FFmpeg: libavcodec/atrac3.c Source File

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atrac3.c

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1 /*

2 * Atrac 3 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 * Atrac 3 compatible decoder.

26 * This decoder handles Sony's ATRAC3 data.

27 *

28 * Container formats used to store atrac 3 data:

29 * RealMedia (.rm), RIFF WAV (.wav, .at3), Sony OpenMG (.oma, .aa3).

30 *

31 * To use this decoder, a calling application must supply the extradata

32 * bytes provided in the containers above.

33 */

35 #include <math.h>

36 #include <stddef.h>

37 #include <stdio.h>

39 #include "libavutil/float_dsp.h"

40 #include "libavutil/libm.h"

41 #include "avcodec.h"

42 #include "bytestream.h"

43 #include "fft.h"

44 #include "fmtconvert.h"

45 #include "get_bits.h"

46 #include "internal.h"

48 #include "atrac.h"

49 #include "atrac3data.h"

51 #define JOINT_STEREO 0x12

52 #define STEREO 0x2

54 #define SAMPLES_PER_FRAME 1024

55 #define MDCT_SIZE 512

57 typedef struct GainInfo {

58 int num_gain_data;

59 int lev_code[8];

60 int loc_code[8];

61 } GainInfo;

63 typedef struct GainBlock {

64 GainInfo g_block[4];

65 } GainBlock;

67 typedef struct TonalComponent {

68 int pos;

69 int num_coefs;

70 float coef[8];

71 } TonalComponent;

73 typedef struct ChannelUnit {

74 int bands_coded;

75 int num_components;

76 float prev_frame[SAMPLES_PER_FRAME];

77 int gc_blk_switch;

78 TonalComponent components[64];

79 GainBlock gain_block[2];

81 DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME];

82 DECLARE_ALIGNED(32, float, imdct_buf)[SAMPLES_PER_FRAME];

84 float delay_buf1[46]; ///<qmf delay buffers

85 float delay_buf2[46];

86 float delay_buf3[46];

87 } ChannelUnit;

89 typedef struct ATRAC3Context {

90 AVFrame frame;

91 GetBitContext gb;

92 //@{

93 /** stream data */

94 int coding_mode;

96 ChannelUnit *units;

97 //@}

98 //@{

99 /** joint-stereo related variables */

100 int matrix_coeff_index_prev[4];

101 int matrix_coeff_index_now[4];

102 int matrix_coeff_index_next[4];

103 int weighting_delay[6];

104 //@}

105 //@{

106 /** data buffers */

107 uint8_t *decoded_bytes_buffer;

108 float temp_buf[1070];

109 //@}

110 //@{

111 /** extradata */

112 int scrambled_stream;

113 //@}

114

115 FFTContext mdct_ctx;

116 FmtConvertContext fmt_conv;

117 AVFloatDSPContext fdsp;

118 } ATRAC3Context;

119

120 static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE];

121 static VLC_TYPE atrac3_vlc_table[4096][2];

122 static VLC spectral_coeff_tab[7];

123 static float gain_tab1[16];

124 static float gain_tab2[31];

125

126

127 /**

128 * Regular 512 points IMDCT without overlapping, with the exception of the

129 * swapping of odd bands caused by the reverse spectra of the QMF.

130 *

131 * @param odd_band 1 if the band is an odd band

132 */

133 static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band)

134 {

135 int i;

136

137 if (odd_band) {

138 /**

139 * Reverse the odd bands before IMDCT, this is an effect of the QMF

140 * transform or it gives better compression to do it this way.

141 * FIXME: It should be possible to handle this in imdct_calc

142 * for that to happen a modification of the prerotation step of

143 * all SIMD code and C code is needed.

144 * Or fix the functions before so they generate a pre reversed spectrum.

145 */

146 for (i = 0; i < 128; i++)

147 FFSWAP(float, input[i], input[255 - i]);

148 }

149

150 q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input);

151

152 /* Perform windowing on the output. */

153 q->fdsp.vector_fmul(output, output, mdct_window, MDCT_SIZE);

154 }

155

156 /*

157 * indata descrambling, only used for data coming from the rm container

158 */

159 static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes)

160 {

161 int i, off;

162 uint32_t c;

163 const uint32_t *buf;

164 uint32_t *output = (uint32_t *)out;

165

166 off = (intptr_t)input & 3;

167 buf = (const uint32_t *)(input - off);

168 if (off)

169 c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8))));

170 else

171 c = av_be2ne32(0x537F6103U);

172 bytes += 3 + off;

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

174 output[i] = c ^ buf[i];

175

176 if (off)

177 av_log_ask_for_sample(NULL, "Offset of %d not handled.\n", off);

178

179 return off;

180 }

181

182 static av_cold void init_atrac3_window(void)

183 {

184 int i, j;

185

186 /* generate the mdct window, for details see

187 * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */

188 for (i = 0, j = 255; i < 128; i++, j--) {

189 float wi = sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;

190 float wj = sin(((j + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;

191 float w = 0.5 * (wi * wi + wj * wj);

192 mdct_window[i] = mdct_window[511 - i] = wi / w;

193 mdct_window[j] = mdct_window[511 - j] = wj / w;

194 }

195 }

196

197 static av_cold int atrac3_decode_close(AVCodecContext *avctx)

198 {

199 ATRAC3Context *q = avctx->priv_data;

200

201 av_free(q->units);

202 av_free(q->decoded_bytes_buffer);

203

204 ff_mdct_end(&q->mdct_ctx);

205

206 return 0;

207 }

208

209 /**

210 * Mantissa decoding

211 *

212 * @param selector which table the output values are coded with

213 * @param coding_flag constant length coding or variable length coding

214 * @param mantissas mantissa output table

215 * @param num_codes number of values to get

216 */

217 static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,

218 int coding_flag, int *mantissas,

219 int num_codes)

220 {

221 int i, code, huff_symb;

222

223 if (selector == 1)

224 num_codes /= 2;

225

226 if (coding_flag != 0) {

227 /* constant length coding (CLC) */

228 int num_bits = clc_length_tab[selector];

229

230 if (selector > 1) {

231 for (i = 0; i < num_codes; i++) {

232 if (num_bits)

233 code = get_sbits(gb, num_bits);

234 else

235 code = 0;

236 mantissas[i] = code;

237 }

238 } else {

239 for (i = 0; i < num_codes; i++) {

240 if (num_bits)

241 code = get_bits(gb, num_bits); // num_bits is always 4 in this case

242 else

243 code = 0;

244 mantissas[i * 2 ] = mantissa_clc_tab[code >> 2];

245 mantissas[i * 2 + 1] = mantissa_clc_tab[code & 3];

246 }

247 }

248 } else {

249 /* variable length coding (VLC) */

250 if (selector != 1) {

251 for (i = 0; i < num_codes; i++) {

252 huff_symb = get_vlc2(gb, spectral_coeff_tab[selector-1].table,

253 spectral_coeff_tab[selector-1].bits, 3);

254 huff_symb += 1;

255 code = huff_symb >> 1;

256 if (huff_symb & 1)

257 code = -code;

258 mantissas[i] = code;

259 }

260 } else {

261 for (i = 0; i < num_codes; i++) {

262 huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table,

263 spectral_coeff_tab[selector - 1].bits, 3);

264 mantissas[i * 2 ] = mantissa_vlc_tab[huff_symb * 2 ];

265 mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1];

266 }

267 }

268 }

269 }

270

271 /**

272 * Restore the quantized band spectrum coefficients

273 *

274 * @return subband count, fix for broken specification/files

275 */

276 static int decode_spectrum(GetBitContext *gb, float *output)

277 {

278 int num_subbands, coding_mode, i, j, first, last, subband_size;

279 int subband_vlc_index[32], sf_index[32];

280 int mantissas[128];

281 float scale_factor;

282

283 num_subbands = get_bits(gb, 5); // number of coded subbands

284 coding_mode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC

285

286 /* get the VLC selector table for the subbands, 0 means not coded */

287 for (i = 0; i <= num_subbands; i++)

288 subband_vlc_index[i] = get_bits(gb, 3);

289

290 /* read the scale factor indexes from the stream */

291 for (i = 0; i <= num_subbands; i++) {

292 if (subband_vlc_index[i] != 0)

293 sf_index[i] = get_bits(gb, 6);

294 }

295

296 for (i = 0; i <= num_subbands; i++) {

297 first = subband_tab[i ];

298 last = subband_tab[i + 1];

299

300 subband_size = last - first;

301

302 if (subband_vlc_index[i] != 0) {

303 /* decode spectral coefficients for this subband */

304 /* TODO: This can be done faster is several blocks share the

305 * same VLC selector (subband_vlc_index) */

306 read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode,

307 mantissas, subband_size);

308

309 /* decode the scale factor for this subband */

310 scale_factor = ff_atrac_sf_table[sf_index[i]] *

311 inv_max_quant[subband_vlc_index[i]];

312

313 /* inverse quantize the coefficients */

314 for (j = 0; first < last; first++, j++)

315 output[first] = mantissas[j] * scale_factor;

316 } else {

317 /* this subband was not coded, so zero the entire subband */

318 memset(output + first, 0, subband_size * sizeof(*output));

319 }

320 }

321

322 /* clear the subbands that were not coded */

323 first = subband_tab[i];

324 memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(*output));

325 return num_subbands;

326 }

327

328 /**

329 * Restore the quantized tonal components

330 *

331 * @param components tonal components

332 * @param num_bands number of coded bands

333 */

334 static int decode_tonal_components(GetBitContext *gb,

335 TonalComponent *components, int num_bands)

336 {

337 int i, b, c, m;

338 int nb_components, coding_mode_selector, coding_mode;

339 int band_flags[4], mantissa[8];

340 int component_count = 0;

341

342 nb_components = get_bits(gb, 5);

343

344 /* no tonal components */

345 if (nb_components == 0)

346 return 0;

347

348 coding_mode_selector = get_bits(gb, 2);

349 if (coding_mode_selector == 2)

350 return AVERROR_INVALIDDATA;

351

352 coding_mode = coding_mode_selector & 1;

353

354 for (i = 0; i < nb_components; i++) {

355 int coded_values_per_component, quant_step_index;

356

357 for (b = 0; b <= num_bands; b++)

358 band_flags[b] = get_bits1(gb);

359

360 coded_values_per_component = get_bits(gb, 3);

361

362 quant_step_index = get_bits(gb, 3);

363 if (quant_step_index <= 1)

364 return AVERROR_INVALIDDATA;

365

366 if (coding_mode_selector == 3)

367 coding_mode = get_bits1(gb);

368

369 for (b = 0; b < (num_bands + 1) * 4; b++) {

370 int coded_components;

371

372 if (band_flags[b >> 2] == 0)

373 continue;

374

375 coded_components = get_bits(gb, 3);

376

377 for (c = 0; c < coded_components; c++) {

378 TonalComponent *cmp = &components[component_count];

379 int sf_index, coded_values, max_coded_values;

380 float scale_factor;

381

382 sf_index = get_bits(gb, 6);

383 if (component_count >= 64)

384 return AVERROR_INVALIDDATA;

385

386 cmp->pos = b * 64 + get_bits(gb, 6);

387

388 max_coded_values = SAMPLES_PER_FRAME - cmp->pos;

389 coded_values = coded_values_per_component + 1;

390 coded_values = FFMIN(max_coded_values, coded_values);

391

392 scale_factor = ff_atrac_sf_table[sf_index] *

393 inv_max_quant[quant_step_index];

394

395 read_quant_spectral_coeffs(gb, quant_step_index, coding_mode,

396 mantissa, coded_values);

397

398 cmp->num_coefs = coded_values;

399

400 /* inverse quant */

401 for (m = 0; m < coded_values; m++)

402 cmp->coef[m] = mantissa[m] * scale_factor;

403

404 component_count++;

405 }

406 }

407 }

408

409 return component_count;

410 }

411

412 /**

413 * Decode gain parameters for the coded bands

414 *

415 * @param block the gainblock for the current band

416 * @param num_bands amount of coded bands

417 */

418 static int decode_gain_control(GetBitContext *gb, GainBlock *block,

419 int num_bands)

420 {

421 int i, cf, num_data;

422 int *level, *loc;

423

424 GainInfo *gain = block->g_block;

425

426 for (i = 0; i <= num_bands; i++) {

427 num_data = get_bits(gb, 3);

428 gain[i].num_gain_data = num_data;

429 level = gain[i].lev_code;

430 loc = gain[i].loc_code;

431

432 for (cf = 0; cf < gain[i].num_gain_data; cf++) {

433 level[cf] = get_bits(gb, 4);

434 loc [cf] = get_bits(gb, 5);

435 if (cf && loc[cf] <= loc[cf - 1])

436 return AVERROR_INVALIDDATA;

437 }

438 }

439

440 /* Clear the unused blocks. */

441 for (; i < 4 ; i++)

442 gain[i].num_gain_data = 0;

443

444 return 0;

445 }

446

447 /**

448 * Apply gain parameters and perform the MDCT overlapping part

449 *

450 * @param input input buffer

451 * @param prev previous buffer to perform overlap against

452 * @param output output buffer

453 * @param gain1 current band gain info

454 * @param gain2 next band gain info

455 */

456 static void gain_compensate_and_overlap(float *input, float *prev,

457 float *output, GainInfo *gain1,

458 GainInfo *gain2)

459 {

460 float g1, g2, gain_inc;

461 int i, j, num_data, start_loc, end_loc;

462

463

464 if (gain2->num_gain_data == 0)

465 g1 = 1.0;

466 else

467 g1 = gain_tab1[gain2->lev_code[0]];

468

469 if (gain1->num_gain_data == 0) {

470 for (i = 0; i < 256; i++)

471 output[i] = input[i] * g1 + prev[i];

472 } else {

473 num_data = gain1->num_gain_data;

474 gain1->loc_code[num_data] = 32;

475 gain1->lev_code[num_data] = 4;

476

477 for (i = 0, j = 0; i < num_data; i++) {

478 start_loc = gain1->loc_code[i] * 8;

479 end_loc = start_loc + 8;

480

481 g2 = gain_tab1[gain1->lev_code[i]];

482 gain_inc = gain_tab2[gain1->lev_code[i + 1] -

483 gain1->lev_code[i ] + 15];

484

485 /* interpolate */

486 for (; j < start_loc; j++)

487 output[j] = (input[j] * g1 + prev[j]) * g2;

488

489 /* interpolation is done over eight samples */

490 for (; j < end_loc; j++) {

491 output[j] = (input[j] * g1 + prev[j]) * g2;

492 g2 *= gain_inc;

493 }

494 }

495

496 for (; j < 256; j++)

497 output[j] = input[j] * g1 + prev[j];

498 }

499

500 /* Delay for the overlapping part. */

501 memcpy(prev, &input[256], 256 * sizeof(*prev));

502 }

503

504 /**

505 * Combine the tonal band spectrum and regular band spectrum

506 *

507 * @param spectrum output spectrum buffer

508 * @param num_components number of tonal components

509 * @param components tonal components for this band

510 * @return position of the last tonal coefficient

511 */

512 static int add_tonal_components(float *spectrum, int num_components,

513 TonalComponent *components)

514 {

515 int i, j, last_pos = -1;

516 float *input, *output;

517

518 for (i = 0; i < num_components; i++) {

519 last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos);

520 input = components[i].coef;

521 output = &spectrum[components[i].pos];

522

523 for (j = 0; j < components[i].num_coefs; j++)

524 output[j] += input[j];

525 }

526

527 return last_pos;

528 }

529

530 #define INTERPOLATE(old, new, nsample) \

531 ((old) + (nsample) * 0.125 * ((new) - (old)))

532

533 static void reverse_matrixing(float *su1, float *su2, int *prev_code,

534 int *curr_code)

535 {

536 int i, nsample, band;

537 float mc1_l, mc1_r, mc2_l, mc2_r;

538

539 for (i = 0, band = 0; band < 4 * 256; band += 256, i++) {

540 int s1 = prev_code[i];

541 int s2 = curr_code[i];

542 nsample = band;

543

544 if (s1 != s2) {

545 /* Selector value changed, interpolation needed. */

546 mc1_l = matrix_coeffs[s1 * 2 ];

547 mc1_r = matrix_coeffs[s1 * 2 + 1];

548 mc2_l = matrix_coeffs[s2 * 2 ];

549 mc2_r = matrix_coeffs[s2 * 2 + 1];

550

551 /* Interpolation is done over the first eight samples. */

552 for (; nsample < band + 8; nsample++) {

553 float c1 = su1[nsample];

554 float c2 = su2[nsample];

555 c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample - band) +

556 c2 * INTERPOLATE(mc1_r, mc2_r, nsample - band);

557 su1[nsample] = c2;

558 su2[nsample] = c1 * 2.0 - c2;

559 }

560 }

561

562 /* Apply the matrix without interpolation. */

563 switch (s2) {

564 case 0: /* M/S decoding */

565 for (; nsample < band + 256; nsample++) {

566 float c1 = su1[nsample];

567 float c2 = su2[nsample];

568 su1[nsample] = c2 * 2.0;

569 su2[nsample] = (c1 - c2) * 2.0;

570 }

571 break;

572 case 1:

573 for (; nsample < band + 256; nsample++) {

574 float c1 = su1[nsample];

575 float c2 = su2[nsample];

576 su1[nsample] = (c1 + c2) * 2.0;

577 su2[nsample] = c2 * -2.0;

578 }

579 break;

580 case 2:

581 case 3:

582 for (; nsample < band + 256; nsample++) {

583 float c1 = su1[nsample];

584 float c2 = su2[nsample];

585 su1[nsample] = c1 + c2;

586 su2[nsample] = c1 - c2;

587 }

588 break;

589 default:

590 av_assert1(0);

591 }

592 }

593 }

594

595 static void get_channel_weights(int index, int flag, float ch[2])

596 {

597 if (index == 7) {

598 ch[0] = 1.0;

599 ch[1] = 1.0;

600 } else {

601 ch[0] = (index & 7) / 7.0;

602 ch[1] = sqrt(2 - ch[0] * ch[0]);

603 if (flag)

604 FFSWAP(float, ch[0], ch[1]);

605 }

606 }

607

608 static void channel_weighting(float *su1, float *su2, int *p3)

609 {

610 int band, nsample;

611 /* w[x][y] y=0 is left y=1 is right */

612 float w[2][2];

613

614 if (p3[1] != 7 || p3[3] != 7) {

615 get_channel_weights(p3[1], p3[0], w[0]);

616 get_channel_weights(p3[3], p3[2], w[1]);

617

618 for (band = 256; band < 4 * 256; band += 256) {

619 for (nsample = band; nsample < band + 8; nsample++) {

620 su1[nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample - band);

621 su2[nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample - band);

622 }

623 for(; nsample < band + 256; nsample++) {

624 su1[nsample] *= w[1][0];

625 su2[nsample] *= w[1][1];

626 }

627 }

628 }

629 }

630

631 /**

632 * Decode a Sound Unit

633 *

634 * @param snd the channel unit to be used

635 * @param output the decoded samples before IQMF in float representation

636 * @param channel_num channel number

637 * @param coding_mode the coding mode (JOINT_STEREO or regular stereo/mono)

638 */

639 static int decode_channel_sound_unit(ATRAC3Context *q, GetBitContext *gb,

640 ChannelUnit *snd, float *output,

641 int channel_num, int coding_mode)

642 {

643 int band, ret, num_subbands, last_tonal, num_bands;

644 GainBlock *gain1 = &snd->gain_block[ snd->gc_blk_switch];

645 GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch];

646

647 if (coding_mode == JOINT_STEREO && channel_num == 1) {

648 if (get_bits(gb, 2) != 3) {

649 av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");

650 return AVERROR_INVALIDDATA;

651 }

652 } else {

653 if (get_bits(gb, 6) != 0x28) {

654 av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");

655 return AVERROR_INVALIDDATA;

656 }

657 }

658

659 /* number of coded QMF bands */

660 snd->bands_coded = get_bits(gb, 2);

661

662 ret = decode_gain_control(gb, gain2, snd->bands_coded);

663 if (ret)

664 return ret;

665

666 snd->num_components = decode_tonal_components(gb, snd->components,

667 snd->bands_coded);

668 if (snd->num_components == -1)

669 return -1;

670

671 num_subbands = decode_spectrum(gb, snd->spectrum);

672

673 /* Merge the decoded spectrum and tonal components. */

674 last_tonal = add_tonal_components(snd->spectrum, snd->num_components,

675 snd->components);

676

677

678 /* calculate number of used MLT/QMF bands according to the amount of coded

679 spectral lines */

680 num_bands = (subband_tab[num_subbands] - 1) >> 8;

681 if (last_tonal >= 0)

682 num_bands = FFMAX((last_tonal + 256) >> 8, num_bands);

683

684

685 /* Reconstruct time domain samples. */

686 for (band = 0; band < 4; band++) {

687 /* Perform the IMDCT step without overlapping. */

688 if (band <= num_bands)

689 imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1);

690 else

691 memset(snd->imdct_buf, 0, 512 * sizeof(*snd->imdct_buf));

692

693 /* gain compensation and overlapping */

694 gain_compensate_and_overlap(snd->imdct_buf,

695 &snd->prev_frame[band * 256],

696 &output[band * 256],

697 &gain1->g_block[band],

698 &gain2->g_block[band]);

699 }

700

701 /* Swap the gain control buffers for the next frame. */

702 snd->gc_blk_switch ^= 1;

703

704 return 0;

705 }

706

707 static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf,

708 float **out_samples)

709 {

710 ATRAC3Context *q = avctx->priv_data;

711 int ret, i;

712 uint8_t *ptr1;

713

714 if (q->coding_mode == JOINT_STEREO) {

715 /* channel coupling mode */

716 /* decode Sound Unit 1 */

717 init_get_bits(&q->gb, databuf, avctx->block_align * 8);

718

719 ret = decode_channel_sound_unit(q, &q->gb, q->units, out_samples[0], 0,

720 JOINT_STEREO);

721 if (ret != 0)

722 return ret;

723

724 /* Framedata of the su2 in the joint-stereo mode is encoded in

725 * reverse byte order so we need to swap it first. */

726 if (databuf == q->decoded_bytes_buffer) {

727 uint8_t *ptr2 = q->decoded_bytes_buffer + avctx->block_align - 1;

728 ptr1 = q->decoded_bytes_buffer;

729 for (i = 0; i < avctx->block_align / 2; i++, ptr1++, ptr2--)

730 FFSWAP(uint8_t, *ptr1, *ptr2);

731 } else {

732 const uint8_t *ptr2 = databuf + avctx->block_align - 1;

733 for (i = 0; i < avctx->block_align; i++)

734 q->decoded_bytes_buffer[i] = *ptr2--;

735 }

736

737 /* Skip the sync codes (0xF8). */

738 ptr1 = q->decoded_bytes_buffer;

739 for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {

740 if (i >= avctx->block_align)

741 return AVERROR_INVALIDDATA;

742 }

743

744

745 /* set the bitstream reader at the start of the second Sound Unit*/

746 init_get_bits(&q->gb, ptr1, avctx->block_align * 8);

747

748 /* Fill the Weighting coeffs delay buffer */

749 memmove(q->weighting_delay, &q->weighting_delay[2],

750 4 * sizeof(*q->weighting_delay));

751 q->weighting_delay[4] = get_bits1(&q->gb);

752 q->weighting_delay[5] = get_bits(&q->gb, 3);

753

754 for (i = 0; i < 4; i++) {

755 q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i];

756 q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i];

757 q->matrix_coeff_index_next[i] = get_bits(&q->gb, 2);

758 }

759

760 /* Decode Sound Unit 2. */

761 ret = decode_channel_sound_unit(q, &q->gb, &q->units[1],

762 out_samples[1], 1, JOINT_STEREO);

763 if (ret != 0)

764 return ret;

765

766 /* Reconstruct the channel coefficients. */

767 reverse_matrixing(out_samples[0], out_samples[1],

768 q->matrix_coeff_index_prev,

769 q->matrix_coeff_index_now);

770

771 channel_weighting(out_samples[0], out_samples[1], q->weighting_delay);

772 } else {

773 /* normal stereo mode or mono */

774 /* Decode the channel sound units. */

775 for (i = 0; i < avctx->channels; i++) {

776 /* Set the bitstream reader at the start of a channel sound unit. */

777 init_get_bits(&q->gb,

778 databuf + i * avctx->block_align / avctx->channels,

779 avctx->block_align * 8 / avctx->channels);

780

781 ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],

782 out_samples[i], i, q->coding_mode);

783 if (ret != 0)

784 return ret;

785 }

786 }

787

788 /* Apply the iQMF synthesis filter. */

789 for (i = 0; i < avctx->channels; i++) {

790 float *p1 = out_samples[i];

791 float *p2 = p1 + 256;

792 float *p3 = p2 + 256;

793 float *p4 = p3 + 256;

794 ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);

795 ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);

796 ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);

797 }

798

799 return 0;

800 }

801

802 static int atrac3_decode_frame(AVCodecContext *avctx, void *data,

803 int *got_frame_ptr, AVPacket *avpkt)

804 {

805 const uint8_t *buf = avpkt->data;

806 int buf_size = avpkt->size;

807 ATRAC3Context *q = avctx->priv_data;

808 int ret;

809 const uint8_t *databuf;

810

811 if (buf_size < avctx->block_align) {

812 av_log(avctx, AV_LOG_ERROR,

813 "Frame too small (%d bytes). Truncated file?\n", buf_size);

814 return AVERROR_INVALIDDATA;

815 }

816

817 /* get output buffer */

818 q->frame.nb_samples = SAMPLES_PER_FRAME;

819 if ((ret = ff_get_buffer(avctx, &q->frame)) < 0) {

820 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");

821 return ret;

822 }

823

824 /* Check if we need to descramble and what buffer to pass on. */

825 if (q->scrambled_stream) {

826 decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);

827 databuf = q->decoded_bytes_buffer;

828 } else {

829 databuf = buf;

830 }

831

832 ret = decode_frame(avctx, databuf, (float **)q->frame.extended_data);

833 if (ret) {

834 av_log(NULL, AV_LOG_ERROR, "Frame decoding error!\n");

835 return ret;

836 }

837

838 *got_frame_ptr = 1;

839 *(AVFrame *)data = q->frame;

840

841 return avctx->block_align;

842 }

843

844 static void atrac3_init_static_data(void)

845 {

846 int i;

847

848 init_atrac3_window();

849 ff_atrac_generate_tables();

850

851 /* Initialize the VLC tables. */

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

853 spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]];

854 spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] -

855 atrac3_vlc_offs[i ];

856 init_vlc(&spectral_coeff_tab[i], 9, huff_tab_sizes[i],

857 huff_bits[i], 1, 1,

858 huff_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);

859 }

860

861 /* Generate gain tables */

862 for (i = 0; i < 16; i++)

863 gain_tab1[i] = exp2f (4 - i);

864

865 for (i = -15; i < 16; i++)

866 gain_tab2[i + 15] = exp2f (i * -0.125);

867 }

868

869 static av_cold int atrac3_decode_init(AVCodecContext *avctx)

870 {

871 static int static_init_done;

872 int i, ret;

873 int version, delay, samples_per_frame, frame_factor;

874 const uint8_t *edata_ptr = avctx->extradata;

875 ATRAC3Context *q = avctx->priv_data;

876

877 if (avctx->channels <= 0 || avctx->channels > 2) {

878 av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n");

879 return AVERROR(EINVAL);

880 }

881

882 if (!static_init_done)

883 atrac3_init_static_data();

884 static_init_done = 1;

885

886 /* Take care of the codec-specific extradata. */

887 if (avctx->extradata_size == 14) {

888 /* Parse the extradata, WAV format */

889 av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n",

890 bytestream_get_le16(&edata_ptr)); // Unknown value always 1

891 edata_ptr += 4; // samples per channel

892 q->coding_mode = bytestream_get_le16(&edata_ptr);

893 av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n",

894 bytestream_get_le16(&edata_ptr)); //Dupe of coding mode

895 frame_factor = bytestream_get_le16(&edata_ptr); // Unknown always 1

896 av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n",

897 bytestream_get_le16(&edata_ptr)); // Unknown always 0

898

899 /* setup */

900 samples_per_frame = SAMPLES_PER_FRAME * avctx->channels;

901 version = 4;

902 delay = 0x88E;

903 q->coding_mode = q->coding_mode ? JOINT_STEREO : STEREO;

904 q->scrambled_stream = 0;

905

906 if (avctx->block_align != 96 * avctx->channels * frame_factor &&

907 avctx->block_align != 152 * avctx->channels * frame_factor &&

908 avctx->block_align != 192 * avctx->channels * frame_factor) {

909 av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor "

910 "configuration %d/%d/%d\n", avctx->block_align,

911 avctx->channels, frame_factor);

912 return AVERROR_INVALIDDATA;

913 }

914 } else if (avctx->extradata_size == 10) {

915 /* Parse the extradata, RM format. */

916 version = bytestream_get_be32(&edata_ptr);

917 samples_per_frame = bytestream_get_be16(&edata_ptr);

918 delay = bytestream_get_be16(&edata_ptr);

919 q->coding_mode = bytestream_get_be16(&edata_ptr);

920 q->scrambled_stream = 1;

921

922 } else {

923 av_log(NULL, AV_LOG_ERROR, "Unknown extradata size %d.\n",

924 avctx->extradata_size);

925 return AVERROR(EINVAL);

926 }

927

928 if (q->coding_mode == JOINT_STEREO && avctx->channels < 2) {

929 av_log(avctx, AV_LOG_ERROR, "Invalid coding mode\n");

930 return AVERROR_INVALIDDATA;

931 }

932

933 /* Check the extradata */

934

935 if (version != 4) {

936 av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version);

937 return AVERROR_INVALIDDATA;

938 }

939

940 if (samples_per_frame != SAMPLES_PER_FRAME &&

941 samples_per_frame != SAMPLES_PER_FRAME * 2) {

942 av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n",

943 samples_per_frame);

944 return AVERROR_INVALIDDATA;

945 }

946

947 if (delay != 0x88E) {

948 av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n",

949 delay);

950 return AVERROR_INVALIDDATA;

951 }

952

953 if (q->coding_mode == STEREO)

954 av_log(avctx, AV_LOG_DEBUG, "Normal stereo detected.\n");

955 else if (q->coding_mode == JOINT_STEREO)

956 av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n");

957 else {

958 av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n",

959 q->coding_mode);

960 return AVERROR_INVALIDDATA;

961 }

962

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

964 return AVERROR(EINVAL);

965

966 q->decoded_bytes_buffer = av_mallocz(FFALIGN(avctx->block_align, 4) +

967 FF_INPUT_BUFFER_PADDING_SIZE);

968 if (q->decoded_bytes_buffer == NULL)

969 return AVERROR(ENOMEM);

970

971 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

972

973 /* initialize the MDCT transform */

974 if ((ret = ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768)) < 0) {

975 av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");

976 av_freep(&q->decoded_bytes_buffer);

977 return ret;

978 }

979

980 /* init the joint-stereo decoding data */

981 q->weighting_delay[0] = 0;

982 q->weighting_delay[1] = 7;

983 q->weighting_delay[2] = 0;

984 q->weighting_delay[3] = 7;

985 q->weighting_delay[4] = 0;

986 q->weighting_delay[5] = 7;

987

988 for (i = 0; i < 4; i++) {

989 q->matrix_coeff_index_prev[i] = 3;

990 q->matrix_coeff_index_now[i] = 3;

991 q->matrix_coeff_index_next[i] = 3;

992 }

993

994 avpriv_float_dsp_init(&q->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);

995 ff_fmt_convert_init(&q->fmt_conv, avctx);

996

997 q->units = av_mallocz(sizeof(*q->units) * avctx->channels);

998 if (!q->units) {

999 atrac3_decode_close(avctx);

1000 return AVERROR(ENOMEM);

1001 }

1002

1003 avcodec_get_frame_defaults(&q->frame);

1004 avctx->coded_frame = &q->frame;

1005

1006 return 0;

1007 }

1008

1009 AVCodec ff_atrac3_decoder = {

1010 .name = "atrac3",

1011 .type = AVMEDIA_TYPE_AUDIO,

1012 .id = AV_CODEC_ID_ATRAC3,

1013 .priv_data_size = sizeof(ATRAC3Context),

1014 .init = atrac3_decode_init,

1015 .close = atrac3_decode_close,

1016 .decode = atrac3_decode_frame,

1017 .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,

1018 .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),

1019 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,

1020 AV_SAMPLE_FMT_NONE },

1021 };

Generated on Sat May 25 2013 03:58:31 for FFmpeg by doxygen 1.8.2