

385 return AVERROR_INVALIDDATA;

386

387 decoded -= filter_order;

388 subframe_size += filter_order;

389

390 if (filter_order > subframe_size)

391 return AVERROR_INVALIDDATA;

392 } else {

393 int lpc_mode;

394

395 if (filter_order > subframe_size)

396 return AVERROR_INVALIDDATA;

397

398 lpc_mode = get_bits(gb, 2);

399 if (lpc_mode > 2)

400 return AVERROR_INVALIDDATA;

401

402 if ((ret = decode_residues(s, decoded, filter_order)) < 0)

403 return ret;

404

405 if (lpc_mode)

406 decode_lpc(decoded, lpc_mode, filter_order);

407 }

408

409 dshift = get_bits_esc4(gb);

410 size = get_bits1(gb) + 6;

411

412 filter_quant = 10;

413 if (get_bits1(gb)) {

414 filter_quant -= get_bits(gb, 3) + 1;

415 if (filter_quant < 3)

416 return AVERROR_INVALIDDATA;

417 }

418

419 s->predictors[0] = get_sbits(gb, 10);

420 s->predictors[1] = get_sbits(gb, 10);

421 s->predictors[2] = get_sbits(gb, size) << (10 - size);

422 s->predictors[3] = get_sbits(gb, size) << (10 - size);

423 if (filter_order > 4) {

424 int tmp = size - get_bits1(gb);

425

426 for (i = 4; i < filter_order; i++) {

427 if (!(i & 3))

428 x = tmp - get_bits(gb, 2);

429 s->predictors[i] = get_sbits(gb, x) << (10 - size);

430 }

431 }

432

433 tfilter[0] = s->predictors[0] << 6;

434 for (i = 1; i < filter_order; i++) {

435 int32_t *p1 = &tfilter[0];

436 int32_t *p2 = &tfilter[i - 1];

437

438 for (j = 0; j < (i + 1) / 2; j++) {

439 x = *p1 + (s->predictors[i] * *p2 + 256 >> 9);

440 *p2 += s->predictors[i] * *p1 + 256 >> 9;

441 *p1++ = x;

442 p2--;

443 }

444

445 tfilter[i] = s->predictors[i] << 6;

446 }

447

448 x = 1 << (32 - (15 - filter_quant));

449 y = 1 << ((15 - filter_quant) - 1);

450 for (i = 0, j = filter_order - 1; i < filter_order / 2; i++, j--) {

451 s->filter[j] = x - ((tfilter[i] + y) >> (15 - filter_quant));

452 s->filter[i] = x - ((tfilter[j] + y) >> (15 - filter_quant));

453 }

454

455 if ((ret = decode_residues(s, &decoded[filter_order],

456 subframe_size - filter_order)) < 0)

457 return ret;

458

459 for (i = 0; i < filter_order; i++)

460 s->residues[i] = *decoded++ >> dshift;

461

462 y = FF_ARRAY_ELEMS(s->residues) - filter_order;

463 x = subframe_size - filter_order;

464 while (x > 0) {

465 int tmp = FFMIN(y, x);

466

467 for (i = 0; i < tmp; i++) {

468 int v = 1 << (filter_quant - 1);

469

470 if (filter_order & -16)

471 v += s->adsp.scalarproduct_int16(&s->residues[i], s->filter,

472 filter_order & -16);

473 for (j = filter_order & -16; j < filter_order; j += 4) {

474 v += s->residues[i + j + 3] * s->filter[j + 3] +

475 s->residues[i + j + 2] * s->filter[j + 2] +

476 s->residues[i + j + 1] * s->filter[j + 1] +

477 s->residues[i + j ] * s->filter[j ];

478 }

479 v = (av_clip_intp2(v >> filter_quant, 13) << dshift) - *decoded;

480 *decoded++ = v;

481 s->residues[filter_order + i] = v >> dshift;

482 }

483

484 x -= tmp;

485 if (x > 0)

486 memcpy(s->residues, &s->residues[y], 2 * filter_order);

487 }

488

489 emms_c();

490

491 return 0;

492 }

493

494 static int decode_channel(TAKDecContext *s, int chan)

495 {

496 AVCodecContext *avctx = s->avctx;

497 GetBitContext *gb = &s->gb;

498 int32_t *decoded = s->decoded[chan];

499 int left = s->nb_samples - 1;

500 int i = 0, ret, prev = 0;

501

502 s->sample_shift[chan] = get_bits_esc4(gb);

503 if (s->sample_shift[chan] >= avctx->bits_per_raw_sample)

504 return AVERROR_INVALIDDATA;

505

506 *decoded++ = get_sbits(gb, avctx->bits_per_raw_sample - s->sample_shift[chan]);

507 s->lpc_mode[chan] = get_bits(gb, 2);

508 s->nb_subframes = get_bits(gb, 3) + 1;

509

510 if (s->nb_subframes > 1) {

511 if (get_bits_left(gb) < (s->nb_subframes - 1) * 6)

512 return AVERROR_INVALIDDATA;

513

514 for (; i < s->nb_subframes - 1; i++) {

515 int v = get_bits(gb, 6);

516

517 s->subframe_len[i] = (v - prev) * s->subframe_scale;

518 if (s->subframe_len[i] <= 0)

519 return AVERROR_INVALIDDATA;

520

521 left -= s->subframe_len[i];

522 prev = v;

523 }

524

525 if (left <= 0)

526 return AVERROR_INVALIDDATA;

527 }

528 s->subframe_len[i] = left;

529

530 prev = 0;

531 for (i = 0; i < s->nb_subframes; i++) {

532 if ((ret = decode_subframe(s, decoded, s->subframe_len[i], prev)) < 0)

533 return ret;

534 decoded += s->subframe_len[i];

535 prev = s->subframe_len[i];

536 }

537

538 return 0;

539 }

540

541 static int decorrelate(TAKDecContext *s, int c1, int c2, int length)

542 {

543 GetBitContext *gb = &s->gb;

544 int32_t *p1 = s->decoded[c1] + 1;

545 int32_t *p2 = s->decoded[c2] + 1;

546 int i;

547 int dshift, dfactor;

548

549 switch (s->dmode) {

550 case 1: /* left/side */

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

552 int32_t a = p1[i];

553 int32_t b = p2[i];

554 p2[i] = a + b;

555 }

556 break;

557 case 2: /* side/right */

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

559 int32_t a = p1[i];

560 int32_t b = p2[i];

561 p1[i] = b - a;

562 }

563 break;

564 case 3: /* side/mid */

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

566 int32_t a = p1[i];

567 int32_t b = p2[i];

568 a -= b >> 1;

569 p1[i] = a;

570 p2[i] = a + b;

571 }

572 break;

573 case 4: /* side/left with scale factor */

574 FFSWAP(int32_t*, p1, p2);

575 case 5: /* side/right with scale factor */

576 dshift = get_bits_esc4(gb);

577 dfactor = get_sbits(gb, 10);

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

579 int32_t a = p1[i];

580 int32_t b = p2[i];

581 b = dfactor * (b >> dshift) + 128 >> 8 << dshift;

582 p1[i] = b - a;

583 }

584 break;

585 case 6:

586 FFSWAP(int32_t*, p1, p2);

587 case 7: {

588 int length2, order_half, filter_order, dval1, dval2;

589 int tmp, x, code_size;

590

591 if (length < 256)

592 return AVERROR_INVALIDDATA;

593

594 dshift = get_bits_esc4(gb);

595 filter_order = 8 << get_bits1(gb);

596 dval1 = get_bits1(gb);

597 dval2 = get_bits1(gb);

598

599 for (i = 0; i < filter_order; i++) {

600 if (!(i & 3))

601 code_size = 14 - get_bits(gb, 3);

602 s->filter[i] = get_sbits(gb, code_size);

603 }

604

605 order_half = filter_order / 2;

606 length2 = length - (filter_order - 1);

607

608 /* decorrelate beginning samples */

609 if (dval1) {

610 for (i = 0; i < order_half; i++) {

611 int32_t a = p1[i];

612 int32_t b = p2[i];

613 p1[i] = a + b;

614 }

615 }

616

617 /* decorrelate ending samples */

618 if (dval2) {

619 for (i = length2 + order_half; i < length; i++) {

620 int32_t a = p1[i];

621 int32_t b = p2[i];

622 p1[i] = a + b;

623 }

624 }

625

626

627 for (i = 0; i < filter_order; i++)

628 s->residues[i] = *p2++ >> dshift;

629

630 p1 += order_half;

631 x = FF_ARRAY_ELEMS(s->residues) - filter_order;

632 for (; length2 > 0; length2 -= tmp) {

633 tmp = FFMIN(length2, x);

634

635 for (i = 0; i < tmp; i++)

636 s->residues[filter_order + i] = *p2++ >> dshift;

637

638 for (i = 0; i < tmp; i++) {

639 int v = 1 << 9;

640

641 if (filter_order == 16) {

642 v += s->adsp.scalarproduct_int16(&s->residues[i], s->filter,

643 filter_order);

644 } else {

645 v += s->residues[i + 7] * s->filter[7] +

646 s->residues[i + 6] * s->filter[6] +

647 s->residues[i + 5] * s->filter[5] +

648 s->residues[i + 4] * s->filter[4] +

649 s->residues[i + 3] * s->filter[3] +

650 s->residues[i + 2] * s->filter[2] +

651 s->residues[i + 1] * s->filter[1] +

652 s->residues[i ] * s->filter[0];

653 }

654

655 v = (av_clip_intp2(v >> 10, 13) << dshift) - *p1;

656 *p1++ = v;

657 }

658

659 memcpy(s->residues, &s->residues[tmp], 2 * filter_order);

660 }

661

662 emms_c();

663 break;

664 }

665 }

666

667 return 0;

668 }

669

670 static int tak_decode_frame(AVCodecContext *avctx, void *data,

671 int *got_frame_ptr, AVPacket *pkt)

672 {

673 TAKDecContext *s = avctx->priv_data;

674 AVFrame *frame = data;

675 ThreadFrame tframe = { .f = data };

676 GetBitContext *gb = &s->gb;

677 int chan, i, ret, hsize;

678

679 if (pkt->size < TAK_MIN_FRAME_HEADER_BYTES)

680 return AVERROR_INVALIDDATA;

681

682 if ((ret = init_get_bits8(gb, pkt->data, pkt->size)) < 0)

683 return ret;

684

685 if ((ret = ff_tak_decode_frame_header(avctx, gb, &s->ti, 0)) < 0)

686 return ret;

687

688 hsize = get_bits_count(gb) / 8;

689 if (avctx->err_recognition & (AV_EF_CRCCHECK|AV_EF_COMPLIANT)) {

690 if (ff_tak_check_crc(pkt->data, hsize)) {

691 av_log(avctx, AV_LOG_ERROR, "CRC error\n");

692 if (avctx->err_recognition & AV_EF_EXPLODE)

693 return AVERROR_INVALIDDATA;

694 }

695 }

696

697 if (s->ti.codec != TAK_CODEC_MONO_STEREO &&

698 s->ti.codec != TAK_CODEC_MULTICHANNEL) {

699 av_log(avctx, AV_LOG_ERROR, "unsupported codec: %d\n", s->ti.codec);

700 return AVERROR_PATCHWELCOME;

701 }

702 if (s->ti.data_type) {

703 av_log(avctx, AV_LOG_ERROR,

704 "unsupported data type: %d\n", s->ti.data_type);

705 return AVERROR_INVALIDDATA;

706 }

707 if (s->ti.codec == TAK_CODEC_MONO_STEREO && s->ti.channels > 2) {

708 av_log(avctx, AV_LOG_ERROR,

709 "invalid number of channels: %d\n", s->ti.channels);

710 return AVERROR_INVALIDDATA;

711 }

712 if (s->ti.channels > 6) {

713 av_log(avctx, AV_LOG_ERROR,

714 "unsupported number of channels: %d\n", s->ti.channels);

715 return AVERROR_INVALIDDATA;

716 }

717

718 if (s->ti.frame_samples <= 0) {

719 av_log(avctx, AV_LOG_ERROR, "unsupported/invalid number of samples\n");

720 return AVERROR_INVALIDDATA;

721 }

722

723 avctx->bits_per_raw_sample = s->ti.bps;

724 if ((ret = set_bps_params(avctx)) < 0)

725 return ret;

726 if (s->ti.sample_rate != avctx->sample_rate) {

727 avctx->sample_rate = s->ti.sample_rate;

728 set_sample_rate_params(avctx);

729 }

730 if (s->ti.ch_layout)

731 avctx->channel_layout = s->ti.ch_layout;

732 avctx->channels = s->ti.channels;

733

734 s->nb_samples = s->ti.last_frame_samples ? s->ti.last_frame_samples

735 : s->ti.frame_samples;

736

737 frame->nb_samples = s->nb_samples;

738 if ((ret = ff_thread_get_buffer(avctx, &tframe, 0)) < 0)

739 return ret;

740 ff_thread_finish_setup(avctx);

741

742 if (avctx->bits_per_raw_sample <= 16) {

743 int buf_size = av_samples_get_buffer_size(NULL, avctx->channels,

744 s->nb_samples,

745 AV_SAMPLE_FMT_S32P, 0);

746 if (buf_size < 0)

747 return buf_size;

748 av_fast_malloc(&s->decode_buffer, &s->decode_buffer_size, buf_size);

749 if (!s->decode_buffer)

750 return AVERROR(ENOMEM);

751 ret = av_samples_fill_arrays((uint8_t **)s->decoded, NULL,

752 s->decode_buffer, avctx->channels,

753 s->nb_samples, AV_SAMPLE_FMT_S32P, 0);

754 if (ret < 0)

755 return ret;

756 } else {

757 for (chan = 0; chan < avctx->channels; chan++)

758 s->decoded[chan] = (int32_t *)frame->extended_data[chan];

759 }

760

761 if (s->nb_samples < 16) {

762 for (chan = 0; chan < avctx->channels; chan++) {

763 int32_t *decoded = s->decoded[chan];

764 for (i = 0; i < s->nb_samples; i++)

765 decoded[i] = get_sbits(gb, avctx->bits_per_raw_sample);

766 }

767 } else {

768 if (s->ti.codec == TAK_CODEC_MONO_STEREO) {

769 for (chan = 0; chan < avctx->channels; chan++)

770 if (ret = decode_channel(s, chan))

771 return ret;

772

773 if (avctx->channels == 2) {

774 s->nb_subframes = get_bits(gb, 1) + 1;

775 if (s->nb_subframes > 1) {

776 s->subframe_len[1] = get_bits(gb, 6);

777 }

778

779 s->dmode = get_bits(gb, 3);

780 if (ret = decorrelate(s, 0, 1, s->nb_samples - 1))

781 return ret;

782 }

783 } else if (s->ti.codec == TAK_CODEC_MULTICHANNEL) {

784 if (get_bits1(gb)) {

785 int ch_mask = 0;

786

787 chan = get_bits(gb, 4) + 1;

788 if (chan > avctx->channels)

789 return AVERROR_INVALIDDATA;

790

791 for (i = 0; i < chan; i++) {

792 int nbit = get_bits(gb, 4);

793

794 if (nbit >= avctx->channels)

795 return AVERROR_INVALIDDATA;

796

797 if (ch_mask & 1 << nbit)

798 return AVERROR_INVALIDDATA;

799

800 s->mcdparams[i].present = get_bits1(gb);

801 if (s->mcdparams[i].present) {

802 s->mcdparams[i].index = get_bits(gb, 2);

803 s->mcdparams[i].chan2 = get_bits(gb, 4);

804 if (s->mcdparams[i].index == 1) {

805 if ((nbit == s->mcdparams[i].chan2) ||

806 (ch_mask & 1 << s->mcdparams[i].chan2))

807 return AVERROR_INVALIDDATA;

808

809 ch_mask |= 1 << s->mcdparams[i].chan2;

810 } else if (!(ch_mask & 1 << s->mcdparams[i].chan2)) {

811 return AVERROR_INVALIDDATA;

812 }

813 }

814 s->mcdparams[i].chan1 = nbit;

815

816 ch_mask |= 1 << nbit;

817 }

818 } else {

819 chan = avctx->channels;

820 for (i = 0; i < chan; i++) {

821 s->mcdparams[i].present = 0;

822 s->mcdparams[i].chan1 = i;

823 }

824 }

825

826 for (i = 0; i < chan; i++) {

827 if (s->mcdparams[i].present && s->mcdparams[i].index == 1)

828 if (ret = decode_channel(s, s->mcdparams[i].chan2))

829 return ret;

830

831 if (ret = decode_channel(s, s->mcdparams[i].chan1))

832 return ret;

833

834 if (s->mcdparams[i].present) {

835 s->dmode = mc_dmodes[s->mcdparams[i].index];

836 if (ret = decorrelate(s,

837 s->mcdparams[i].chan2,

838 s->mcdparams[i].chan1,

839 s->nb_samples - 1))

840 return ret;

841 }

842 }

843 }

844

845 for (chan = 0; chan < avctx->channels; chan++) {

846 int32_t *decoded = s->decoded[chan];

847

848 if (s->lpc_mode[chan])

849 decode_lpc(decoded, s->lpc_mode[chan], s->nb_samples);

850

851 if (s->sample_shift[chan] > 0)

852 for (i = 0; i < s->nb_samples; i++)

853 decoded[i] <<= s->sample_shift[chan];

854 }

855 }

856

857 align_get_bits(gb);

858 skip_bits(gb, 24);

859 if (get_bits_left(gb) < 0)

860 av_log(avctx, AV_LOG_DEBUG, "overread\n");

861 else if (get_bits_left(gb) > 0)

862 av_log(avctx, AV_LOG_DEBUG, "underread\n");

863

864 if (avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_COMPLIANT)) {

865 if (ff_tak_check_crc(pkt->data + hsize,

866 get_bits_count(gb) / 8 - hsize)) {

867 av_log(avctx, AV_LOG_ERROR, "CRC error\n");

868 if (avctx->err_recognition & AV_EF_EXPLODE)

869 return AVERROR_INVALIDDATA;

870 }

871 }

872

873 /* convert to output buffer */

874 switch (avctx->sample_fmt) {

875 case AV_SAMPLE_FMT_U8P:

876 for (chan = 0; chan < avctx->channels; chan++) {

877 uint8_t *samples = (uint8_t *)frame->extended_data[chan];

878 int32_t *decoded = s->decoded[chan];

879 for (i = 0; i < s->nb_samples; i++)

880 samples[i] = decoded[i] + 0x80;

881 }

882 break;

883 case AV_SAMPLE_FMT_S16P:

884 for (chan = 0; chan < avctx->channels; chan++) {

885 int16_t *samples = (int16_t *)frame->extended_data[chan];

886 int32_t *decoded = s->decoded[chan];

887 for (i = 0; i < s->nb_samples; i++)

888 samples[i] = decoded[i];

889 }

890 break;

891 case AV_SAMPLE_FMT_S32P:

892 for (chan = 0; chan < avctx->channels; chan++) {

893 int32_t *samples = (int32_t *)frame->extended_data[chan];

894 for (i = 0; i < s->nb_samples; i++)

895 samples[i] <<= 8;

896 }

897 break;

898 }

899

900 *got_frame_ptr = 1;

901

902 return pkt->size;

903 }

904

905 static int init_thread_copy(AVCodecContext *avctx)

906 {

907 TAKDecContext *s = avctx->priv_data;

908 s->avctx = avctx;

909 return 0;

910 }

911

912 static int update_thread_context(AVCodecContext *dst,

913 const AVCodecContext *src)

914 {

915 TAKDecContext *tsrc = src->priv_data;

916 TAKDecContext *tdst = dst->priv_data;

917

918 if (dst == src)

919 return 0;

920 memcpy(&tdst->ti, &tsrc->ti, sizeof(TAKStreamInfo));

921 return 0;

922 }

923

924 static av_cold int tak_decode_close(AVCodecContext *avctx)

925 {

926 TAKDecContext *s = avctx->priv_data;

927

928 av_freep(&s->decode_buffer);

929

930 return 0;

931 }

932

933 AVCodec ff_tak_decoder = {

934 .name = "tak",

935 .long_name = NULL_IF_CONFIG_SMALL("TAK (Tom's lossless Audio Kompressor)"),

936 .type = AVMEDIA_TYPE_AUDIO,

937 .id = AV_CODEC_ID_TAK,

938 .priv_data_size = sizeof(TAKDecContext),

939 .init = tak_decode_init,

940 .close = tak_decode_close,

941 .decode = tak_decode_frame,

942 .init_thread_copy = ONLY_IF_THREADS_ENABLED(init_thread_copy),

943 .update_thread_context = ONLY_IF_THREADS_ENABLED(update_thread_context),

944 .capabilities = CODEC_CAP_DR1 | CODEC_CAP_FRAME_THREADS,

945 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,

946 AV_SAMPLE_FMT_S16P,

947 AV_SAMPLE_FMT_S32P,

948 AV_SAMPLE_FMT_NONE },

949 };

TAK_MAX_CHANNELS

#define TAK_MAX_CHANNELS

Definition: tak.h:65

#define NULL

Definition: coverity.c:32

v

float v

Definition: avf_showspectrum.c:96

static int set_bps_params(AVCodecContext *avctx)

Definition: takdec.c:140

s

const char * s

Definition: avisynth_c.h:631

predictor_sizes

static const uint16_t predictor_sizes[]

Definition: takdec.c:77

AVERROR_INVALIDDATA

#define AVERROR_INVALIDDATA

Invalid data found when processing input.

Definition: error.h:59

ff_tak_decode_frame_header

int ff_tak_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb, TAKStreamInfo *ti, int log_level_offset)

Validate and decode a frame header.

Definition: tak.c:126

AudioDSPContext::scalarproduct_int16

int32_t(* scalarproduct_int16)(const int16_t *v1, const int16_t *v2, int len)

Calculate scalar product of two vectors.

Definition: audiodsp.h:29

static int shift(int a, int b)

Definition: sonic.c:82

TAKDecContext::subframe_len

int16_t subframe_len[MAX_SUBFRAMES]

subframe length in samples

Definition: takdec.c:63

This structure describes decoded (raw) audio or video data.

Definition: frame.h:171

TAKStreamInfo::channels

int channels

Definition: tak.h:134

ptrdiff_t const GLvoid * data

Definition: opengl_enc.c:101

static unsigned int get_bits(GetBitContext *s, int n)

Read 1-25 bits.

Definition: get_bits.h:260

AVFrame * f

Definition: thread.h:36

CParam::aescape

int aescape

Definition: takdec.c:85

static av_cold int init(AVCodecContext *avctx)

Definition: avrndec.c:35

TAKDecContext::mcdparams

MCDParam mcdparams[TAK_MAX_CHANNELS]

multichannel decorrelation parameters

Definition: takdec.c:68

TAKDecContext::filter

int16_t filter[MAX_PREDICTORS]

Definition: takdec.c:71

int size

Definition: avcodec.h:1163

b

const char * b

Definition: vf_curves.c:109

AV_SAMPLE_FMT_NONE

Definition: samplefmt.h:60

decode_subframe

static int decode_subframe(TAKDecContext *s, int32_t *decoded, int subframe_size, int prev_subframe_size)

Definition: takdec.c:370

DECLARE_ALIGNED

#define DECLARE_ALIGNED(n, t, v)

Definition: mem.h:53

ff_audiodsp_init

av_cold void ff_audiodsp_init(AudioDSPContext *c)

Definition: audiodsp.c:106

TAKStreamInfo::ch_layout

uint64_t ch_layout

Definition: tak.h:139

#define a1

Definition: regdef.h:47

#define FF_ARRAY_ELEMS(a)

Definition: sinewin_tablegen.c:28

static AVPacket pkt

Definition: demuxing_decoding.c:54

AVCodecContext::bits_per_raw_sample

int bits_per_raw_sample

Bits per sample/pixel of internal libavcodec pixel/sample format.

Definition: avcodec.h:2727

static int decode_segment(TAKDecContext *s, int8_t mode, int32_t *decoded, int len)

Definition: takdec.c:239

AVCodec.

Definition: avcodec.h:3181

static int get_sbits(GetBitContext *s, int n)

Definition: get_bits.h:245

#define FFALIGN(x, a)

Definition: common.h:71

TAKDecContext::lpc_mode

int8_t lpc_mode[TAK_MAX_CHANNELS]

Definition: takdec.c:59

TAKDecContext::decode_buffer_size

unsigned int decode_buffer_size

Definition: takdec.c:56

#define a3

Definition: regdef.h:49

if()

Definition: avfilter.c:975

int init

Definition: takdec.c:82

AVCodecContext::sample_fmt

enum AVSampleFormat sample_fmt

audio sample format

Definition: avcodec.h:1993

uint8_t

Definition: audio_convert.c:194

#define av_cold

Definition: attributes.h:74

mode

Definition: f_perms.c:27

tak_decode_close

static av_cold int tak_decode_close(AVCodecContext *avctx)

Definition: takdec.c:924

Multithreading support functions.

ff_tak_check_crc

int ff_tak_check_crc(const uint8_t *buf, unsigned int buf_size)

Definition: tak.c:75

init_thread_copy

static int init_thread_copy(AVCodecContext *avctx)

Definition: takdec.c:905

TAK_CODEC_MULTICHANNEL

Definition: tak.h:102

#define CODEC_CAP_DR1

Codec uses get_buffer() for allocating buffers and supports custom allocators.

Definition: avcodec.h:789

static AVFrame * frame

Definition: demuxing_decoding.c:53

TAKDecContext::avctx

AVCodecContext * avctx

parent AVCodecContext

Definition: takdec.c:48

uint8_t * data

Definition: avcodec.h:1162

static int get_bits_count(const GetBitContext *s)

Definition: get_bits.h:212

static const uint64_t c1

Definition: murmur3.c:49

int escape

Definition: takdec.c:83

ptrdiff_t size

Definition: opengl_enc.c:101

AVCodecContext::bits_per_coded_sample

int bits_per_coded_sample

bits per sample/pixel from the demuxer (needed for huffyuv).

Definition: avcodec.h:2720

ff_thread_finish_setup

void ff_thread_finish_setup(AVCodecContext *avctx)

If the codec defines update_thread_context(), call this when they are ready for the next thread to st...

Definition: pthread_frame.c:514

TAKStreamInfo::data_type

int data_type

Definition: tak.h:132

static int decode_channel(TAKDecContext *s, int chan)

Definition: takdec.c:494

#define av_log(a,...)

Definition: tableprint_vlc.h:28

#define MAX_SUBFRAMES

max number of subframes per channel

Definition: takdec.c:37

MCDParam::index

int8_t index

index into array of decorrelation types

Definition: takdec.c:42

static int get_bits_left(GetBitContext *gb)

Definition: get_bits.h:588

#define AV_LOG_ERROR

Something went wrong and cannot losslessly be recovered.

Definition: log.h:176

#define AV_EF_EXPLODE

abort decoding on minor error detection

Definition: avcodec.h:2623

TAKDecContext::uval

int uval

Definition: takdec.c:53

#define AVERROR(e)

Definition: error.h:43

NULL_IF_CONFIG_SMALL

#define NULL_IF_CONFIG_SMALL(x)

Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...

Definition: internal.h:175

#define AV_LOG_DEBUG

Stuff which is only useful for libav* developers.

Definition: log.h:197

AVMEDIA_TYPE_AUDIO

Definition: avutil.h:195

GLsizei GLsizei * length

Definition: opengl_enc.c:115

const char * name

Name of the codec implementation.

Definition: avcodec.h:3188

MCDParam::present

int8_t present

decorrelation parameter availability for this channel

Definition: takdec.c:41

TAKStreamInfo::last_frame_samples

int last_frame_samples

Definition: tak.h:138

decode_residues

static int decode_residues(TAKDecContext *s, int32_t *decoded, int length)

Definition: takdec.c:283

#define FFMAX(a, b)

Definition: common.h:64

Libavcodec external API header.

AVCodecContext::channel_layout

uint64_t channel_layout

Audio channel layout.

Definition: avcodec.h:2046

ONLY_IF_THREADS_ENABLED

#define ONLY_IF_THREADS_ENABLED(x)

Define a function with only the non-default version specified.

Definition: internal.h:214

TAKStreamInfo::bps

int bps

Definition: tak.h:135

common internal API header

TAKDecContext::coding_mode

int8_t coding_mode[128]

Definition: takdec.c:70

update_thread_context

static int update_thread_context(AVCodecContext *dst, const AVCodecContext *src)

Definition: takdec.c:912

tak_decode_init

static av_cold int tak_decode_init(AVCodecContext *avctx)

Definition: takdec.c:170

MCDParam::chan2

int8_t chan2

Definition: takdec.c:44

AVCodecContext::err_recognition

int err_recognition

Error recognition; may misdetect some more or less valid parts as errors.

Definition: avcodec.h:2612

#define FFMIN(a, b)

Definition: common.h:66

y

float y

Definition: avf_showspectrum.c:96

AV_SAMPLE_FMT_S32P

signed 32 bits, planar

Definition: samplefmt.h:69

static const struct CParam xcodes[50]

ret

Definition: avfilter.c:974

int scale

Definition: takdec.c:84

int32_t

Definition: audio_convert.c:194

#define a2

Definition: regdef.h:48

static int get_bits_esc4(GetBitContext *gb)

Definition: takdec.c:362

TAK_CODEC_MONO_STEREO

Definition: tak.h:101

AV_SAMPLE_FMT_U8P

unsigned 8 bits, planar

Definition: samplefmt.h:67

TAKDecContext::adsp

AudioDSPContext adsp

Definition: takdec.c:49

Definition: thread.h:35

TAKDecContext::decode_buffer

uint8_t * decode_buffer

Definition: takdec.c:55

static int decorrelate(TAKDecContext *s, int c1, int c2, int length)

Definition: takdec.c:541

static void decode_lpc(int32_t *coeffs, int mode, int length)

Definition: takdec.c:184

AVERROR_PATCHWELCOME

#define AVERROR_PATCHWELCOME

Not yet implemented in FFmpeg, patches welcome.

Definition: error.h:62

AVCodec ff_tak_decoder

Definition: takdec.c:933

Definition: takdec.c:81

Definition: tak.h:129

#define a5

Definition: regdef.h:51

AVS_Value src

Definition: avisynth_c.h:482

AVSampleFormat

Audio sample formats.

Definition: samplefmt.h:59

TAK_MIN_FRAME_HEADER_BYTES

#define TAK_MIN_FRAME_HEADER_BYTES

Definition: tak.h:98

AVCodecContext::sample_rate

int sample_rate

samples per second

Definition: avcodec.h:1985

TAKDecContext::predictors

int16_t predictors[MAX_PREDICTORS]

Definition: takdec.c:61

static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)

Initialize GetBitContext.

Definition: get_bits.h:441

ff_thread_get_buffer

int ff_thread_get_buffer(AVCodecContext *avctx, ThreadFrame *f, int flags)

Wrapper around get_buffer() for frame-multithreaded codecs.

Definition: pthread_frame.c:853

main external API structure.

Definition: avcodec.h:1241

TAKStreamInfo::codec

enum TAKCodecType codec

Definition: tak.h:131

TAKDecContext::decoded

int32_t * decoded[TAK_MAX_CHANNELS]

decoded samples for each channel

Definition: takdec.c:57

static unsigned int get_bits1(GetBitContext *s)

Definition: get_bits.h:304

static void skip_bits(GetBitContext *s, int n)

Definition: get_bits.h:297

TAKStreamInfo::frame_samples

int frame_samples

Definition: tak.h:137

av_samples_get_buffer_size

int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align)

Get the required buffer size for the given audio parameters.

Definition: samplefmt.c:117

TAK (Tom's lossless Audio Kompressor) decoder/demuxer common functions.

TAKDecContext::gb

GetBitContext gb

bitstream reader initialized to start at the current frame

Definition: takdec.c:51

TAKDecContext::dmode

int8_t dmode

channel decorrelation type in the current frame

Definition: takdec.c:66

Definition: takdec.c:40

static unsigned int get_bits_long(GetBitContext *s, int n)

Read 0-32 bits.

Definition: get_bits.h:337

TAKStreamInfo::sample_rate

int sample_rate

Definition: tak.h:133

static int decode(AVCodecContext *avctx, void *data, int *got_sub, AVPacket *avpkt)

Definition: ccaption_dec.c:522

int bias

Definition: takdec.c:86

TAKDecContext::nb_samples

int nb_samples

number of samples in the current frame

Definition: takdec.c:54

void av_fast_malloc(void *ptr, unsigned int *size, size_t min_size)

Allocate a buffer, reusing the given one if large enough.

Definition: mem.c:513

TAKDecContext::ti

TAKStreamInfo ti

Definition: takdec.c:50

Definition: get_bits.h:54

common internal api header.

CODEC_CAP_FRAME_THREADS

#define CODEC_CAP_FRAME_THREADS

Codec supports frame-level multithreading.

Definition: avcodec.h:866

MCDParam::chan1

int8_t chan1

Definition: takdec.c:43

static int get_unary(GetBitContext *gb, int stop, int len)

Get unary code of limited length.

Definition: unary.h:33

c

static double c[64]

Definition: vsrc_mptestsrc.c:87

TAKDecContext::residues

int16_t residues[544]

Definition: takdec.c:72

static const uint64_t c2

Definition: murmur3.c:50

static const int8_t mc_dmodes[]

Definition: takdec.c:75

TAKDecContext::subframe_scale

int subframe_scale

Definition: takdec.c:64

#define AV_EF_CRCCHECK

Verify checksums embedded in the bitstream (could be of either encoded or decoded data...

Definition: avcodec.h:2620

TAKDecContext::nb_subframes

int nb_subframes

number of subframes in the current frame

Definition: takdec.c:62

#define a4

Definition: regdef.h:50

AVCodecContext::priv_data

void * priv_data

Definition: avcodec.h:1283

AV_CODEC_ID_TAK

Definition: avcodec.h:495

av_samples_fill_arrays

int av_samples_fill_arrays(uint8_t **audio_data, int *linesize, const uint8_t *buf, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align)

Fill plane data pointers and linesize for samples with sample format sample_fmt.

Definition: samplefmt.c:149

tak_decode_frame

static int tak_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *pkt)

Definition: takdec.c:670

int len

Definition: vorbis_enc_data.h:452

AVCodecContext::channels

int channels

number of audio channels

Definition: avcodec.h:1986

AV_EF_COMPLIANT

#define AV_EF_COMPLIANT

consider all spec non compliances as errors

Definition: avcodec.h:2627

static const uint8_t * align_get_bits(GetBitContext *s)

Definition: get_bits.h:449

#define MAX_PREDICTORS

Definition: takdec.c:38

Definition: takdec.c:47

set_sample_rate_params

static void set_sample_rate_params(AVCodecContext *avctx)

Definition: takdec.c:161

#define av_freep(p)

Definition: tableprint_vlc.h:35

TAKDecContext::sample_shift

int8_t sample_shift[TAK_MAX_CHANNELS]

shift applied to every sample in the channel

Definition: takdec.c:60

AV_SAMPLE_FMT_S16P

signed 16 bits, planar

Definition: samplefmt.h:68

#define FFSWAP(type, a, b)

Definition: common.h:69

AVFrame::extended_data

uint8_t ** extended_data

pointers to the data planes/channels.

Definition: frame.h:215

This structure stores compressed data.

Definition: avcodec.h:1139

AVFrame::nb_samples

int nb_samples

number of audio samples (per channel) described by this frame

Definition: frame.h:225

for(j=16;j >0;--j)

Definition: h264pred_template.c:469

a

a

Definition: h264pred_template.c:468

AudioDSPContext

Definition: audiodsp.h:24

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