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  1. /*
  2. * The simplest mpeg audio layer 2 encoder
  3. * Copyright (c) 2000, 2001 Fabrice Bellard.
  4. *
  5. * This library is free software; you can redistribute it and/or
  6. * modify it under the terms of the GNU Lesser General Public
  7. * License as published by the Free Software Foundation; either
  8. * version 2 of the License, or (at your option) any later version.
  9. *
  10. * This library is distributed in the hope that it will be useful,
  11. * but WITHOUT ANY WARRANTY; without even the implied warranty of
  12. * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
  13. * Lesser General Public License for more details.
  14. *
  15. * You should have received a copy of the GNU Lesser General Public
  16. * License along with this library; if not, write to the Free Software
  17. * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
  18. */
  19. /**
  20. * @file mpegaudio.c
  21. * The simplest mpeg audio layer 2 encoder.
  22. */
  23. #include "avcodec.h"
  24. #include "mpegaudio.h"
  25. /* currently, cannot change these constants (need to modify
  26. quantization stage) */
  27. #define FRAC_BITS 15
  28. #define WFRAC_BITS 14
  29. #define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
  30. #define FIX(a) ((int)((a) * (1 << FRAC_BITS)))
  31. #define SAMPLES_BUF_SIZE 4096
  32. typedef struct MpegAudioContext {
  33. PutBitContext pb;
  34. int nb_channels;
  35. int freq, bit_rate;
  36. int lsf; /* 1 if mpeg2 low bitrate selected */
  37. int bitrate_index; /* bit rate */
  38. int freq_index;
  39. int frame_size; /* frame size, in bits, without padding */
  40. int64_t nb_samples; /* total number of samples encoded */
  41. /* padding computation */
  42. int frame_frac, frame_frac_incr, do_padding;
  43. short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
  44. int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */
  45. int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
  46. unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
  47. /* code to group 3 scale factors */
  48. unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
  49. int sblimit; /* number of used subbands */
  50. const unsigned char *alloc_table;
  51. } MpegAudioContext;
  52. /* define it to use floats in quantization (I don't like floats !) */
  53. //#define USE_FLOATS
  54. #include "mpegaudiotab.h"
  55. static int MPA_encode_init(AVCodecContext *avctx)
  56. {
  57. MpegAudioContext *s = avctx->priv_data;
  58. int freq = avctx->sample_rate;
  59. int bitrate = avctx->bit_rate;
  60. int channels = avctx->channels;
  61. int i, v, table;
  62. float a;
  63. if (channels > 2)
  64. return -1;
  65. bitrate = bitrate / 1000;
  66. s->nb_channels = channels;
  67. s->freq = freq;
  68. s->bit_rate = bitrate * 1000;
  69. avctx->frame_size = MPA_FRAME_SIZE;
  70. /* encoding freq */
  71. s->lsf = 0;
  72. for(i=0;i<3;i++) {
  73. if (mpa_freq_tab[i] == freq)
  74. break;
  75. if ((mpa_freq_tab[i] / 2) == freq) {
  76. s->lsf = 1;
  77. break;
  78. }
  79. }
  80. if (i == 3)
  81. return -1;
  82. s->freq_index = i;
  83. /* encoding bitrate & frequency */
  84. for(i=0;i<15;i++) {
  85. if (mpa_bitrate_tab[s->lsf][1][i] == bitrate)
  86. break;
  87. }
  88. if (i == 15)
  89. return -1;
  90. s->bitrate_index = i;
  91. /* compute total header size & pad bit */
  92. a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
  93. s->frame_size = ((int)a) * 8;
  94. /* frame fractional size to compute padding */
  95. s->frame_frac = 0;
  96. s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
  97. /* select the right allocation table */
  98. table = l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
  99. /* number of used subbands */
  100. s->sblimit = sblimit_table[table];
  101. s->alloc_table = alloc_tables[table];
  102. #ifdef DEBUG
  103. printf("%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
  104. bitrate, freq, s->frame_size, table, s->frame_frac_incr);
  105. #endif
  106. for(i=0;i<s->nb_channels;i++)
  107. s->samples_offset[i] = 0;
  108. for(i=0;i<257;i++) {
  109. int v;
  110. v = mpa_enwindow[i];
  111. #if WFRAC_BITS != 16
  112. v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
  113. #endif
  114. filter_bank[i] = v;
  115. if ((i & 63) != 0)
  116. v = -v;
  117. if (i != 0)
  118. filter_bank[512 - i] = v;
  119. }
  120. for(i=0;i<64;i++) {
  121. v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
  122. if (v <= 0)
  123. v = 1;
  124. scale_factor_table[i] = v;
  125. #ifdef USE_FLOATS
  126. scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
  127. #else
  128. #define P 15
  129. scale_factor_shift[i] = 21 - P - (i / 3);
  130. scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
  131. #endif
  132. }
  133. for(i=0;i<128;i++) {
  134. v = i - 64;
  135. if (v <= -3)
  136. v = 0;
  137. else if (v < 0)
  138. v = 1;
  139. else if (v == 0)
  140. v = 2;
  141. else if (v < 3)
  142. v = 3;
  143. else
  144. v = 4;
  145. scale_diff_table[i] = v;
  146. }
  147. for(i=0;i<17;i++) {
  148. v = quant_bits[i];
  149. if (v < 0)
  150. v = -v;
  151. else
  152. v = v * 3;
  153. total_quant_bits[i] = 12 * v;
  154. }
  155. avctx->coded_frame= avcodec_alloc_frame();
  156. avctx->coded_frame->key_frame= 1;
  157. return 0;
  158. }
  159. /* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
  160. static void idct32(int *out, int *tab)
  161. {
  162. int i, j;
  163. int *t, *t1, xr;
  164. const int *xp = costab32;
  165. for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
  166. t = tab + 30;
  167. t1 = tab + 2;
  168. do {
  169. t[0] += t[-4];
  170. t[1] += t[1 - 4];
  171. t -= 4;
  172. } while (t != t1);
  173. t = tab + 28;
  174. t1 = tab + 4;
  175. do {
  176. t[0] += t[-8];
  177. t[1] += t[1-8];
  178. t[2] += t[2-8];
  179. t[3] += t[3-8];
  180. t -= 8;
  181. } while (t != t1);
  182. t = tab;
  183. t1 = tab + 32;
  184. do {
  185. t[ 3] = -t[ 3];
  186. t[ 6] = -t[ 6];
  187. t[11] = -t[11];
  188. t[12] = -t[12];
  189. t[13] = -t[13];
  190. t[15] = -t[15];
  191. t += 16;
  192. } while (t != t1);
  193. t = tab;
  194. t1 = tab + 8;
  195. do {
  196. int x1, x2, x3, x4;
  197. x3 = MUL(t[16], FIX(SQRT2*0.5));
  198. x4 = t[0] - x3;
  199. x3 = t[0] + x3;
  200. x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
  201. x1 = MUL((t[8] - x2), xp[0]);
  202. x2 = MUL((t[8] + x2), xp[1]);
  203. t[ 0] = x3 + x1;
  204. t[ 8] = x4 - x2;
  205. t[16] = x4 + x2;
  206. t[24] = x3 - x1;
  207. t++;
  208. } while (t != t1);
  209. xp += 2;
  210. t = tab;
  211. t1 = tab + 4;
  212. do {
  213. xr = MUL(t[28],xp[0]);
  214. t[28] = (t[0] - xr);
  215. t[0] = (t[0] + xr);
  216. xr = MUL(t[4],xp[1]);
  217. t[ 4] = (t[24] - xr);
  218. t[24] = (t[24] + xr);
  219. xr = MUL(t[20],xp[2]);
  220. t[20] = (t[8] - xr);
  221. t[ 8] = (t[8] + xr);
  222. xr = MUL(t[12],xp[3]);
  223. t[12] = (t[16] - xr);
  224. t[16] = (t[16] + xr);
  225. t++;
  226. } while (t != t1);
  227. xp += 4;
  228. for (i = 0; i < 4; i++) {
  229. xr = MUL(tab[30-i*4],xp[0]);
  230. tab[30-i*4] = (tab[i*4] - xr);
  231. tab[ i*4] = (tab[i*4] + xr);
  232. xr = MUL(tab[ 2+i*4],xp[1]);
  233. tab[ 2+i*4] = (tab[28-i*4] - xr);
  234. tab[28-i*4] = (tab[28-i*4] + xr);
  235. xr = MUL(tab[31-i*4],xp[0]);
  236. tab[31-i*4] = (tab[1+i*4] - xr);
  237. tab[ 1+i*4] = (tab[1+i*4] + xr);
  238. xr = MUL(tab[ 3+i*4],xp[1]);
  239. tab[ 3+i*4] = (tab[29-i*4] - xr);
  240. tab[29-i*4] = (tab[29-i*4] + xr);
  241. xp += 2;
  242. }
  243. t = tab + 30;
  244. t1 = tab + 1;
  245. do {
  246. xr = MUL(t1[0], *xp);
  247. t1[0] = (t[0] - xr);
  248. t[0] = (t[0] + xr);
  249. t -= 2;
  250. t1 += 2;
  251. xp++;
  252. } while (t >= tab);
  253. for(i=0;i<32;i++) {
  254. out[i] = tab[bitinv32[i]];
  255. }
  256. }
  257. #define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
  258. static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
  259. {
  260. short *p, *q;
  261. int sum, offset, i, j;
  262. int tmp[64];
  263. int tmp1[32];
  264. int *out;
  265. // print_pow1(samples, 1152);
  266. offset = s->samples_offset[ch];
  267. out = &s->sb_samples[ch][0][0][0];
  268. for(j=0;j<36;j++) {
  269. /* 32 samples at once */
  270. for(i=0;i<32;i++) {
  271. s->samples_buf[ch][offset + (31 - i)] = samples[0];
  272. samples += incr;
  273. }
  274. /* filter */
  275. p = s->samples_buf[ch] + offset;
  276. q = filter_bank;
  277. /* maxsum = 23169 */
  278. for(i=0;i<64;i++) {
  279. sum = p[0*64] * q[0*64];
  280. sum += p[1*64] * q[1*64];
  281. sum += p[2*64] * q[2*64];
  282. sum += p[3*64] * q[3*64];
  283. sum += p[4*64] * q[4*64];
  284. sum += p[5*64] * q[5*64];
  285. sum += p[6*64] * q[6*64];
  286. sum += p[7*64] * q[7*64];
  287. tmp[i] = sum;
  288. p++;
  289. q++;
  290. }
  291. tmp1[0] = tmp[16] >> WSHIFT;
  292. for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
  293. for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
  294. idct32(out, tmp1);
  295. /* advance of 32 samples */
  296. offset -= 32;
  297. out += 32;
  298. /* handle the wrap around */
  299. if (offset < 0) {
  300. memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
  301. s->samples_buf[ch], (512 - 32) * 2);
  302. offset = SAMPLES_BUF_SIZE - 512;
  303. }
  304. }
  305. s->samples_offset[ch] = offset;
  306. // print_pow(s->sb_samples, 1152);
  307. }
  308. static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
  309. unsigned char scale_factors[SBLIMIT][3],
  310. int sb_samples[3][12][SBLIMIT],
  311. int sblimit)
  312. {
  313. int *p, vmax, v, n, i, j, k, code;
  314. int index, d1, d2;
  315. unsigned char *sf = &scale_factors[0][0];
  316. for(j=0;j<sblimit;j++) {
  317. for(i=0;i<3;i++) {
  318. /* find the max absolute value */
  319. p = &sb_samples[i][0][j];
  320. vmax = abs(*p);
  321. for(k=1;k<12;k++) {
  322. p += SBLIMIT;
  323. v = abs(*p);
  324. if (v > vmax)
  325. vmax = v;
  326. }
  327. /* compute the scale factor index using log 2 computations */
  328. if (vmax > 0) {
  329. n = av_log2(vmax);
  330. /* n is the position of the MSB of vmax. now
  331. use at most 2 compares to find the index */
  332. index = (21 - n) * 3 - 3;
  333. if (index >= 0) {
  334. while (vmax <= scale_factor_table[index+1])
  335. index++;
  336. } else {
  337. index = 0; /* very unlikely case of overflow */
  338. }
  339. } else {
  340. index = 62; /* value 63 is not allowed */
  341. }
  342. #if 0
  343. printf("%2d:%d in=%x %x %d\n",
  344. j, i, vmax, scale_factor_table[index], index);
  345. #endif
  346. /* store the scale factor */
  347. assert(index >=0 && index <= 63);
  348. sf[i] = index;
  349. }
  350. /* compute the transmission factor : look if the scale factors
  351. are close enough to each other */
  352. d1 = scale_diff_table[sf[0] - sf[1] + 64];
  353. d2 = scale_diff_table[sf[1] - sf[2] + 64];
  354. /* handle the 25 cases */
  355. switch(d1 * 5 + d2) {
  356. case 0*5+0:
  357. case 0*5+4:
  358. case 3*5+4:
  359. case 4*5+0:
  360. case 4*5+4:
  361. code = 0;
  362. break;
  363. case 0*5+1:
  364. case 0*5+2:
  365. case 4*5+1:
  366. case 4*5+2:
  367. code = 3;
  368. sf[2] = sf[1];
  369. break;
  370. case 0*5+3:
  371. case 4*5+3:
  372. code = 3;
  373. sf[1] = sf[2];
  374. break;
  375. case 1*5+0:
  376. case 1*5+4:
  377. case 2*5+4:
  378. code = 1;
  379. sf[1] = sf[0];
  380. break;
  381. case 1*5+1:
  382. case 1*5+2:
  383. case 2*5+0:
  384. case 2*5+1:
  385. case 2*5+2:
  386. code = 2;
  387. sf[1] = sf[2] = sf[0];
  388. break;
  389. case 2*5+3:
  390. case 3*5+3:
  391. code = 2;
  392. sf[0] = sf[1] = sf[2];
  393. break;
  394. case 3*5+0:
  395. case 3*5+1:
  396. case 3*5+2:
  397. code = 2;
  398. sf[0] = sf[2] = sf[1];
  399. break;
  400. case 1*5+3:
  401. code = 2;
  402. if (sf[0] > sf[2])
  403. sf[0] = sf[2];
  404. sf[1] = sf[2] = sf[0];
  405. break;
  406. default:
  407. av_abort();
  408. }
  409. #if 0
  410. printf("%d: %2d %2d %2d %d %d -> %d\n", j,
  411. sf[0], sf[1], sf[2], d1, d2, code);
  412. #endif
  413. scale_code[j] = code;
  414. sf += 3;
  415. }
  416. }
  417. /* The most important function : psycho acoustic module. In this
  418. encoder there is basically none, so this is the worst you can do,
  419. but also this is the simpler. */
  420. static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
  421. {
  422. int i;
  423. for(i=0;i<s->sblimit;i++) {
  424. smr[i] = (int)(fixed_smr[i] * 10);
  425. }
  426. }
  427. #define SB_NOTALLOCATED 0
  428. #define SB_ALLOCATED 1
  429. #define SB_NOMORE 2
  430. /* Try to maximize the smr while using a number of bits inferior to
  431. the frame size. I tried to make the code simpler, faster and
  432. smaller than other encoders :-) */
  433. static void compute_bit_allocation(MpegAudioContext *s,
  434. short smr1[MPA_MAX_CHANNELS][SBLIMIT],
  435. unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
  436. int *padding)
  437. {
  438. int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
  439. int incr;
  440. short smr[MPA_MAX_CHANNELS][SBLIMIT];
  441. unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
  442. const unsigned char *alloc;
  443. memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
  444. memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
  445. memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
  446. /* compute frame size and padding */
  447. max_frame_size = s->frame_size;
  448. s->frame_frac += s->frame_frac_incr;
  449. if (s->frame_frac >= 65536) {
  450. s->frame_frac -= 65536;
  451. s->do_padding = 1;
  452. max_frame_size += 8;
  453. } else {
  454. s->do_padding = 0;
  455. }
  456. /* compute the header + bit alloc size */
  457. current_frame_size = 32;
  458. alloc = s->alloc_table;
  459. for(i=0;i<s->sblimit;i++) {
  460. incr = alloc[0];
  461. current_frame_size += incr * s->nb_channels;
  462. alloc += 1 << incr;
  463. }
  464. for(;;) {
  465. /* look for the subband with the largest signal to mask ratio */
  466. max_sb = -1;
  467. max_ch = -1;
  468. max_smr = 0x80000000;
  469. for(ch=0;ch<s->nb_channels;ch++) {
  470. for(i=0;i<s->sblimit;i++) {
  471. if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
  472. max_smr = smr[ch][i];
  473. max_sb = i;
  474. max_ch = ch;
  475. }
  476. }
  477. }
  478. #if 0
  479. printf("current=%d max=%d max_sb=%d alloc=%d\n",
  480. current_frame_size, max_frame_size, max_sb,
  481. bit_alloc[max_sb]);
  482. #endif
  483. if (max_sb < 0)
  484. break;
  485. /* find alloc table entry (XXX: not optimal, should use
  486. pointer table) */
  487. alloc = s->alloc_table;
  488. for(i=0;i<max_sb;i++) {
  489. alloc += 1 << alloc[0];
  490. }
  491. if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
  492. /* nothing was coded for this band: add the necessary bits */
  493. incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
  494. incr += total_quant_bits[alloc[1]];
  495. } else {
  496. /* increments bit allocation */
  497. b = bit_alloc[max_ch][max_sb];
  498. incr = total_quant_bits[alloc[b + 1]] -
  499. total_quant_bits[alloc[b]];
  500. }
  501. if (current_frame_size + incr <= max_frame_size) {
  502. /* can increase size */
  503. b = ++bit_alloc[max_ch][max_sb];
  504. current_frame_size += incr;
  505. /* decrease smr by the resolution we added */
  506. smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
  507. /* max allocation size reached ? */
  508. if (b == ((1 << alloc[0]) - 1))
  509. subband_status[max_ch][max_sb] = SB_NOMORE;
  510. else
  511. subband_status[max_ch][max_sb] = SB_ALLOCATED;
  512. } else {
  513. /* cannot increase the size of this subband */
  514. subband_status[max_ch][max_sb] = SB_NOMORE;
  515. }
  516. }
  517. *padding = max_frame_size - current_frame_size;
  518. assert(*padding >= 0);
  519. #if 0
  520. for(i=0;i<s->sblimit;i++) {
  521. printf("%d ", bit_alloc[i]);
  522. }
  523. printf("\n");
  524. #endif
  525. }
  526. /*
  527. * Output the mpeg audio layer 2 frame. Note how the code is small
  528. * compared to other encoders :-)
  529. */
  530. static void encode_frame(MpegAudioContext *s,
  531. unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
  532. int padding)
  533. {
  534. int i, j, k, l, bit_alloc_bits, b, ch;
  535. unsigned char *sf;
  536. int q[3];
  537. PutBitContext *p = &s->pb;
  538. /* header */
  539. put_bits(p, 12, 0xfff);
  540. put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
  541. put_bits(p, 2, 4-2); /* layer 2 */
  542. put_bits(p, 1, 1); /* no error protection */
  543. put_bits(p, 4, s->bitrate_index);
  544. put_bits(p, 2, s->freq_index);
  545. put_bits(p, 1, s->do_padding); /* use padding */
  546. put_bits(p, 1, 0); /* private_bit */
  547. put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
  548. put_bits(p, 2, 0); /* mode_ext */
  549. put_bits(p, 1, 0); /* no copyright */
  550. put_bits(p, 1, 1); /* original */
  551. put_bits(p, 2, 0); /* no emphasis */
  552. /* bit allocation */
  553. j = 0;
  554. for(i=0;i<s->sblimit;i++) {
  555. bit_alloc_bits = s->alloc_table[j];
  556. for(ch=0;ch<s->nb_channels;ch++) {
  557. put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
  558. }
  559. j += 1 << bit_alloc_bits;
  560. }
  561. /* scale codes */
  562. for(i=0;i<s->sblimit;i++) {
  563. for(ch=0;ch<s->nb_channels;ch++) {
  564. if (bit_alloc[ch][i])
  565. put_bits(p, 2, s->scale_code[ch][i]);
  566. }
  567. }
  568. /* scale factors */
  569. for(i=0;i<s->sblimit;i++) {
  570. for(ch=0;ch<s->nb_channels;ch++) {
  571. if (bit_alloc[ch][i]) {
  572. sf = &s->scale_factors[ch][i][0];
  573. switch(s->scale_code[ch][i]) {
  574. case 0:
  575. put_bits(p, 6, sf[0]);
  576. put_bits(p, 6, sf[1]);
  577. put_bits(p, 6, sf[2]);
  578. break;
  579. case 3:
  580. case 1:
  581. put_bits(p, 6, sf[0]);
  582. put_bits(p, 6, sf[2]);
  583. break;
  584. case 2:
  585. put_bits(p, 6, sf[0]);
  586. break;
  587. }
  588. }
  589. }
  590. }
  591. /* quantization & write sub band samples */
  592. for(k=0;k<3;k++) {
  593. for(l=0;l<12;l+=3) {
  594. j = 0;
  595. for(i=0;i<s->sblimit;i++) {
  596. bit_alloc_bits = s->alloc_table[j];
  597. for(ch=0;ch<s->nb_channels;ch++) {
  598. b = bit_alloc[ch][i];
  599. if (b) {
  600. int qindex, steps, m, sample, bits;
  601. /* we encode 3 sub band samples of the same sub band at a time */
  602. qindex = s->alloc_table[j+b];
  603. steps = quant_steps[qindex];
  604. for(m=0;m<3;m++) {
  605. sample = s->sb_samples[ch][k][l + m][i];
  606. /* divide by scale factor */
  607. #ifdef USE_FLOATS
  608. {
  609. float a;
  610. a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
  611. q[m] = (int)((a + 1.0) * steps * 0.5);
  612. }
  613. #else
  614. {
  615. int q1, e, shift, mult;
  616. e = s->scale_factors[ch][i][k];
  617. shift = scale_factor_shift[e];
  618. mult = scale_factor_mult[e];
  619. /* normalize to P bits */
  620. if (shift < 0)
  621. q1 = sample << (-shift);
  622. else
  623. q1 = sample >> shift;
  624. q1 = (q1 * mult) >> P;
  625. q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
  626. }
  627. #endif
  628. if (q[m] >= steps)
  629. q[m] = steps - 1;
  630. assert(q[m] >= 0 && q[m] < steps);
  631. }
  632. bits = quant_bits[qindex];
  633. if (bits < 0) {
  634. /* group the 3 values to save bits */
  635. put_bits(p, -bits,
  636. q[0] + steps * (q[1] + steps * q[2]));
  637. #if 0
  638. printf("%d: gr1 %d\n",
  639. i, q[0] + steps * (q[1] + steps * q[2]));
  640. #endif
  641. } else {
  642. #if 0
  643. printf("%d: gr3 %d %d %d\n",
  644. i, q[0], q[1], q[2]);
  645. #endif
  646. put_bits(p, bits, q[0]);
  647. put_bits(p, bits, q[1]);
  648. put_bits(p, bits, q[2]);
  649. }
  650. }
  651. }
  652. /* next subband in alloc table */
  653. j += 1 << bit_alloc_bits;
  654. }
  655. }
  656. }
  657. /* padding */
  658. for(i=0;i<padding;i++)
  659. put_bits(p, 1, 0);
  660. /* flush */
  661. flush_put_bits(p);
  662. }
  663. static int MPA_encode_frame(AVCodecContext *avctx,
  664. unsigned char *frame, int buf_size, void *data)
  665. {
  666. MpegAudioContext *s = avctx->priv_data;
  667. short *samples = data;
  668. short smr[MPA_MAX_CHANNELS][SBLIMIT];
  669. unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
  670. int padding, i;
  671. for(i=0;i<s->nb_channels;i++) {
  672. filter(s, i, samples + i, s->nb_channels);
  673. }
  674. for(i=0;i<s->nb_channels;i++) {
  675. compute_scale_factors(s->scale_code[i], s->scale_factors[i],
  676. s->sb_samples[i], s->sblimit);
  677. }
  678. for(i=0;i<s->nb_channels;i++) {
  679. psycho_acoustic_model(s, smr[i]);
  680. }
  681. compute_bit_allocation(s, smr, bit_alloc, &padding);
  682. init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE, NULL, NULL);
  683. encode_frame(s, bit_alloc, padding);
  684. s->nb_samples += MPA_FRAME_SIZE;
  685. return pbBufPtr(&s->pb) - s->pb.buf;
  686. }
  687. static int MPA_encode_close(AVCodecContext *avctx)
  688. {
  689. av_freep(&avctx->coded_frame);
  690. return 0;
  691. }
  692. AVCodec mp2_encoder = {
  693. "mp2",
  694. CODEC_TYPE_AUDIO,
  695. CODEC_ID_MP2,
  696. sizeof(MpegAudioContext),
  697. MPA_encode_init,
  698. MPA_encode_frame,
  699. MPA_encode_close,
  700. NULL,
  701. };
  702. #undef FIX