FFmpeg: libavcodec/mpegaudioenc_template.c Source File

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mpegaudioenc_template.c

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1 /*

2 * The simplest mpeg audio layer 2 encoder

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 */

22 /**

23 * @file

24 * The simplest mpeg audio layer 2 encoder.

25 */

27 #include "libavutil/channel_layout.h"

29 #include "avcodec.h"

30 #include "internal.h"

31 #include "put_bits.h"

33 #define FRAC_BITS 15 /* fractional bits for sb_samples and dct */

34 #define WFRAC_BITS 14 /* fractional bits for window */

36 #include "mpegaudio.h"

37 #include "mpegaudiodsp.h"

38 #include "mpegaudiodata.h"

39 #include "mpegaudiotab.h"

41 /* currently, cannot change these constants (need to modify

42 quantization stage) */

43 #define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)

45 #define SAMPLES_BUF_SIZE 4096

47 typedef struct MpegAudioContext {

48 PutBitContext pb;

49 int nb_channels;

50 int lsf; /* 1 if mpeg2 low bitrate selected */

51 int bitrate_index; /* bit rate */

52 int freq_index;

53 int frame_size; /* frame size, in bits, without padding */

54 /* padding computation */

55 int frame_frac, frame_frac_incr, do_padding;

56 short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */

57 int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */

58 int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];

59 unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */

60 /* code to group 3 scale factors */

61 unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];

62 int sblimit; /* number of used subbands */

63 const unsigned char *alloc_table;

64 int16_t filter_bank[512];

65 int scale_factor_table[64];

66 unsigned char scale_diff_table[128];

67 #if USE_FLOATS

68 float scale_factor_inv_table[64];

69 #else

70 int8_t scale_factor_shift[64];

71 unsigned short scale_factor_mult[64];

72 #endif

73 unsigned short total_quant_bits[17]; /* total number of bits per allocation group */

74 } MpegAudioContext;

76 static av_cold int MPA_encode_init(AVCodecContext *avctx)

77 {

78 MpegAudioContext *s = avctx->priv_data;

79 int freq = avctx->sample_rate;

80 int bitrate = avctx->bit_rate;

81 int channels = avctx->channels;

82 int i, v, table;

83 float a;

85 if (channels <= 0 || channels > 2){

86 av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);

87 return AVERROR(EINVAL);

88 }

89 bitrate = bitrate / 1000;

90 s->nb_channels = channels;

91 avctx->frame_size = MPA_FRAME_SIZE;

92 avctx->initial_padding = 512 - 32 + 1;

94 /* encoding freq */

95 s->lsf = 0;

96 for(i=0;i<3;i++) {

97 if (avpriv_mpa_freq_tab[i] == freq)

98 break;

99 if ((avpriv_mpa_freq_tab[i] / 2) == freq) {

100 s->lsf = 1;

101 break;

102 }

103 }

104 if (i == 3){

105 av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);

106 return AVERROR(EINVAL);

107 }

108 s->freq_index = i;

109

110 /* encoding bitrate & frequency */

111 for(i=1;i<15;i++) {

112 if (avpriv_mpa_bitrate_tab[s->lsf][1][i] == bitrate)

113 break;

114 }

115 if (i == 15 && !avctx->bit_rate) {

116 i = 14;

117 bitrate = avpriv_mpa_bitrate_tab[s->lsf][1][i];

118 avctx->bit_rate = bitrate * 1000;

119 }

120 if (i == 15){

121 av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);

122 return AVERROR(EINVAL);

123 }

124 s->bitrate_index = i;

125

126 /* compute total header size & pad bit */

127

128 a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);

129 s->frame_size = ((int)a) * 8;

130

131 /* frame fractional size to compute padding */

132 s->frame_frac = 0;

133 s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);

134

135 /* select the right allocation table */

136 table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);

137

138 /* number of used subbands */

139 s->sblimit = ff_mpa_sblimit_table[table];

140 s->alloc_table = ff_mpa_alloc_tables[table];

141

142 av_dlog(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",

143 bitrate, freq, s->frame_size, table, s->frame_frac_incr);

144

145 for(i=0;i<s->nb_channels;i++)

146 s->samples_offset[i] = 0;

147

148 for(i=0;i<257;i++) {

149 int v;

150 v = ff_mpa_enwindow[i];

151 #if WFRAC_BITS != 16

152 v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);

153 #endif

154 s->filter_bank[i] = v;

155 if ((i & 63) != 0)

156 v = -v;

157 if (i != 0)

158 s->filter_bank[512 - i] = v;

159 }

160

161 for(i=0;i<64;i++) {

162 v = (int)(exp2((3 - i) / 3.0) * (1 << 20));

163 if (v <= 0)

164 v = 1;

165 s->scale_factor_table[i] = v;

166 #if USE_FLOATS

167 s->scale_factor_inv_table[i] = exp2(-(3 - i) / 3.0) / (float)(1 << 20);

168 #else

169 #define P 15

170 s->scale_factor_shift[i] = 21 - P - (i / 3);

171 s->scale_factor_mult[i] = (1 << P) * exp2((i % 3) / 3.0);

172 #endif

173 }

174 for(i=0;i<128;i++) {

175 v = i - 64;

176 if (v <= -3)

177 v = 0;

178 else if (v < 0)

179 v = 1;

180 else if (v == 0)

181 v = 2;

182 else if (v < 3)

183 v = 3;

184 else

185 v = 4;

186 s->scale_diff_table[i] = v;

187 }

188

189 for(i=0;i<17;i++) {

190 v = ff_mpa_quant_bits[i];

191 if (v < 0)

192 v = -v;

193 else

194 v = v * 3;

195 s->total_quant_bits[i] = 12 * v;

196 }

197

198 return 0;

199 }

200

201 /* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */

202 static void idct32(int *out, int *tab)

203 {

204 int i, j;

205 int *t, *t1, xr;

206 const int *xp = costab32;

207

208 for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];

209

210 t = tab + 30;

211 t1 = tab + 2;

212 do {

213 t[0] += t[-4];

214 t[1] += t[1 - 4];

215 t -= 4;

216 } while (t != t1);

217

218 t = tab + 28;

219 t1 = tab + 4;

220 do {

221 t[0] += t[-8];

222 t[1] += t[1-8];

223 t[2] += t[2-8];

224 t[3] += t[3-8];

225 t -= 8;

226 } while (t != t1);

227

228 t = tab;

229 t1 = tab + 32;

230 do {

231 t[ 3] = -t[ 3];

232 t[ 6] = -t[ 6];

233

234 t[11] = -t[11];

235 t[12] = -t[12];

236 t[13] = -t[13];

237 t[15] = -t[15];

238 t += 16;

239 } while (t != t1);

240

241

242 t = tab;

243 t1 = tab + 8;

244 do {

245 int x1, x2, x3, x4;

246

247 x3 = MUL(t[16], FIX(SQRT2*0.5));

248 x4 = t[0] - x3;

249 x3 = t[0] + x3;

250

251 x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));

252 x1 = MUL((t[8] - x2), xp[0]);

253 x2 = MUL((t[8] + x2), xp[1]);

254

255 t[ 0] = x3 + x1;

256 t[ 8] = x4 - x2;

257 t[16] = x4 + x2;

258 t[24] = x3 - x1;

259 t++;

260 } while (t != t1);

261

262 xp += 2;

263 t = tab;

264 t1 = tab + 4;

265 do {

266 xr = MUL(t[28],xp[0]);

267 t[28] = (t[0] - xr);

268 t[0] = (t[0] + xr);

269

270 xr = MUL(t[4],xp[1]);

271 t[ 4] = (t[24] - xr);

272 t[24] = (t[24] + xr);

273

274 xr = MUL(t[20],xp[2]);

275 t[20] = (t[8] - xr);

276 t[ 8] = (t[8] + xr);

277

278 xr = MUL(t[12],xp[3]);

279 t[12] = (t[16] - xr);

280 t[16] = (t[16] + xr);

281 t++;

282 } while (t != t1);

283 xp += 4;

284

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

286 xr = MUL(tab[30-i*4],xp[0]);

287 tab[30-i*4] = (tab[i*4] - xr);

288 tab[ i*4] = (tab[i*4] + xr);

289

290 xr = MUL(tab[ 2+i*4],xp[1]);

291 tab[ 2+i*4] = (tab[28-i*4] - xr);

292 tab[28-i*4] = (tab[28-i*4] + xr);

293

294 xr = MUL(tab[31-i*4],xp[0]);

295 tab[31-i*4] = (tab[1+i*4] - xr);

296 tab[ 1+i*4] = (tab[1+i*4] + xr);

297

298 xr = MUL(tab[ 3+i*4],xp[1]);

299 tab[ 3+i*4] = (tab[29-i*4] - xr);

300 tab[29-i*4] = (tab[29-i*4] + xr);

301

302 xp += 2;

303 }

304

305 t = tab + 30;

306 t1 = tab + 1;

307 do {

308 xr = MUL(t1[0], *xp);

309 t1[0] = (t[0] - xr);

310 t[0] = (t[0] + xr);

311 t -= 2;

312 t1 += 2;

313 xp++;

314 } while (t >= tab);

315

316 for(i=0;i<32;i++) {

317 out[i] = tab[bitinv32[i]];

318 }

319 }

320

321 #define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)

322

323 static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)

324 {

325 short *p, *q;

326 int sum, offset, i, j;

327 int tmp[64];

328 int tmp1[32];

329 int *out;

330

331 offset = s->samples_offset[ch];

332 out = &s->sb_samples[ch][0][0][0];

333 for(j=0;j<36;j++) {

334 /* 32 samples at once */

335 for(i=0;i<32;i++) {

336 s->samples_buf[ch][offset + (31 - i)] = samples[0];

337 samples += incr;

338 }

339

340 /* filter */

341 p = s->samples_buf[ch] + offset;

342 q = s->filter_bank;

343 /* maxsum = 23169 */

344 for(i=0;i<64;i++) {

345 sum = p[0*64] * q[0*64];

346 sum += p[1*64] * q[1*64];

347 sum += p[2*64] * q[2*64];

348 sum += p[3*64] * q[3*64];

349 sum += p[4*64] * q[4*64];

350 sum += p[5*64] * q[5*64];

351 sum += p[6*64] * q[6*64];

352 sum += p[7*64] * q[7*64];

353 tmp[i] = sum;

354 p++;

355 q++;

356 }

357 tmp1[0] = tmp[16] >> WSHIFT;

358 for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;

359 for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;

360

361 idct32(out, tmp1);

362

363 /* advance of 32 samples */

364 offset -= 32;

365 out += 32;

366 /* handle the wrap around */

367 if (offset < 0) {

368 memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),

369 s->samples_buf[ch], (512 - 32) * 2);

370 offset = SAMPLES_BUF_SIZE - 512;

371 }

372 }

373 s->samples_offset[ch] = offset;

374 }

375

376 static void compute_scale_factors(MpegAudioContext *s,

377 unsigned char scale_code[SBLIMIT],

378 unsigned char scale_factors[SBLIMIT][3],

379 int sb_samples[3][12][SBLIMIT],

380 int sblimit)

381 {

382 int *p, vmax, v, n, i, j, k, code;

383 int index, d1, d2;

384 unsigned char *sf = &scale_factors[0][0];

385

386 for(j=0;j<sblimit;j++) {

387 for(i=0;i<3;i++) {

388 /* find the max absolute value */

389 p = &sb_samples[i][0][j];

390 vmax = abs(*p);

391 for(k=1;k<12;k++) {

392 p += SBLIMIT;

393 v = abs(*p);

394 if (v > vmax)

395 vmax = v;

396 }

397 /* compute the scale factor index using log 2 computations */

398 if (vmax > 1) {

399 n = av_log2(vmax);

400 /* n is the position of the MSB of vmax. now

401 use at most 2 compares to find the index */

402 index = (21 - n) * 3 - 3;

403 if (index >= 0) {

404 while (vmax <= s->scale_factor_table[index+1])

405 index++;

406 } else {

407 index = 0; /* very unlikely case of overflow */

408 }

409 } else {

410 index = 62; /* value 63 is not allowed */

411 }

412

413 av_dlog(NULL, "%2d:%d in=%x %x %d\n",

414 j, i, vmax, s->scale_factor_table[index], index);

415 /* store the scale factor */

416 av_assert2(index >=0 && index <= 63);

417 sf[i] = index;

418 }

419

420 /* compute the transmission factor : look if the scale factors

421 are close enough to each other */

422 d1 = s->scale_diff_table[sf[0] - sf[1] + 64];

423 d2 = s->scale_diff_table[sf[1] - sf[2] + 64];

424

425 /* handle the 25 cases */

426 switch(d1 * 5 + d2) {

427 case 0*5+0:

428 case 0*5+4:

429 case 3*5+4:

430 case 4*5+0:

431 case 4*5+4:

432 code = 0;

433 break;

434 case 0*5+1:

435 case 0*5+2:

436 case 4*5+1:

437 case 4*5+2:

438 code = 3;

439 sf[2] = sf[1];

440 break;

441 case 0*5+3:

442 case 4*5+3:

443 code = 3;

444 sf[1] = sf[2];

445 break;

446 case 1*5+0:

447 case 1*5+4:

448 case 2*5+4:

449 code = 1;

450 sf[1] = sf[0];

451 break;

452 case 1*5+1:

453 case 1*5+2:

454 case 2*5+0:

455 case 2*5+1:

456 case 2*5+2:

457 code = 2;

458 sf[1] = sf[2] = sf[0];

459 break;

460 case 2*5+3:

461 case 3*5+3:

462 code = 2;

463 sf[0] = sf[1] = sf[2];

464 break;

465 case 3*5+0:

466 case 3*5+1:

467 case 3*5+2:

468 code = 2;

469 sf[0] = sf[2] = sf[1];

470 break;

471 case 1*5+3:

472 code = 2;

473 if (sf[0] > sf[2])

474 sf[0] = sf[2];

475 sf[1] = sf[2] = sf[0];

476 break;

477 default:

478 av_assert2(0); //cannot happen

479 code = 0; /* kill warning */

480 }

481

482 av_dlog(NULL, "%d: %2d %2d %2d %d %d -> %d\n", j,

483 sf[0], sf[1], sf[2], d1, d2, code);

484 scale_code[j] = code;

485 sf += 3;

486 }

487 }

488

489 /* The most important function : psycho acoustic module. In this

490 encoder there is basically none, so this is the worst you can do,

491 but also this is the simpler. */

492 static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])

493 {

494 int i;

495

496 for(i=0;i<s->sblimit;i++) {

497 smr[i] = (int)(fixed_smr[i] * 10);

498 }

499 }

500

501

502 #define SB_NOTALLOCATED 0

503 #define SB_ALLOCATED 1

504 #define SB_NOMORE 2

505

506 /* Try to maximize the smr while using a number of bits inferior to

507 the frame size. I tried to make the code simpler, faster and

508 smaller than other encoders :-) */

509 static void compute_bit_allocation(MpegAudioContext *s,

510 short smr1[MPA_MAX_CHANNELS][SBLIMIT],

511 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],

512 int *padding)

513 {

514 int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;

515 int incr;

516 short smr[MPA_MAX_CHANNELS][SBLIMIT];

517 unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];

518 const unsigned char *alloc;

519

520 memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);

521 memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);

522 memset(bit_alloc, 0, s->nb_channels * SBLIMIT);

523

524 /* compute frame size and padding */

525 max_frame_size = s->frame_size;

526 s->frame_frac += s->frame_frac_incr;

527 if (s->frame_frac >= 65536) {

528 s->frame_frac -= 65536;

529 s->do_padding = 1;

530 max_frame_size += 8;

531 } else {

532 s->do_padding = 0;

533 }

534

535 /* compute the header + bit alloc size */

536 current_frame_size = 32;

537 alloc = s->alloc_table;

538 for(i=0;i<s->sblimit;i++) {

539 incr = alloc[0];

540 current_frame_size += incr * s->nb_channels;

541 alloc += 1 << incr;

542 }

543 for(;;) {

544 /* look for the subband with the largest signal to mask ratio */

545 max_sb = -1;

546 max_ch = -1;

547 max_smr = INT_MIN;

548 for(ch=0;ch<s->nb_channels;ch++) {

549 for(i=0;i<s->sblimit;i++) {

550 if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {

551 max_smr = smr[ch][i];

552 max_sb = i;

553 max_ch = ch;

554 }

555 }

556 }

557 if (max_sb < 0)

558 break;

559 av_dlog(NULL, "current=%d max=%d max_sb=%d max_ch=%d alloc=%d\n",

560 current_frame_size, max_frame_size, max_sb, max_ch,

561 bit_alloc[max_ch][max_sb]);

562

563 /* find alloc table entry (XXX: not optimal, should use

564 pointer table) */

565 alloc = s->alloc_table;

566 for(i=0;i<max_sb;i++) {

567 alloc += 1 << alloc[0];

568 }

569

570 if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {

571 /* nothing was coded for this band: add the necessary bits */

572 incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;

573 incr += s->total_quant_bits[alloc[1]];

574 } else {

575 /* increments bit allocation */

576 b = bit_alloc[max_ch][max_sb];

577 incr = s->total_quant_bits[alloc[b + 1]] -

578 s->total_quant_bits[alloc[b]];

579 }

580

581 if (current_frame_size + incr <= max_frame_size) {

582 /* can increase size */

583 b = ++bit_alloc[max_ch][max_sb];

584 current_frame_size += incr;

585 /* decrease smr by the resolution we added */

586 smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];

587 /* max allocation size reached ? */

588 if (b == ((1 << alloc[0]) - 1))

589 subband_status[max_ch][max_sb] = SB_NOMORE;

590 else

591 subband_status[max_ch][max_sb] = SB_ALLOCATED;

592 } else {

593 /* cannot increase the size of this subband */

594 subband_status[max_ch][max_sb] = SB_NOMORE;

595 }

596 }

597 *padding = max_frame_size - current_frame_size;

598 av_assert0(*padding >= 0);

599 }

600

601 /*

602 * Output the mpeg audio layer 2 frame. Note how the code is small

603 * compared to other encoders :-)

604 */

605 static void encode_frame(MpegAudioContext *s,

606 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],

607 int padding)

608 {

609 int i, j, k, l, bit_alloc_bits, b, ch;

610 unsigned char *sf;

611 int q[3];

612 PutBitContext *p = &s->pb;

613

614 /* header */

615

616 put_bits(p, 12, 0xfff);

617 put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */

618 put_bits(p, 2, 4-2); /* layer 2 */

619 put_bits(p, 1, 1); /* no error protection */

620 put_bits(p, 4, s->bitrate_index);

621 put_bits(p, 2, s->freq_index);

622 put_bits(p, 1, s->do_padding); /* use padding */

623 put_bits(p, 1, 0); /* private_bit */

624 put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);

625 put_bits(p, 2, 0); /* mode_ext */

626 put_bits(p, 1, 0); /* no copyright */

627 put_bits(p, 1, 1); /* original */

628 put_bits(p, 2, 0); /* no emphasis */

629

630 /* bit allocation */

631 j = 0;

632 for(i=0;i<s->sblimit;i++) {

633 bit_alloc_bits = s->alloc_table[j];

634 for(ch=0;ch<s->nb_channels;ch++) {

635 put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);

636 }

637 j += 1 << bit_alloc_bits;

638 }

639

640 /* scale codes */

641 for(i=0;i<s->sblimit;i++) {

642 for(ch=0;ch<s->nb_channels;ch++) {

643 if (bit_alloc[ch][i])

644 put_bits(p, 2, s->scale_code[ch][i]);

645 }

646 }

647

648 /* scale factors */

649 for(i=0;i<s->sblimit;i++) {

650 for(ch=0;ch<s->nb_channels;ch++) {

651 if (bit_alloc[ch][i]) {

652 sf = &s->scale_factors[ch][i][0];

653 switch(s->scale_code[ch][i]) {

654 case 0:

655 put_bits(p, 6, sf[0]);

656 put_bits(p, 6, sf[1]);

657 put_bits(p, 6, sf[2]);

658 break;

659 case 3:

660 case 1:

661 put_bits(p, 6, sf[0]);

662 put_bits(p, 6, sf[2]);

663 break;

664 case 2:

665 put_bits(p, 6, sf[0]);

666 break;

667 }

668 }

669 }

670 }

671

672 /* quantization & write sub band samples */

673

674 for(k=0;k<3;k++) {

675 for(l=0;l<12;l+=3) {

676 j = 0;

677 for(i=0;i<s->sblimit;i++) {

678 bit_alloc_bits = s->alloc_table[j];

679 for(ch=0;ch<s->nb_channels;ch++) {

680 b = bit_alloc[ch][i];

681 if (b) {

682 int qindex, steps, m, sample, bits;

683 /* we encode 3 sub band samples of the same sub band at a time */

684 qindex = s->alloc_table[j+b];

685 steps = ff_mpa_quant_steps[qindex];

686 for(m=0;m<3;m++) {

687 sample = s->sb_samples[ch][k][l + m][i];

688 /* divide by scale factor */

689 #if USE_FLOATS

690 {

691 float a;

692 a = (float)sample * s->scale_factor_inv_table[s->scale_factors[ch][i][k]];

693 q[m] = (int)((a + 1.0) * steps * 0.5);

694 }

695 #else

696 {

697 int q1, e, shift, mult;

698 e = s->scale_factors[ch][i][k];

699 shift = s->scale_factor_shift[e];

700 mult = s->scale_factor_mult[e];

701

702 /* normalize to P bits */

703 if (shift < 0)

704 q1 = sample << (-shift);

705 else

706 q1 = sample >> shift;

707 q1 = (q1 * mult) >> P;

708 q1 += 1 << P;

709 if (q1 < 0)

710 q1 = 0;

711 q[m] = (q1 * (unsigned)steps) >> (P + 1);

712 }

713 #endif

714 if (q[m] >= steps)

715 q[m] = steps - 1;

716 av_assert2(q[m] >= 0 && q[m] < steps);

717 }

718 bits = ff_mpa_quant_bits[qindex];

719 if (bits < 0) {

720 /* group the 3 values to save bits */

721 put_bits(p, -bits,

722 q[0] + steps * (q[1] + steps * q[2]));

723 } else {

724 put_bits(p, bits, q[0]);

725 put_bits(p, bits, q[1]);

726 put_bits(p, bits, q[2]);

727 }

728 }

729 }

730 /* next subband in alloc table */

731 j += 1 << bit_alloc_bits;

732 }

733 }

734 }

735

736 /* padding */

737 for(i=0;i<padding;i++)

738 put_bits(p, 1, 0);

739

740 /* flush */

741 flush_put_bits(p);

742 }

743

744 static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,

745 const AVFrame *frame, int *got_packet_ptr)

746 {

747 MpegAudioContext *s = avctx->priv_data;

748 const int16_t *samples = (const int16_t *)frame->data[0];

749 short smr[MPA_MAX_CHANNELS][SBLIMIT];

750 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];

751 int padding, i, ret;

752

753 for(i=0;i<s->nb_channels;i++) {

754 filter(s, i, samples + i, s->nb_channels);

755 }

756

757 for(i=0;i<s->nb_channels;i++) {

758 compute_scale_factors(s, s->scale_code[i], s->scale_factors[i],

759 s->sb_samples[i], s->sblimit);

760 }

761 for(i=0;i<s->nb_channels;i++) {

762 psycho_acoustic_model(s, smr[i]);

763 }

764 compute_bit_allocation(s, smr, bit_alloc, &padding);

765

766 if ((ret = ff_alloc_packet2(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE)) < 0)

767 return ret;

768

769 init_put_bits(&s->pb, avpkt->data, avpkt->size);

770

771 encode_frame(s, bit_alloc, padding);

772

773 if (frame->pts != AV_NOPTS_VALUE)

774 avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->initial_padding);

775

776 avpkt->size = put_bits_count(&s->pb) / 8;

777 *got_packet_ptr = 1;

778 return 0;

779 }

780

781 static const AVCodecDefault mp2_defaults[] = {

782 { "b", "0" },

783 { NULL },

784 };

785

Generated on Sun Mar 8 2015 02:34:57 for FFmpeg by doxygen 1.8.2