FFmpeg: libavcodec/atrac3.c Source File

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atrac3.c

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1 /*

2 * ATRAC3 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 * ATRAC3 compatible decoder.

26 * This decoder handles Sony's ATRAC3 data.

27 *

28 * Container formats used to store ATRAC3 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/attributes.h"

40 #include "libavutil/float_dsp.h"

41 #include "libavutil/libm.h"

42 #include "avcodec.h"

43 #include "bytestream.h"

44 #include "fft.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 GainBlock {

58 AtracGainInfo g_block[4];

59 } GainBlock;

61 typedef struct TonalComponent {

62 int pos;

63 int num_coefs;

64 float coef[8];

65 } TonalComponent;

67 typedef struct ChannelUnit {

68 int bands_coded;

69 int num_components;

70 float prev_frame[SAMPLES_PER_FRAME];

71 int gc_blk_switch;

72 TonalComponent components[64];

73 GainBlock gain_block[2];

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

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

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

79 float delay_buf2[46];

80 float delay_buf3[46];

81 } ChannelUnit;

83 typedef struct ATRAC3Context {

84 GetBitContext gb;

85 //@{

86 /** stream data */

87 int coding_mode;

89 ChannelUnit *units;

90 //@}

91 //@{

92 /** joint-stereo related variables */

93 int matrix_coeff_index_prev[4];

94 int matrix_coeff_index_now[4];

95 int matrix_coeff_index_next[4];

96 int weighting_delay[6];

97 //@}

98 //@{

99 /** data buffers */

100 uint8_t *decoded_bytes_buffer;

101 float temp_buf[1070];

102 //@}

103 //@{

104 /** extradata */

105 int scrambled_stream;

106 //@}

107

108 AtracGCContext gainc_ctx;

109 FFTContext mdct_ctx;

110 AVFloatDSPContext *fdsp;

111 } ATRAC3Context;

112

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

114 static VLC_TYPE atrac3_vlc_table[4096][2];

115 static VLC spectral_coeff_tab[7];

116

117 /**

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

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

120 *

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

122 */

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

124 {

125 int i;

126

127 if (odd_band) {

128 /**

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

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

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

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

133 * all SIMD code and C code is needed.

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

135 */

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

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

138 }

139

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

141

142 /* Perform windowing on the output. */

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

144 }

145

146 /*

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

148 */

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

150 {

151 int i, off;

152 uint32_t c;

153 const uint32_t *buf;

154 uint32_t *output = (uint32_t *)out;

155

156 off = (intptr_t)input & 3;

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

158 if (off)

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

160 else

161 c = av_be2ne32(0x537F6103U);

162 bytes += 3 + off;

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

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

165

166 if (off)

167 avpriv_request_sample(NULL, "Offset of %d", off);

168

169 return off;

170 }

171

172 static av_cold void init_imdct_window(void)

173 {

174 int i, j;

175

176 /* generate the mdct window, for details see

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

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

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

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

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

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

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

184 }

185 }

186

187 static av_cold int atrac3_decode_close(AVCodecContext *avctx)

188 {

189 ATRAC3Context *q = avctx->priv_data;

190

191 av_freep(&q->units);

192 av_freep(&q->decoded_bytes_buffer);

193 av_freep(&q->fdsp);

194

195 ff_mdct_end(&q->mdct_ctx);

196

197 return 0;

198 }

199

200 /**

201 * Mantissa decoding

202 *

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

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

205 * @param mantissas mantissa output table

206 * @param num_codes number of values to get

207 */

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

209 int coding_flag, int *mantissas,

210 int num_codes)

211 {

212 int i, code, huff_symb;

213

214 if (selector == 1)

215 num_codes /= 2;

216

217 if (coding_flag != 0) {

218 /* constant length coding (CLC) */

219 int num_bits = clc_length_tab[selector];

220

221 if (selector > 1) {

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

223 if (num_bits)

224 code = get_sbits(gb, num_bits);

225 else

226 code = 0;

227 mantissas[i] = code;

228 }

229 } else {

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

231 if (num_bits)

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

233 else

234 code = 0;

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

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

237 }

238 }

239 } else {

240 /* variable length coding (VLC) */

241 if (selector != 1) {

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

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

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

245 huff_symb += 1;

246 code = huff_symb >> 1;

247 if (huff_symb & 1)

248 code = -code;

249 mantissas[i] = code;

250 }

251 } else {

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

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

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

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

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

257 }

258 }

259 }

260 }

261

262 /**

263 * Restore the quantized band spectrum coefficients

264 *

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

266 */

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

268 {

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

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

271 int mantissas[128];

272 float scale_factor;

273

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

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

276

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

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

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

280

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

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

283 if (subband_vlc_index[i] != 0)

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

285 }

286

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

288 first = subband_tab[i ];

289 last = subband_tab[i + 1];

290

291 subband_size = last - first;

292

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

294 /* decode spectral coefficients for this subband */

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

296 * same VLC selector (subband_vlc_index) */

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

298 mantissas, subband_size);

299

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

301 scale_factor = ff_atrac_sf_table[sf_index[i]] *

302 inv_max_quant[subband_vlc_index[i]];

303

304 /* inverse quantize the coefficients */

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

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

307 } else {

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

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

310 }

311 }

312

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

314 first = subband_tab[i];

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

316 return num_subbands;

317 }

318

319 /**

320 * Restore the quantized tonal components

321 *

322 * @param components tonal components

323 * @param num_bands number of coded bands

324 */

325 static int decode_tonal_components(GetBitContext *gb,

326 TonalComponent *components, int num_bands)

327 {

328 int i, b, c, m;

329 int nb_components, coding_mode_selector, coding_mode;

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

331 int component_count = 0;

332

333 nb_components = get_bits(gb, 5);

334

335 /* no tonal components */

336 if (nb_components == 0)

337 return 0;

338

339 coding_mode_selector = get_bits(gb, 2);

340 if (coding_mode_selector == 2)

341 return AVERROR_INVALIDDATA;

342

343 coding_mode = coding_mode_selector & 1;

344

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

346 int coded_values_per_component, quant_step_index;

347

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

349 band_flags[b] = get_bits1(gb);

350

351 coded_values_per_component = get_bits(gb, 3);

352

353 quant_step_index = get_bits(gb, 3);

354 if (quant_step_index <= 1)

355 return AVERROR_INVALIDDATA;

356

357 if (coding_mode_selector == 3)

358 coding_mode = get_bits1(gb);

359

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

361 int coded_components;

362

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

364 continue;

365

366 coded_components = get_bits(gb, 3);

367

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

369 TonalComponent *cmp = &components[component_count];

370 int sf_index, coded_values, max_coded_values;

371 float scale_factor;

372

373 sf_index = get_bits(gb, 6);

374 if (component_count >= 64)

375 return AVERROR_INVALIDDATA;

376

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

378

379 max_coded_values = SAMPLES_PER_FRAME - cmp->pos;

380 coded_values = coded_values_per_component + 1;

381 coded_values = FFMIN(max_coded_values, coded_values);

382

383 scale_factor = ff_atrac_sf_table[sf_index] *

384 inv_max_quant[quant_step_index];

385

386 read_quant_spectral_coeffs(gb, quant_step_index, coding_mode,

387 mantissa, coded_values);

388

389 cmp->num_coefs = coded_values;

390

391 /* inverse quant */

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

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

394

395 component_count++;

396 }

397 }

398 }

399

400 return component_count;

401 }

402

403 /**

404 * Decode gain parameters for the coded bands

405 *

406 * @param block the gainblock for the current band

407 * @param num_bands amount of coded bands

408 */

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

410 int num_bands)

411 {

412 int b, j;

413 int *level, *loc;

414

415 AtracGainInfo *gain = block->g_block;

416

417 for (b = 0; b <= num_bands; b++) {

418 gain[b].num_points = get_bits(gb, 3);

419 level = gain[b].lev_code;

420 loc = gain[b].loc_code;

421

422 for (j = 0; j < gain[b].num_points; j++) {

423 level[j] = get_bits(gb, 4);

424 loc[j] = get_bits(gb, 5);

425 if (j && loc[j] <= loc[j - 1])

426 return AVERROR_INVALIDDATA;

427 }

428 }

429

430 /* Clear the unused blocks. */

431 for (; b < 4 ; b++)

432 gain[b].num_points = 0;

433

434 return 0;

435 }

436

437 /**

438 * Combine the tonal band spectrum and regular band spectrum

439 *

440 * @param spectrum output spectrum buffer

441 * @param num_components number of tonal components

442 * @param components tonal components for this band

443 * @return position of the last tonal coefficient

444 */

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

446 TonalComponent *components)

447 {

448 int i, j, last_pos = -1;

449 float *input, *output;

450

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

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

453 input = components[i].coef;

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

455

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

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

458 }

459

460 return last_pos;

461 }

462

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

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

465

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

467 int *curr_code)

468 {

469 int i, nsample, band;

470 float mc1_l, mc1_r, mc2_l, mc2_r;

471

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

473 int s1 = prev_code[i];

474 int s2 = curr_code[i];

475 nsample = band;

476

477 if (s1 != s2) {

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

479 mc1_l = matrix_coeffs[s1 * 2 ];

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

481 mc2_l = matrix_coeffs[s2 * 2 ];

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

483

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

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

486 float c1 = su1[nsample];

487 float c2 = su2[nsample];

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

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

490 su1[nsample] = c2;

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

492 }

493 }

494

495 /* Apply the matrix without interpolation. */

496 switch (s2) {

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

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

499 float c1 = su1[nsample];

500 float c2 = su2[nsample];

501 su1[nsample] = c2 * 2.0;

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

503 }

504 break;

505 case 1:

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

507 float c1 = su1[nsample];

508 float c2 = su2[nsample];

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

510 su2[nsample] = c2 * -2.0;

511 }

512 break;

513 case 2:

514 case 3:

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

516 float c1 = su1[nsample];

517 float c2 = su2[nsample];

518 su1[nsample] = c1 + c2;

519 su2[nsample] = c1 - c2;

520 }

521 break;

522 default:

523 av_assert1(0);

524 }

525 }

526 }

527

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

529 {

530 if (index == 7) {

531 ch[0] = 1.0;

532 ch[1] = 1.0;

533 } else {

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

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

536 if (flag)

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

538 }

539 }

540

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

542 {

543 int band, nsample;

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

545 float w[2][2];

546

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

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

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

550

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

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

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

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

555 }

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

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

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

559 }

560 }

561 }

562 }

563

564 /**

565 * Decode a Sound Unit

566 *

567 * @param snd the channel unit to be used

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

569 * @param channel_num channel number

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

571 */

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

573 ChannelUnit *snd, float *output,

574 int channel_num, int coding_mode)

575 {

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

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

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

579

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

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

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

583 return AVERROR_INVALIDDATA;

584 }

585 } else {

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

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

588 return AVERROR_INVALIDDATA;

589 }

590 }

591

592 /* number of coded QMF bands */

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

594

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

596 if (ret)

597 return ret;

598

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

600 snd->bands_coded);

601 if (snd->num_components < 0)

602 return snd->num_components;

603

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

605

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

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

608 snd->components);

609

610

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

612 spectral lines */

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

614 if (last_tonal >= 0)

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

616

617

618 /* Reconstruct time domain samples. */

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

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

621 if (band <= num_bands)

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

623 else

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

625

626 /* gain compensation and overlapping */

627 ff_atrac_gain_compensation(&q->gainc_ctx, snd->imdct_buf,

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

629 &gain1->g_block[band], &gain2->g_block[band],

630 256, &output[band * 256]);

631 }

632

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

634 snd->gc_blk_switch ^= 1;

635

636 return 0;

637 }

638

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

640 float **out_samples)

641 {

642 ATRAC3Context *q = avctx->priv_data;

643 int ret, i;

644 uint8_t *ptr1;

645

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

647 /* channel coupling mode */

648 /* decode Sound Unit 1 */

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

650

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

652 JOINT_STEREO);

653 if (ret != 0)

654 return ret;

655

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

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

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

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

660 ptr1 = q->decoded_bytes_buffer;

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

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

663 } else {

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

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

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

667 }

668

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

670 ptr1 = q->decoded_bytes_buffer;

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

672 if (i >= avctx->block_align)

673 return AVERROR_INVALIDDATA;

674 }

675

676

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

678 init_get_bits8(&q->gb, ptr1, q->decoded_bytes_buffer + avctx->block_align - ptr1);

679

680 /* Fill the Weighting coeffs delay buffer */

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

682 4 * sizeof(*q->weighting_delay));

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

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

685

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

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

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

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

690 }

691

692 /* Decode Sound Unit 2. */

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

694 out_samples[1], 1, JOINT_STEREO);

695 if (ret != 0)

696 return ret;

697

698 /* Reconstruct the channel coefficients. */

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

700 q->matrix_coeff_index_prev,

701 q->matrix_coeff_index_now);

702

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

704 } else {

705 /* normal stereo mode or mono */

706 /* Decode the channel sound units. */

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

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

709 init_get_bits(&q->gb,

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

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

712

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

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

715 if (ret != 0)

716 return ret;

717 }

718 }

719

720 /* Apply the iQMF synthesis filter. */

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

722 float *p1 = out_samples[i];

723 float *p2 = p1 + 256;

724 float *p3 = p2 + 256;

725 float *p4 = p3 + 256;

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

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

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

729 }

730

731 return 0;

732 }

733

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

735 int *got_frame_ptr, AVPacket *avpkt)

736 {

737 AVFrame *frame = data;

738 const uint8_t *buf = avpkt->data;

739 int buf_size = avpkt->size;

740 ATRAC3Context *q = avctx->priv_data;

741 int ret;

742 const uint8_t *databuf;

743

744 if (buf_size < avctx->block_align) {

745 av_log(avctx, AV_LOG_ERROR,

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

747 return AVERROR_INVALIDDATA;

748 }

749

750 /* get output buffer */

751 frame->nb_samples = SAMPLES_PER_FRAME;

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

753 return ret;

754

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

756 if (q->scrambled_stream) {

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

758 databuf = q->decoded_bytes_buffer;

759 } else {

760 databuf = buf;

761 }

762

763 ret = decode_frame(avctx, databuf, (float **)frame->extended_data);

764 if (ret) {

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

766 return ret;

767 }

768

769 *got_frame_ptr = 1;

770

771 return avctx->block_align;

772 }

773

774 static av_cold void atrac3_init_static_data(void)

775 {

776 int i;

777

778 init_imdct_window();

779 ff_atrac_generate_tables();

780

781 /* Initialize the VLC tables. */

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

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

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

785 atrac3_vlc_offs[i ];

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

787 huff_bits[i], 1, 1,

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

789 }

790 }

791

792 static av_cold int atrac3_decode_init(AVCodecContext *avctx)

793 {

794 static int static_init_done;

795 int i, ret;

796 int version, delay, samples_per_frame, frame_factor;

797 const uint8_t *edata_ptr = avctx->extradata;

798 ATRAC3Context *q = avctx->priv_data;

799

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

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

802 return AVERROR(EINVAL);

803 }

804

805 if (!static_init_done)

806 atrac3_init_static_data();

807 static_init_done = 1;

808

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

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

811 /* Parse the extradata, WAV format */

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

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

814 edata_ptr += 4; // samples per channel

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

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

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

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

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

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

821

822 /* setup */

823 samples_per_frame = SAMPLES_PER_FRAME * avctx->channels;

824 version = 4;

825 delay = 0x88E;

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

827 q->scrambled_stream = 0;

828

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

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

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

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

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

834 avctx->channels, frame_factor);

835 return AVERROR_INVALIDDATA;

836 }

837 } else if (avctx->extradata_size == 12 || avctx->extradata_size == 10) {

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

839 version = bytestream_get_be32(&edata_ptr);

840 samples_per_frame = bytestream_get_be16(&edata_ptr);

841 delay = bytestream_get_be16(&edata_ptr);

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

843 q->scrambled_stream = 1;

844

845 } else {

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

847 avctx->extradata_size);

848 return AVERROR(EINVAL);

849 }

850

851 /* Check the extradata */

852

853 if (version != 4) {

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

855 return AVERROR_INVALIDDATA;

856 }

857

858 if (samples_per_frame != SAMPLES_PER_FRAME &&

859 samples_per_frame != SAMPLES_PER_FRAME * 2) {

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

861 samples_per_frame);

862 return AVERROR_INVALIDDATA;

863 }

864

865 if (delay != 0x88E) {

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

867 delay);

868 return AVERROR_INVALIDDATA;

869 }

870

871 if (q->coding_mode == STEREO)

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

873 else if (q->coding_mode == JOINT_STEREO) {

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

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

876 return AVERROR_INVALIDDATA;

877 }

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

879 } else {

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

881 q->coding_mode);

882 return AVERROR_INVALIDDATA;

883 }

884

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

886 return AVERROR(EINVAL);

887

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

889 FF_INPUT_BUFFER_PADDING_SIZE);

890 if (!q->decoded_bytes_buffer)

891 return AVERROR(ENOMEM);

892

893 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

894

895 /* initialize the MDCT transform */

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

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

898 av_freep(&q->decoded_bytes_buffer);

899 return ret;

900 }

901

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

903 q->weighting_delay[0] = 0;

904 q->weighting_delay[1] = 7;

905 q->weighting_delay[2] = 0;

906 q->weighting_delay[3] = 7;

907 q->weighting_delay[4] = 0;

908 q->weighting_delay[5] = 7;

909

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

911 q->matrix_coeff_index_prev[i] = 3;

912 q->matrix_coeff_index_now[i] = 3;

913 q->matrix_coeff_index_next[i] = 3;

914 }

915

916 ff_atrac_init_gain_compensation(&q->gainc_ctx, 4, 3);

917 q->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);

918

919 q->units = av_mallocz_array(avctx->channels, sizeof(*q->units));

920 if (!q->units || !q->fdsp) {

921 atrac3_decode_close(avctx);

922 return AVERROR(ENOMEM);

923 }

924

925 return 0;

926 }

927

928 AVCodec ff_atrac3_decoder = {

929 .name = "atrac3",

930 .long_name = NULL_IF_CONFIG_SMALL("ATRAC3 (Adaptive TRansform Acoustic Coding 3)"),

931 .type = AVMEDIA_TYPE_AUDIO,

932 .id = AV_CODEC_ID_ATRAC3,

933 .priv_data_size = sizeof(ATRAC3Context),

934 .init = atrac3_decode_init,

935 .close = atrac3_decode_close,

936 .decode = atrac3_decode_frame,

937 .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,

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

939 AV_SAMPLE_FMT_NONE },

940 };

Generated on Sun Mar 8 2015 02:34:48 for FFmpeg by doxygen 1.8.2