FFmpeg: libavcodec/mpegaudioenc.c Source File

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mpegaudioenc.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"

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

40 quantization stage) */

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

43 #define SAMPLES_BUF_SIZE 4096

45 typedef struct MpegAudioContext {

46 PutBitContext pb;

47 int nb_channels;

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

49 int bitrate_index; /* bit rate */

50 int freq_index;

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

52 /* padding computation */

53 int frame_frac, frame_frac_incr, do_padding;

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

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

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

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

58 /* code to group 3 scale factors */

59 unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];

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

61 const unsigned char *alloc_table;

62 } MpegAudioContext;

64 /* define it to use floats in quantization (I don't like floats !) */

65 #define USE_FLOATS

67 #include "mpegaudiodata.h"

68 #include "mpegaudiotab.h"

70 static av_cold int MPA_encode_init(AVCodecContext *avctx)

71 {

72 MpegAudioContext *s = avctx->priv_data;

73 int freq = avctx->sample_rate;

74 int bitrate = avctx->bit_rate;

75 int channels = avctx->channels;

76 int i, v, table;

77 float a;

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

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

81 return AVERROR(EINVAL);

82 }

83 bitrate = bitrate / 1000;

84 s->nb_channels = channels;

85 avctx->frame_size = MPA_FRAME_SIZE;

86 avctx->delay = 512 - 32 + 1;

88 /* encoding freq */

89 s->lsf = 0;

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

91 if (avpriv_mpa_freq_tab[i] == freq)

92 break;

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

94 s->lsf = 1;

95 break;

96 }

97 }

98 if (i == 3){

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

100 return AVERROR(EINVAL);

101 }

102 s->freq_index = i;

103

104 /* encoding bitrate & frequency */

105 for(i=0;i<15;i++) {

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

107 break;

108 }

109 if (i == 15){

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

111 return AVERROR(EINVAL);

112 }

113 s->bitrate_index = i;

114

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

116

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

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

119

120 /* frame fractional size to compute padding */

121 s->frame_frac = 0;

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

123

124 /* select the right allocation table */

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

126

127 /* number of used subbands */

128 s->sblimit = ff_mpa_sblimit_table[table];

129 s->alloc_table = ff_mpa_alloc_tables[table];

130

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

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

133

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

135 s->samples_offset[i] = 0;

136

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

138 int v;

139 v = ff_mpa_enwindow[i];

140 #if WFRAC_BITS != 16

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

142 #endif

143 filter_bank[i] = v;

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

145 v = -v;

146 if (i != 0)

147 filter_bank[512 - i] = v;

148 }

149

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

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

152 if (v <= 0)

153 v = 1;

154 scale_factor_table[i] = v;

155 #ifdef USE_FLOATS

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

157 #else

158 #define P 15

159 scale_factor_shift[i] = 21 - P - (i / 3);

160 scale_factor_mult[i] = (1 << P) * exp2((i % 3) / 3.0);

161 #endif

162 }

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

164 v = i - 64;

165 if (v <= -3)

166 v = 0;

167 else if (v < 0)

168 v = 1;

169 else if (v == 0)

170 v = 2;

171 else if (v < 3)

172 v = 3;

173 else

174 v = 4;

175 scale_diff_table[i] = v;

176 }

177

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

179 v = ff_mpa_quant_bits[i];

180 if (v < 0)

181 v = -v;

182 else

183 v = v * 3;

184 total_quant_bits[i] = 12 * v;

185 }

186

187 #if FF_API_OLD_ENCODE_AUDIO

188 avctx->coded_frame= avcodec_alloc_frame();

189 if (!avctx->coded_frame)

190 return AVERROR(ENOMEM);

191 #endif

192

193 return 0;

194 }

195

196 /* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */

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

198 {

199 int i, j;

200 int *t, *t1, xr;

201 const int *xp = costab32;

202

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

204

205 t = tab + 30;

206 t1 = tab + 2;

207 do {

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

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

210 t -= 4;

211 } while (t != t1);

212

213 t = tab + 28;

214 t1 = tab + 4;

215 do {

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

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

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

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

220 t -= 8;

221 } while (t != t1);

222

223 t = tab;

224 t1 = tab + 32;

225 do {

226 t[ 3] = -t[ 3];

227 t[ 6] = -t[ 6];

228

229 t[11] = -t[11];

230 t[12] = -t[12];

231 t[13] = -t[13];

232 t[15] = -t[15];

233 t += 16;

234 } while (t != t1);

235

236

237 t = tab;

238 t1 = tab + 8;

239 do {

240 int x1, x2, x3, x4;

241

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

243 x4 = t[0] - x3;

244 x3 = t[0] + x3;

245

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

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

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

249

250 t[ 0] = x3 + x1;

251 t[ 8] = x4 - x2;

252 t[16] = x4 + x2;

253 t[24] = x3 - x1;

254 t++;

255 } while (t != t1);

256

257 xp += 2;

258 t = tab;

259 t1 = tab + 4;

260 do {

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

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

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

264

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

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

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

268

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

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

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

272

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

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

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

276 t++;

277 } while (t != t1);

278 xp += 4;

279

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

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

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

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

284

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

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

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

288

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

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

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

292

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

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

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

296

297 xp += 2;

298 }

299

300 t = tab + 30;

301 t1 = tab + 1;

302 do {

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

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

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

306 t -= 2;

307 t1 += 2;

308 xp++;

309 } while (t >= tab);

310

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

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

313 }

314 }

315

316 #define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)

317

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

319 {

320 short *p, *q;

321 int sum, offset, i, j;

322 int tmp[64];

323 int tmp1[32];

324 int *out;

325

326 offset = s->samples_offset[ch];

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

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

329 /* 32 samples at once */

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

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

332 samples += incr;

333 }

334

335 /* filter */

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

337 q = filter_bank;

338 /* maxsum = 23169 */

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

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

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

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

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

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

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

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

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

348 tmp[i] = sum;

349 p++;

350 q++;

351 }

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

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

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

355

356 idct32(out, tmp1);

357

358 /* advance of 32 samples */

359 offset -= 32;

360 out += 32;

361 /* handle the wrap around */

362 if (offset < 0) {

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

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

365 offset = SAMPLES_BUF_SIZE - 512;

366 }

367 }

368 s->samples_offset[ch] = offset;

369 }

370

371 static void compute_scale_factors(unsigned char scale_code[SBLIMIT],

372 unsigned char scale_factors[SBLIMIT][3],

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

374 int sblimit)

375 {

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

377 int index, d1, d2;

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

379

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

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

382 /* find the max absolute value */

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

384 vmax = abs(*p);

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

386 p += SBLIMIT;

387 v = abs(*p);

388 if (v > vmax)

389 vmax = v;

390 }

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

392 if (vmax > 1) {

393 n = av_log2(vmax);

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

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

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

397 if (index >= 0) {

398 while (vmax <= scale_factor_table[index+1])

399 index++;

400 } else {

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

402 }

403 } else {

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

405 }

406

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

408 j, i, vmax, scale_factor_table[index], index);

409 /* store the scale factor */

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

411 sf[i] = index;

412 }

413

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

415 are close enough to each other */

416 d1 = scale_diff_table[sf[0] - sf[1] + 64];

417 d2 = scale_diff_table[sf[1] - sf[2] + 64];

418

419 /* handle the 25 cases */

420 switch(d1 * 5 + d2) {

421 case 0*5+0:

422 case 0*5+4:

423 case 3*5+4:

424 case 4*5+0:

425 case 4*5+4:

426 code = 0;

427 break;

428 case 0*5+1:

429 case 0*5+2:

430 case 4*5+1:

431 case 4*5+2:

432 code = 3;

433 sf[2] = sf[1];

434 break;

435 case 0*5+3:

436 case 4*5+3:

437 code = 3;

438 sf[1] = sf[2];

439 break;

440 case 1*5+0:

441 case 1*5+4:

442 case 2*5+4:

443 code = 1;

444 sf[1] = sf[0];

445 break;

446 case 1*5+1:

447 case 1*5+2:

448 case 2*5+0:

449 case 2*5+1:

450 case 2*5+2:

451 code = 2;

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

453 break;

454 case 2*5+3:

455 case 3*5+3:

456 code = 2;

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

458 break;

459 case 3*5+0:

460 case 3*5+1:

461 case 3*5+2:

462 code = 2;

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

464 break;

465 case 1*5+3:

466 code = 2;

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

468 sf[0] = sf[2];

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

470 break;

471 default:

472 av_assert2(0); //cannot happen

473 code = 0; /* kill warning */

474 }

475

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

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

478 scale_code[j] = code;

479 sf += 3;

480 }

481 }

482

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

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

485 but also this is the simpler. */

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

487 {

488 int i;

489

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

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

492 }

493 }

494

495

496 #define SB_NOTALLOCATED 0

497 #define SB_ALLOCATED 1

498 #define SB_NOMORE 2

499

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

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

502 smaller than other encoders :-) */

503 static void compute_bit_allocation(MpegAudioContext *s,

504 short smr1[MPA_MAX_CHANNELS][SBLIMIT],

505 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],

506 int *padding)

507 {

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

509 int incr;

510 short smr[MPA_MAX_CHANNELS][SBLIMIT];

511 unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];

512 const unsigned char *alloc;

513

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

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

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

517

518 /* compute frame size and padding */

519 max_frame_size = s->frame_size;

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

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

522 s->frame_frac -= 65536;

523 s->do_padding = 1;

524 max_frame_size += 8;

525 } else {

526 s->do_padding = 0;

527 }

528

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

530 current_frame_size = 32;

531 alloc = s->alloc_table;

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

533 incr = alloc[0];

534 current_frame_size += incr * s->nb_channels;

535 alloc += 1 << incr;

536 }

537 for(;;) {

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

539 max_sb = -1;

540 max_ch = -1;

541 max_smr = INT_MIN;

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

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

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

545 max_smr = smr[ch][i];

546 max_sb = i;

547 max_ch = ch;

548 }

549 }

550 }

551 if (max_sb < 0)

552 break;

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

554 current_frame_size, max_frame_size, max_sb, max_ch,

555 bit_alloc[max_ch][max_sb]);

556

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

558 pointer table) */

559 alloc = s->alloc_table;

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

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

562 }

563

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

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

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

567 incr += total_quant_bits[alloc[1]];

568 } else {

569 /* increments bit allocation */

570 b = bit_alloc[max_ch][max_sb];

571 incr = total_quant_bits[alloc[b + 1]] -

572 total_quant_bits[alloc[b]];

573 }

574

575 if (current_frame_size + incr <= max_frame_size) {

576 /* can increase size */

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

578 current_frame_size += incr;

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

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

581 /* max allocation size reached ? */

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

583 subband_status[max_ch][max_sb] = SB_NOMORE;

584 else

585 subband_status[max_ch][max_sb] = SB_ALLOCATED;

586 } else {

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

588 subband_status[max_ch][max_sb] = SB_NOMORE;

589 }

590 }

591 *padding = max_frame_size - current_frame_size;

592 av_assert0(*padding >= 0);

593 }

594

595 /*

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

597 * compared to other encoders :-)

598 */

599 static void encode_frame(MpegAudioContext *s,

600 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],

601 int padding)

602 {

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

604 unsigned char *sf;

605 int q[3];

606 PutBitContext *p = &s->pb;

607

608 /* header */

609

610 put_bits(p, 12, 0xfff);

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

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

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

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

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

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

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

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

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

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

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

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

623

624 /* bit allocation */

625 j = 0;

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

627 bit_alloc_bits = s->alloc_table[j];

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

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

630 }

631 j += 1 << bit_alloc_bits;

632 }

633

634 /* scale codes */

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

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

637 if (bit_alloc[ch][i])

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

639 }

640 }

641

642 /* scale factors */

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

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

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

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

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

648 case 0:

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

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

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

652 break;

653 case 3:

654 case 1:

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

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

657 break;

658 case 2:

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

660 break;

661 }

662 }

663 }

664 }

665

666 /* quantization & write sub band samples */

667

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

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

670 j = 0;

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

672 bit_alloc_bits = s->alloc_table[j];

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

674 b = bit_alloc[ch][i];

675 if (b) {

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

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

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

679 steps = ff_mpa_quant_steps[qindex];

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

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

682 /* divide by scale factor */

683 #ifdef USE_FLOATS

684 {

685 float a;

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

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

688 }

689 #else

690 {

691 int q1, e, shift, mult;

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

693 shift = scale_factor_shift[e];

694 mult = scale_factor_mult[e];

695

696 /* normalize to P bits */

697 if (shift < 0)

698 q1 = sample << (-shift);

699 else

700 q1 = sample >> shift;

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

702 q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);

703 }

704 #endif

705 if (q[m] >= steps)

706 q[m] = steps - 1;

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

708 }

709 bits = ff_mpa_quant_bits[qindex];

710 if (bits < 0) {

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

712 put_bits(p, -bits,

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

714 } else {

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

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

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

718 }

719 }

720 }

721 /* next subband in alloc table */

722 j += 1 << bit_alloc_bits;

723 }

724 }

725 }

726

727 /* padding */

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

729 put_bits(p, 1, 0);

730

731 /* flush */

732 flush_put_bits(p);

733 }

734

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

736 const AVFrame *frame, int *got_packet_ptr)

737 {

738 MpegAudioContext *s = avctx->priv_data;

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

740 short smr[MPA_MAX_CHANNELS][SBLIMIT];

741 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];

742 int padding, i, ret;

743

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

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

746 }

747

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

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

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

751 }

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

753 psycho_acoustic_model(s, smr[i]);

754 }

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

756

757 if ((ret = ff_alloc_packet2(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE)))

758 return ret;

759

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

761

762 encode_frame(s, bit_alloc, padding);

763

764 if (frame->pts != AV_NOPTS_VALUE)

765 avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);

766

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

768 *got_packet_ptr = 1;

769 return 0;

770 }

771

772 static av_cold int MPA_encode_close(AVCodecContext *avctx)

773 {

774 #if FF_API_OLD_ENCODE_AUDIO

775 av_freep(&avctx->coded_frame);

776 #endif

777 return 0;

778 }

779

780 static const AVCodecDefault mp2_defaults[] = {

781 { "b", "128k" },

782 { NULL },

783 };

784

785 AVCodec ff_mp2_encoder = {

786 .name = "mp2",

787 .type = AVMEDIA_TYPE_AUDIO,

788 .id = AV_CODEC_ID_MP2,

789 .priv_data_size = sizeof(MpegAudioContext),

790 .init = MPA_encode_init,

791 .encode2 = MPA_encode_frame,

792 .close = MPA_encode_close,

793 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,

794 AV_SAMPLE_FMT_NONE },

795 .supported_samplerates = (const int[]){

796 44100, 48000, 32000, 22050, 24000, 16000, 0

797 },

798 .channel_layouts = (const uint64_t[]){ AV_CH_LAYOUT_MONO,

799 AV_CH_LAYOUT_STEREO,

800 0 },

801 .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),

802 .defaults = mp2_defaults,

803 };

Generated on Sat May 25 2013 03:58:38 for FFmpeg by doxygen 1.8.2