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  1. /*
  2. * MPEG Audio decoder
  3. * Copyright (c) 2001, 2002 Fabrice Bellard
  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. */
  21. /**
  22. * @file
  23. * MPEG Audio decoder.
  24. */
  25. #include "avcodec.h"
  26. #include "get_bits.h"
  27. #include "dsputil.h"
  28. /*
  29. * TODO:
  30. * - in low precision mode, use more 16 bit multiplies in synth filter
  31. * - test lsf / mpeg25 extensively.
  32. */
  33. #include "mpegaudio.h"
  34. #include "mpegaudiodecheader.h"
  35. #include "mathops.h"
  36. #if CONFIG_FLOAT
  37. # define SHR(a,b) ((a)*(1.0f/(1<<(b))))
  38. # define compute_antialias compute_antialias_float
  39. # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
  40. # define FIXR(x) ((float)(x))
  41. # define FIXHR(x) ((float)(x))
  42. # define MULH3(x, y, s) ((s)*(y)*(x))
  43. # define MULLx(x, y, s) ((y)*(x))
  44. # define RENAME(a) a ## _float
  45. #else
  46. # define SHR(a,b) ((a)>>(b))
  47. # define compute_antialias compute_antialias_integer
  48. /* WARNING: only correct for posititive numbers */
  49. # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
  50. # define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
  51. # define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
  52. # define MULH3(x, y, s) MULH((s)*(x), y)
  53. # define MULLx(x, y, s) MULL(x,y,s)
  54. # define RENAME(a) a
  55. #endif
  56. /****************/
  57. #define HEADER_SIZE 4
  58. #include "mpegaudiodata.h"
  59. #include "mpegaudiodectab.h"
  60. #if CONFIG_FLOAT
  61. # include "fft.h"
  62. #else
  63. # include "dct32.c"
  64. #endif
  65. static void compute_antialias(MPADecodeContext *s, GranuleDef *g);
  66. static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
  67. int *dither_state, OUT_INT *samples, int incr);
  68. /* vlc structure for decoding layer 3 huffman tables */
  69. static VLC huff_vlc[16];
  70. static VLC_TYPE huff_vlc_tables[
  71. 0+128+128+128+130+128+154+166+
  72. 142+204+190+170+542+460+662+414
  73. ][2];
  74. static const int huff_vlc_tables_sizes[16] = {
  75. 0, 128, 128, 128, 130, 128, 154, 166,
  76. 142, 204, 190, 170, 542, 460, 662, 414
  77. };
  78. static VLC huff_quad_vlc[2];
  79. static VLC_TYPE huff_quad_vlc_tables[128+16][2];
  80. static const int huff_quad_vlc_tables_sizes[2] = {
  81. 128, 16
  82. };
  83. /* computed from band_size_long */
  84. static uint16_t band_index_long[9][23];
  85. #include "mpegaudio_tablegen.h"
  86. /* intensity stereo coef table */
  87. static INTFLOAT is_table[2][16];
  88. static INTFLOAT is_table_lsf[2][2][16];
  89. static int32_t csa_table[8][4];
  90. static float csa_table_float[8][4];
  91. static INTFLOAT mdct_win[8][36];
  92. static int16_t division_tab3[1<<6 ];
  93. static int16_t division_tab5[1<<8 ];
  94. static int16_t division_tab9[1<<11];
  95. static int16_t * const division_tabs[4] = {
  96. division_tab3, division_tab5, NULL, division_tab9
  97. };
  98. /* lower 2 bits: modulo 3, higher bits: shift */
  99. static uint16_t scale_factor_modshift[64];
  100. /* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
  101. static int32_t scale_factor_mult[15][3];
  102. /* mult table for layer 2 group quantization */
  103. #define SCALE_GEN(v) \
  104. { FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
  105. static const int32_t scale_factor_mult2[3][3] = {
  106. SCALE_GEN(4.0 / 3.0), /* 3 steps */
  107. SCALE_GEN(4.0 / 5.0), /* 5 steps */
  108. SCALE_GEN(4.0 / 9.0), /* 9 steps */
  109. };
  110. DECLARE_ALIGNED(16, MPA_INT, RENAME(ff_mpa_synth_window))[512+256];
  111. /**
  112. * Convert region offsets to region sizes and truncate
  113. * size to big_values.
  114. */
  115. static void ff_region_offset2size(GranuleDef *g){
  116. int i, k, j=0;
  117. g->region_size[2] = (576 / 2);
  118. for(i=0;i<3;i++) {
  119. k = FFMIN(g->region_size[i], g->big_values);
  120. g->region_size[i] = k - j;
  121. j = k;
  122. }
  123. }
  124. static void ff_init_short_region(MPADecodeContext *s, GranuleDef *g){
  125. if (g->block_type == 2)
  126. g->region_size[0] = (36 / 2);
  127. else {
  128. if (s->sample_rate_index <= 2)
  129. g->region_size[0] = (36 / 2);
  130. else if (s->sample_rate_index != 8)
  131. g->region_size[0] = (54 / 2);
  132. else
  133. g->region_size[0] = (108 / 2);
  134. }
  135. g->region_size[1] = (576 / 2);
  136. }
  137. static void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2){
  138. int l;
  139. g->region_size[0] =
  140. band_index_long[s->sample_rate_index][ra1 + 1] >> 1;
  141. /* should not overflow */
  142. l = FFMIN(ra1 + ra2 + 2, 22);
  143. g->region_size[1] =
  144. band_index_long[s->sample_rate_index][l] >> 1;
  145. }
  146. static void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g){
  147. if (g->block_type == 2) {
  148. if (g->switch_point) {
  149. /* if switched mode, we handle the 36 first samples as
  150. long blocks. For 8000Hz, we handle the 48 first
  151. exponents as long blocks (XXX: check this!) */
  152. if (s->sample_rate_index <= 2)
  153. g->long_end = 8;
  154. else if (s->sample_rate_index != 8)
  155. g->long_end = 6;
  156. else
  157. g->long_end = 4; /* 8000 Hz */
  158. g->short_start = 2 + (s->sample_rate_index != 8);
  159. } else {
  160. g->long_end = 0;
  161. g->short_start = 0;
  162. }
  163. } else {
  164. g->short_start = 13;
  165. g->long_end = 22;
  166. }
  167. }
  168. /* layer 1 unscaling */
  169. /* n = number of bits of the mantissa minus 1 */
  170. static inline int l1_unscale(int n, int mant, int scale_factor)
  171. {
  172. int shift, mod;
  173. int64_t val;
  174. shift = scale_factor_modshift[scale_factor];
  175. mod = shift & 3;
  176. shift >>= 2;
  177. val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]);
  178. shift += n;
  179. /* NOTE: at this point, 1 <= shift >= 21 + 15 */
  180. return (int)((val + (1LL << (shift - 1))) >> shift);
  181. }
  182. static inline int l2_unscale_group(int steps, int mant, int scale_factor)
  183. {
  184. int shift, mod, val;
  185. shift = scale_factor_modshift[scale_factor];
  186. mod = shift & 3;
  187. shift >>= 2;
  188. val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
  189. /* NOTE: at this point, 0 <= shift <= 21 */
  190. if (shift > 0)
  191. val = (val + (1 << (shift - 1))) >> shift;
  192. return val;
  193. }
  194. /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
  195. static inline int l3_unscale(int value, int exponent)
  196. {
  197. unsigned int m;
  198. int e;
  199. e = table_4_3_exp [4*value + (exponent&3)];
  200. m = table_4_3_value[4*value + (exponent&3)];
  201. e -= (exponent >> 2);
  202. assert(e>=1);
  203. if (e > 31)
  204. return 0;
  205. m = (m + (1 << (e-1))) >> e;
  206. return m;
  207. }
  208. /* all integer n^(4/3) computation code */
  209. #define DEV_ORDER 13
  210. #define POW_FRAC_BITS 24
  211. #define POW_FRAC_ONE (1 << POW_FRAC_BITS)
  212. #define POW_FIX(a) ((int)((a) * POW_FRAC_ONE))
  213. #define POW_MULL(a,b) (((int64_t)(a) * (int64_t)(b)) >> POW_FRAC_BITS)
  214. static int dev_4_3_coefs[DEV_ORDER];
  215. #if 0 /* unused */
  216. static int pow_mult3[3] = {
  217. POW_FIX(1.0),
  218. POW_FIX(1.25992104989487316476),
  219. POW_FIX(1.58740105196819947474),
  220. };
  221. #endif
  222. static av_cold void int_pow_init(void)
  223. {
  224. int i, a;
  225. a = POW_FIX(1.0);
  226. for(i=0;i<DEV_ORDER;i++) {
  227. a = POW_MULL(a, POW_FIX(4.0 / 3.0) - i * POW_FIX(1.0)) / (i + 1);
  228. dev_4_3_coefs[i] = a;
  229. }
  230. }
  231. #if 0 /* unused, remove? */
  232. /* return the mantissa and the binary exponent */
  233. static int int_pow(int i, int *exp_ptr)
  234. {
  235. int e, er, eq, j;
  236. int a, a1;
  237. /* renormalize */
  238. a = i;
  239. e = POW_FRAC_BITS;
  240. while (a < (1 << (POW_FRAC_BITS - 1))) {
  241. a = a << 1;
  242. e--;
  243. }
  244. a -= (1 << POW_FRAC_BITS);
  245. a1 = 0;
  246. for(j = DEV_ORDER - 1; j >= 0; j--)
  247. a1 = POW_MULL(a, dev_4_3_coefs[j] + a1);
  248. a = (1 << POW_FRAC_BITS) + a1;
  249. /* exponent compute (exact) */
  250. e = e * 4;
  251. er = e % 3;
  252. eq = e / 3;
  253. a = POW_MULL(a, pow_mult3[er]);
  254. while (a >= 2 * POW_FRAC_ONE) {
  255. a = a >> 1;
  256. eq++;
  257. }
  258. /* convert to float */
  259. while (a < POW_FRAC_ONE) {
  260. a = a << 1;
  261. eq--;
  262. }
  263. /* now POW_FRAC_ONE <= a < 2 * POW_FRAC_ONE */
  264. #if POW_FRAC_BITS > FRAC_BITS
  265. a = (a + (1 << (POW_FRAC_BITS - FRAC_BITS - 1))) >> (POW_FRAC_BITS - FRAC_BITS);
  266. /* correct overflow */
  267. if (a >= 2 * (1 << FRAC_BITS)) {
  268. a = a >> 1;
  269. eq++;
  270. }
  271. #endif
  272. *exp_ptr = eq;
  273. return a;
  274. }
  275. #endif
  276. static av_cold int decode_init(AVCodecContext * avctx)
  277. {
  278. MPADecodeContext *s = avctx->priv_data;
  279. static int init=0;
  280. int i, j, k;
  281. s->avctx = avctx;
  282. s->apply_window_mp3 = apply_window_mp3_c;
  283. #if HAVE_MMX && CONFIG_FLOAT
  284. ff_mpegaudiodec_init_mmx(s);
  285. #endif
  286. if (HAVE_ALTIVEC && CONFIG_FLOAT) ff_mpegaudiodec_init_altivec(s);
  287. avctx->sample_fmt= OUT_FMT;
  288. s->error_recognition= avctx->error_recognition;
  289. if (!init && !avctx->parse_only) {
  290. int offset;
  291. /* scale factors table for layer 1/2 */
  292. for(i=0;i<64;i++) {
  293. int shift, mod;
  294. /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
  295. shift = (i / 3);
  296. mod = i % 3;
  297. scale_factor_modshift[i] = mod | (shift << 2);
  298. }
  299. /* scale factor multiply for layer 1 */
  300. for(i=0;i<15;i++) {
  301. int n, norm;
  302. n = i + 2;
  303. norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
  304. scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
  305. scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
  306. scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
  307. dprintf(avctx, "%d: norm=%x s=%x %x %x\n",
  308. i, norm,
  309. scale_factor_mult[i][0],
  310. scale_factor_mult[i][1],
  311. scale_factor_mult[i][2]);
  312. }
  313. #if CONFIG_FLOAT
  314. ff_dct_init(&s->dct, 5, DCT_II);
  315. #endif
  316. RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));
  317. /* huffman decode tables */
  318. offset = 0;
  319. for(i=1;i<16;i++) {
  320. const HuffTable *h = &mpa_huff_tables[i];
  321. int xsize, x, y;
  322. uint8_t tmp_bits [512];
  323. uint16_t tmp_codes[512];
  324. memset(tmp_bits , 0, sizeof(tmp_bits ));
  325. memset(tmp_codes, 0, sizeof(tmp_codes));
  326. xsize = h->xsize;
  327. j = 0;
  328. for(x=0;x<xsize;x++) {
  329. for(y=0;y<xsize;y++){
  330. tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ];
  331. tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];
  332. }
  333. }
  334. /* XXX: fail test */
  335. huff_vlc[i].table = huff_vlc_tables+offset;
  336. huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
  337. init_vlc(&huff_vlc[i], 7, 512,
  338. tmp_bits, 1, 1, tmp_codes, 2, 2,
  339. INIT_VLC_USE_NEW_STATIC);
  340. offset += huff_vlc_tables_sizes[i];
  341. }
  342. assert(offset == FF_ARRAY_ELEMS(huff_vlc_tables));
  343. offset = 0;
  344. for(i=0;i<2;i++) {
  345. huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
  346. huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
  347. init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
  348. mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
  349. INIT_VLC_USE_NEW_STATIC);
  350. offset += huff_quad_vlc_tables_sizes[i];
  351. }
  352. assert(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));
  353. for(i=0;i<9;i++) {
  354. k = 0;
  355. for(j=0;j<22;j++) {
  356. band_index_long[i][j] = k;
  357. k += band_size_long[i][j];
  358. }
  359. band_index_long[i][22] = k;
  360. }
  361. /* compute n ^ (4/3) and store it in mantissa/exp format */
  362. int_pow_init();
  363. mpegaudio_tableinit();
  364. for (i = 0; i < 4; i++)
  365. if (ff_mpa_quant_bits[i] < 0)
  366. for (j = 0; j < (1<<(-ff_mpa_quant_bits[i]+1)); j++) {
  367. int val1, val2, val3, steps;
  368. int val = j;
  369. steps = ff_mpa_quant_steps[i];
  370. val1 = val % steps;
  371. val /= steps;
  372. val2 = val % steps;
  373. val3 = val / steps;
  374. division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);
  375. }
  376. for(i=0;i<7;i++) {
  377. float f;
  378. INTFLOAT v;
  379. if (i != 6) {
  380. f = tan((double)i * M_PI / 12.0);
  381. v = FIXR(f / (1.0 + f));
  382. } else {
  383. v = FIXR(1.0);
  384. }
  385. is_table[0][i] = v;
  386. is_table[1][6 - i] = v;
  387. }
  388. /* invalid values */
  389. for(i=7;i<16;i++)
  390. is_table[0][i] = is_table[1][i] = 0.0;
  391. for(i=0;i<16;i++) {
  392. double f;
  393. int e, k;
  394. for(j=0;j<2;j++) {
  395. e = -(j + 1) * ((i + 1) >> 1);
  396. f = pow(2.0, e / 4.0);
  397. k = i & 1;
  398. is_table_lsf[j][k ^ 1][i] = FIXR(f);
  399. is_table_lsf[j][k][i] = FIXR(1.0);
  400. dprintf(avctx, "is_table_lsf %d %d: %x %x\n",
  401. i, j, is_table_lsf[j][0][i], is_table_lsf[j][1][i]);
  402. }
  403. }
  404. for(i=0;i<8;i++) {
  405. float ci, cs, ca;
  406. ci = ci_table[i];
  407. cs = 1.0 / sqrt(1.0 + ci * ci);
  408. ca = cs * ci;
  409. csa_table[i][0] = FIXHR(cs/4);
  410. csa_table[i][1] = FIXHR(ca/4);
  411. csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);
  412. csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);
  413. csa_table_float[i][0] = cs;
  414. csa_table_float[i][1] = ca;
  415. csa_table_float[i][2] = ca + cs;
  416. csa_table_float[i][3] = ca - cs;
  417. }
  418. /* compute mdct windows */
  419. for(i=0;i<36;i++) {
  420. for(j=0; j<4; j++){
  421. double d;
  422. if(j==2 && i%3 != 1)
  423. continue;
  424. d= sin(M_PI * (i + 0.5) / 36.0);
  425. if(j==1){
  426. if (i>=30) d= 0;
  427. else if(i>=24) d= sin(M_PI * (i - 18 + 0.5) / 12.0);
  428. else if(i>=18) d= 1;
  429. }else if(j==3){
  430. if (i< 6) d= 0;
  431. else if(i< 12) d= sin(M_PI * (i - 6 + 0.5) / 12.0);
  432. else if(i< 18) d= 1;
  433. }
  434. //merge last stage of imdct into the window coefficients
  435. d*= 0.5 / cos(M_PI*(2*i + 19)/72);
  436. if(j==2)
  437. mdct_win[j][i/3] = FIXHR((d / (1<<5)));
  438. else
  439. mdct_win[j][i ] = FIXHR((d / (1<<5)));
  440. }
  441. }
  442. /* NOTE: we do frequency inversion adter the MDCT by changing
  443. the sign of the right window coefs */
  444. for(j=0;j<4;j++) {
  445. for(i=0;i<36;i+=2) {
  446. mdct_win[j + 4][i] = mdct_win[j][i];
  447. mdct_win[j + 4][i + 1] = -mdct_win[j][i + 1];
  448. }
  449. }
  450. init = 1;
  451. }
  452. if (avctx->codec_id == CODEC_ID_MP3ADU)
  453. s->adu_mode = 1;
  454. return 0;
  455. }
  456. #if CONFIG_FLOAT
  457. static inline float round_sample(float *sum)
  458. {
  459. float sum1=*sum;
  460. *sum = 0;
  461. return sum1;
  462. }
  463. /* signed 16x16 -> 32 multiply add accumulate */
  464. #define MACS(rt, ra, rb) rt+=(ra)*(rb)
  465. /* signed 16x16 -> 32 multiply */
  466. #define MULS(ra, rb) ((ra)*(rb))
  467. #define MLSS(rt, ra, rb) rt-=(ra)*(rb)
  468. #elif FRAC_BITS <= 15
  469. static inline int round_sample(int *sum)
  470. {
  471. int sum1;
  472. sum1 = (*sum) >> OUT_SHIFT;
  473. *sum &= (1<<OUT_SHIFT)-1;
  474. return av_clip(sum1, OUT_MIN, OUT_MAX);
  475. }
  476. /* signed 16x16 -> 32 multiply add accumulate */
  477. #define MACS(rt, ra, rb) MAC16(rt, ra, rb)
  478. /* signed 16x16 -> 32 multiply */
  479. #define MULS(ra, rb) MUL16(ra, rb)
  480. #define MLSS(rt, ra, rb) MLS16(rt, ra, rb)
  481. #else
  482. static inline int round_sample(int64_t *sum)
  483. {
  484. int sum1;
  485. sum1 = (int)((*sum) >> OUT_SHIFT);
  486. *sum &= (1<<OUT_SHIFT)-1;
  487. return av_clip(sum1, OUT_MIN, OUT_MAX);
  488. }
  489. # define MULS(ra, rb) MUL64(ra, rb)
  490. # define MACS(rt, ra, rb) MAC64(rt, ra, rb)
  491. # define MLSS(rt, ra, rb) MLS64(rt, ra, rb)
  492. #endif
  493. #define SUM8(op, sum, w, p) \
  494. { \
  495. op(sum, (w)[0 * 64], (p)[0 * 64]); \
  496. op(sum, (w)[1 * 64], (p)[1 * 64]); \
  497. op(sum, (w)[2 * 64], (p)[2 * 64]); \
  498. op(sum, (w)[3 * 64], (p)[3 * 64]); \
  499. op(sum, (w)[4 * 64], (p)[4 * 64]); \
  500. op(sum, (w)[5 * 64], (p)[5 * 64]); \
  501. op(sum, (w)[6 * 64], (p)[6 * 64]); \
  502. op(sum, (w)[7 * 64], (p)[7 * 64]); \
  503. }
  504. #define SUM8P2(sum1, op1, sum2, op2, w1, w2, p) \
  505. { \
  506. INTFLOAT tmp;\
  507. tmp = p[0 * 64];\
  508. op1(sum1, (w1)[0 * 64], tmp);\
  509. op2(sum2, (w2)[0 * 64], tmp);\
  510. tmp = p[1 * 64];\
  511. op1(sum1, (w1)[1 * 64], tmp);\
  512. op2(sum2, (w2)[1 * 64], tmp);\
  513. tmp = p[2 * 64];\
  514. op1(sum1, (w1)[2 * 64], tmp);\
  515. op2(sum2, (w2)[2 * 64], tmp);\
  516. tmp = p[3 * 64];\
  517. op1(sum1, (w1)[3 * 64], tmp);\
  518. op2(sum2, (w2)[3 * 64], tmp);\
  519. tmp = p[4 * 64];\
  520. op1(sum1, (w1)[4 * 64], tmp);\
  521. op2(sum2, (w2)[4 * 64], tmp);\
  522. tmp = p[5 * 64];\
  523. op1(sum1, (w1)[5 * 64], tmp);\
  524. op2(sum2, (w2)[5 * 64], tmp);\
  525. tmp = p[6 * 64];\
  526. op1(sum1, (w1)[6 * 64], tmp);\
  527. op2(sum2, (w2)[6 * 64], tmp);\
  528. tmp = p[7 * 64];\
  529. op1(sum1, (w1)[7 * 64], tmp);\
  530. op2(sum2, (w2)[7 * 64], tmp);\
  531. }
  532. void av_cold RENAME(ff_mpa_synth_init)(MPA_INT *window)
  533. {
  534. int i, j;
  535. /* max = 18760, max sum over all 16 coefs : 44736 */
  536. for(i=0;i<257;i++) {
  537. INTFLOAT v;
  538. v = ff_mpa_enwindow[i];
  539. #if CONFIG_FLOAT
  540. v *= 1.0 / (1LL<<(16 + FRAC_BITS));
  541. #elif WFRAC_BITS < 16
  542. v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
  543. #endif
  544. window[i] = v;
  545. if ((i & 63) != 0)
  546. v = -v;
  547. if (i != 0)
  548. window[512 - i] = v;
  549. }
  550. // Needed for avoiding shuffles in ASM implementations
  551. for(i=0; i < 8; i++)
  552. for(j=0; j < 16; j++)
  553. window[512+16*i+j] = window[64*i+32-j];
  554. for(i=0; i < 8; i++)
  555. for(j=0; j < 16; j++)
  556. window[512+128+16*i+j] = window[64*i+48-j];
  557. }
  558. static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
  559. int *dither_state, OUT_INT *samples, int incr)
  560. {
  561. register const MPA_INT *w, *w2, *p;
  562. int j;
  563. OUT_INT *samples2;
  564. #if CONFIG_FLOAT
  565. float sum, sum2;
  566. #elif FRAC_BITS <= 15
  567. int sum, sum2;
  568. #else
  569. int64_t sum, sum2;
  570. #endif
  571. /* copy to avoid wrap */
  572. memcpy(synth_buf + 512, synth_buf, 32 * sizeof(*synth_buf));
  573. samples2 = samples + 31 * incr;
  574. w = window;
  575. w2 = window + 31;
  576. sum = *dither_state;
  577. p = synth_buf + 16;
  578. SUM8(MACS, sum, w, p);
  579. p = synth_buf + 48;
  580. SUM8(MLSS, sum, w + 32, p);
  581. *samples = round_sample(&sum);
  582. samples += incr;
  583. w++;
  584. /* we calculate two samples at the same time to avoid one memory
  585. access per two sample */
  586. for(j=1;j<16;j++) {
  587. sum2 = 0;
  588. p = synth_buf + 16 + j;
  589. SUM8P2(sum, MACS, sum2, MLSS, w, w2, p);
  590. p = synth_buf + 48 - j;
  591. SUM8P2(sum, MLSS, sum2, MLSS, w + 32, w2 + 32, p);
  592. *samples = round_sample(&sum);
  593. samples += incr;
  594. sum += sum2;
  595. *samples2 = round_sample(&sum);
  596. samples2 -= incr;
  597. w++;
  598. w2--;
  599. }
  600. p = synth_buf + 32;
  601. SUM8(MLSS, sum, w + 32, p);
  602. *samples = round_sample(&sum);
  603. *dither_state= sum;
  604. }
  605. /* 32 sub band synthesis filter. Input: 32 sub band samples, Output:
  606. 32 samples. */
  607. /* XXX: optimize by avoiding ring buffer usage */
  608. #if !CONFIG_FLOAT
  609. void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
  610. MPA_INT *window, int *dither_state,
  611. OUT_INT *samples, int incr,
  612. INTFLOAT sb_samples[SBLIMIT])
  613. {
  614. register MPA_INT *synth_buf;
  615. int offset;
  616. #if FRAC_BITS <= 15
  617. int32_t tmp[32];
  618. int j;
  619. #endif
  620. offset = *synth_buf_offset;
  621. synth_buf = synth_buf_ptr + offset;
  622. #if FRAC_BITS <= 15
  623. dct32(tmp, sb_samples);
  624. for(j=0;j<32;j++) {
  625. /* NOTE: can cause a loss in precision if very high amplitude
  626. sound */
  627. synth_buf[j] = av_clip_int16(tmp[j]);
  628. }
  629. #else
  630. dct32(synth_buf, sb_samples);
  631. #endif
  632. apply_window_mp3_c(synth_buf, window, dither_state, samples, incr);
  633. offset = (offset - 32) & 511;
  634. *synth_buf_offset = offset;
  635. }
  636. #endif
  637. #define C3 FIXHR(0.86602540378443864676/2)
  638. /* 0.5 / cos(pi*(2*i+1)/36) */
  639. static const INTFLOAT icos36[9] = {
  640. FIXR(0.50190991877167369479),
  641. FIXR(0.51763809020504152469), //0
  642. FIXR(0.55168895948124587824),
  643. FIXR(0.61038729438072803416),
  644. FIXR(0.70710678118654752439), //1
  645. FIXR(0.87172339781054900991),
  646. FIXR(1.18310079157624925896),
  647. FIXR(1.93185165257813657349), //2
  648. FIXR(5.73685662283492756461),
  649. };
  650. /* 0.5 / cos(pi*(2*i+1)/36) */
  651. static const INTFLOAT icos36h[9] = {
  652. FIXHR(0.50190991877167369479/2),
  653. FIXHR(0.51763809020504152469/2), //0
  654. FIXHR(0.55168895948124587824/2),
  655. FIXHR(0.61038729438072803416/2),
  656. FIXHR(0.70710678118654752439/2), //1
  657. FIXHR(0.87172339781054900991/2),
  658. FIXHR(1.18310079157624925896/4),
  659. FIXHR(1.93185165257813657349/4), //2
  660. // FIXHR(5.73685662283492756461),
  661. };
  662. /* 12 points IMDCT. We compute it "by hand" by factorizing obvious
  663. cases. */
  664. static void imdct12(INTFLOAT *out, INTFLOAT *in)
  665. {
  666. INTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
  667. in0= in[0*3];
  668. in1= in[1*3] + in[0*3];
  669. in2= in[2*3] + in[1*3];
  670. in3= in[3*3] + in[2*3];
  671. in4= in[4*3] + in[3*3];
  672. in5= in[5*3] + in[4*3];
  673. in5 += in3;
  674. in3 += in1;
  675. in2= MULH3(in2, C3, 2);
  676. in3= MULH3(in3, C3, 4);
  677. t1 = in0 - in4;
  678. t2 = MULH3(in1 - in5, icos36h[4], 2);
  679. out[ 7]=
  680. out[10]= t1 + t2;
  681. out[ 1]=
  682. out[ 4]= t1 - t2;
  683. in0 += SHR(in4, 1);
  684. in4 = in0 + in2;
  685. in5 += 2*in1;
  686. in1 = MULH3(in5 + in3, icos36h[1], 1);
  687. out[ 8]=
  688. out[ 9]= in4 + in1;
  689. out[ 2]=
  690. out[ 3]= in4 - in1;
  691. in0 -= in2;
  692. in5 = MULH3(in5 - in3, icos36h[7], 2);
  693. out[ 0]=
  694. out[ 5]= in0 - in5;
  695. out[ 6]=
  696. out[11]= in0 + in5;
  697. }
  698. /* cos(pi*i/18) */
  699. #define C1 FIXHR(0.98480775301220805936/2)
  700. #define C2 FIXHR(0.93969262078590838405/2)
  701. #define C3 FIXHR(0.86602540378443864676/2)
  702. #define C4 FIXHR(0.76604444311897803520/2)
  703. #define C5 FIXHR(0.64278760968653932632/2)
  704. #define C6 FIXHR(0.5/2)
  705. #define C7 FIXHR(0.34202014332566873304/2)
  706. #define C8 FIXHR(0.17364817766693034885/2)
  707. /* using Lee like decomposition followed by hand coded 9 points DCT */
  708. static void imdct36(INTFLOAT *out, INTFLOAT *buf, INTFLOAT *in, INTFLOAT *win)
  709. {
  710. int i, j;
  711. INTFLOAT t0, t1, t2, t3, s0, s1, s2, s3;
  712. INTFLOAT tmp[18], *tmp1, *in1;
  713. for(i=17;i>=1;i--)
  714. in[i] += in[i-1];
  715. for(i=17;i>=3;i-=2)
  716. in[i] += in[i-2];
  717. for(j=0;j<2;j++) {
  718. tmp1 = tmp + j;
  719. in1 = in + j;
  720. t2 = in1[2*4] + in1[2*8] - in1[2*2];
  721. t3 = in1[2*0] + SHR(in1[2*6],1);
  722. t1 = in1[2*0] - in1[2*6];
  723. tmp1[ 6] = t1 - SHR(t2,1);
  724. tmp1[16] = t1 + t2;
  725. t0 = MULH3(in1[2*2] + in1[2*4] , C2, 2);
  726. t1 = MULH3(in1[2*4] - in1[2*8] , -2*C8, 1);
  727. t2 = MULH3(in1[2*2] + in1[2*8] , -C4, 2);
  728. tmp1[10] = t3 - t0 - t2;
  729. tmp1[ 2] = t3 + t0 + t1;
  730. tmp1[14] = t3 + t2 - t1;
  731. tmp1[ 4] = MULH3(in1[2*5] + in1[2*7] - in1[2*1], -C3, 2);
  732. t2 = MULH3(in1[2*1] + in1[2*5], C1, 2);
  733. t3 = MULH3(in1[2*5] - in1[2*7], -2*C7, 1);
  734. t0 = MULH3(in1[2*3], C3, 2);
  735. t1 = MULH3(in1[2*1] + in1[2*7], -C5, 2);
  736. tmp1[ 0] = t2 + t3 + t0;
  737. tmp1[12] = t2 + t1 - t0;
  738. tmp1[ 8] = t3 - t1 - t0;
  739. }
  740. i = 0;
  741. for(j=0;j<4;j++) {
  742. t0 = tmp[i];
  743. t1 = tmp[i + 2];
  744. s0 = t1 + t0;
  745. s2 = t1 - t0;
  746. t2 = tmp[i + 1];
  747. t3 = tmp[i + 3];
  748. s1 = MULH3(t3 + t2, icos36h[j], 2);
  749. s3 = MULLx(t3 - t2, icos36[8 - j], FRAC_BITS);
  750. t0 = s0 + s1;
  751. t1 = s0 - s1;
  752. out[(9 + j)*SBLIMIT] = MULH3(t1, win[9 + j], 1) + buf[9 + j];
  753. out[(8 - j)*SBLIMIT] = MULH3(t1, win[8 - j], 1) + buf[8 - j];
  754. buf[9 + j] = MULH3(t0, win[18 + 9 + j], 1);
  755. buf[8 - j] = MULH3(t0, win[18 + 8 - j], 1);
  756. t0 = s2 + s3;
  757. t1 = s2 - s3;
  758. out[(9 + 8 - j)*SBLIMIT] = MULH3(t1, win[9 + 8 - j], 1) + buf[9 + 8 - j];
  759. out[( j)*SBLIMIT] = MULH3(t1, win[ j], 1) + buf[ j];
  760. buf[9 + 8 - j] = MULH3(t0, win[18 + 9 + 8 - j], 1);
  761. buf[ + j] = MULH3(t0, win[18 + j], 1);
  762. i += 4;
  763. }
  764. s0 = tmp[16];
  765. s1 = MULH3(tmp[17], icos36h[4], 2);
  766. t0 = s0 + s1;
  767. t1 = s0 - s1;
  768. out[(9 + 4)*SBLIMIT] = MULH3(t1, win[9 + 4], 1) + buf[9 + 4];
  769. out[(8 - 4)*SBLIMIT] = MULH3(t1, win[8 - 4], 1) + buf[8 - 4];
  770. buf[9 + 4] = MULH3(t0, win[18 + 9 + 4], 1);
  771. buf[8 - 4] = MULH3(t0, win[18 + 8 - 4], 1);
  772. }
  773. /* return the number of decoded frames */
  774. static int mp_decode_layer1(MPADecodeContext *s)
  775. {
  776. int bound, i, v, n, ch, j, mant;
  777. uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
  778. uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
  779. if (s->mode == MPA_JSTEREO)
  780. bound = (s->mode_ext + 1) * 4;
  781. else
  782. bound = SBLIMIT;
  783. /* allocation bits */
  784. for(i=0;i<bound;i++) {
  785. for(ch=0;ch<s->nb_channels;ch++) {
  786. allocation[ch][i] = get_bits(&s->gb, 4);
  787. }
  788. }
  789. for(i=bound;i<SBLIMIT;i++) {
  790. allocation[0][i] = get_bits(&s->gb, 4);
  791. }
  792. /* scale factors */
  793. for(i=0;i<bound;i++) {
  794. for(ch=0;ch<s->nb_channels;ch++) {
  795. if (allocation[ch][i])
  796. scale_factors[ch][i] = get_bits(&s->gb, 6);
  797. }
  798. }
  799. for(i=bound;i<SBLIMIT;i++) {
  800. if (allocation[0][i]) {
  801. scale_factors[0][i] = get_bits(&s->gb, 6);
  802. scale_factors[1][i] = get_bits(&s->gb, 6);
  803. }
  804. }
  805. /* compute samples */
  806. for(j=0;j<12;j++) {
  807. for(i=0;i<bound;i++) {
  808. for(ch=0;ch<s->nb_channels;ch++) {
  809. n = allocation[ch][i];
  810. if (n) {
  811. mant = get_bits(&s->gb, n + 1);
  812. v = l1_unscale(n, mant, scale_factors[ch][i]);
  813. } else {
  814. v = 0;
  815. }
  816. s->sb_samples[ch][j][i] = v;
  817. }
  818. }
  819. for(i=bound;i<SBLIMIT;i++) {
  820. n = allocation[0][i];
  821. if (n) {
  822. mant = get_bits(&s->gb, n + 1);
  823. v = l1_unscale(n, mant, scale_factors[0][i]);
  824. s->sb_samples[0][j][i] = v;
  825. v = l1_unscale(n, mant, scale_factors[1][i]);
  826. s->sb_samples[1][j][i] = v;
  827. } else {
  828. s->sb_samples[0][j][i] = 0;
  829. s->sb_samples[1][j][i] = 0;
  830. }
  831. }
  832. }
  833. return 12;
  834. }
  835. static int mp_decode_layer2(MPADecodeContext *s)
  836. {
  837. int sblimit; /* number of used subbands */
  838. const unsigned char *alloc_table;
  839. int table, bit_alloc_bits, i, j, ch, bound, v;
  840. unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
  841. unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
  842. unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
  843. int scale, qindex, bits, steps, k, l, m, b;
  844. /* select decoding table */
  845. table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
  846. s->sample_rate, s->lsf);
  847. sblimit = ff_mpa_sblimit_table[table];
  848. alloc_table = ff_mpa_alloc_tables[table];
  849. if (s->mode == MPA_JSTEREO)
  850. bound = (s->mode_ext + 1) * 4;
  851. else
  852. bound = sblimit;
  853. dprintf(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
  854. /* sanity check */
  855. if( bound > sblimit ) bound = sblimit;
  856. /* parse bit allocation */
  857. j = 0;
  858. for(i=0;i<bound;i++) {
  859. bit_alloc_bits = alloc_table[j];
  860. for(ch=0;ch<s->nb_channels;ch++) {
  861. bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
  862. }
  863. j += 1 << bit_alloc_bits;
  864. }
  865. for(i=bound;i<sblimit;i++) {
  866. bit_alloc_bits = alloc_table[j];
  867. v = get_bits(&s->gb, bit_alloc_bits);
  868. bit_alloc[0][i] = v;
  869. bit_alloc[1][i] = v;
  870. j += 1 << bit_alloc_bits;
  871. }
  872. /* scale codes */
  873. for(i=0;i<sblimit;i++) {
  874. for(ch=0;ch<s->nb_channels;ch++) {
  875. if (bit_alloc[ch][i])
  876. scale_code[ch][i] = get_bits(&s->gb, 2);
  877. }
  878. }
  879. /* scale factors */
  880. for(i=0;i<sblimit;i++) {
  881. for(ch=0;ch<s->nb_channels;ch++) {
  882. if (bit_alloc[ch][i]) {
  883. sf = scale_factors[ch][i];
  884. switch(scale_code[ch][i]) {
  885. default:
  886. case 0:
  887. sf[0] = get_bits(&s->gb, 6);
  888. sf[1] = get_bits(&s->gb, 6);
  889. sf[2] = get_bits(&s->gb, 6);
  890. break;
  891. case 2:
  892. sf[0] = get_bits(&s->gb, 6);
  893. sf[1] = sf[0];
  894. sf[2] = sf[0];
  895. break;
  896. case 1:
  897. sf[0] = get_bits(&s->gb, 6);
  898. sf[2] = get_bits(&s->gb, 6);
  899. sf[1] = sf[0];
  900. break;
  901. case 3:
  902. sf[0] = get_bits(&s->gb, 6);
  903. sf[2] = get_bits(&s->gb, 6);
  904. sf[1] = sf[2];
  905. break;
  906. }
  907. }
  908. }
  909. }
  910. /* samples */
  911. for(k=0;k<3;k++) {
  912. for(l=0;l<12;l+=3) {
  913. j = 0;
  914. for(i=0;i<bound;i++) {
  915. bit_alloc_bits = alloc_table[j];
  916. for(ch=0;ch<s->nb_channels;ch++) {
  917. b = bit_alloc[ch][i];
  918. if (b) {
  919. scale = scale_factors[ch][i][k];
  920. qindex = alloc_table[j+b];
  921. bits = ff_mpa_quant_bits[qindex];
  922. if (bits < 0) {
  923. int v2;
  924. /* 3 values at the same time */
  925. v = get_bits(&s->gb, -bits);
  926. v2 = division_tabs[qindex][v];
  927. steps = ff_mpa_quant_steps[qindex];
  928. s->sb_samples[ch][k * 12 + l + 0][i] =
  929. l2_unscale_group(steps, v2 & 15, scale);
  930. s->sb_samples[ch][k * 12 + l + 1][i] =
  931. l2_unscale_group(steps, (v2 >> 4) & 15, scale);
  932. s->sb_samples[ch][k * 12 + l + 2][i] =
  933. l2_unscale_group(steps, v2 >> 8 , scale);
  934. } else {
  935. for(m=0;m<3;m++) {
  936. v = get_bits(&s->gb, bits);
  937. v = l1_unscale(bits - 1, v, scale);
  938. s->sb_samples[ch][k * 12 + l + m][i] = v;
  939. }
  940. }
  941. } else {
  942. s->sb_samples[ch][k * 12 + l + 0][i] = 0;
  943. s->sb_samples[ch][k * 12 + l + 1][i] = 0;
  944. s->sb_samples[ch][k * 12 + l + 2][i] = 0;
  945. }
  946. }
  947. /* next subband in alloc table */
  948. j += 1 << bit_alloc_bits;
  949. }
  950. /* XXX: find a way to avoid this duplication of code */
  951. for(i=bound;i<sblimit;i++) {
  952. bit_alloc_bits = alloc_table[j];
  953. b = bit_alloc[0][i];
  954. if (b) {
  955. int mant, scale0, scale1;
  956. scale0 = scale_factors[0][i][k];
  957. scale1 = scale_factors[1][i][k];
  958. qindex = alloc_table[j+b];
  959. bits = ff_mpa_quant_bits[qindex];
  960. if (bits < 0) {
  961. /* 3 values at the same time */
  962. v = get_bits(&s->gb, -bits);
  963. steps = ff_mpa_quant_steps[qindex];
  964. mant = v % steps;
  965. v = v / steps;
  966. s->sb_samples[0][k * 12 + l + 0][i] =
  967. l2_unscale_group(steps, mant, scale0);
  968. s->sb_samples[1][k * 12 + l + 0][i] =
  969. l2_unscale_group(steps, mant, scale1);
  970. mant = v % steps;
  971. v = v / steps;
  972. s->sb_samples[0][k * 12 + l + 1][i] =
  973. l2_unscale_group(steps, mant, scale0);
  974. s->sb_samples[1][k * 12 + l + 1][i] =
  975. l2_unscale_group(steps, mant, scale1);
  976. s->sb_samples[0][k * 12 + l + 2][i] =
  977. l2_unscale_group(steps, v, scale0);
  978. s->sb_samples[1][k * 12 + l + 2][i] =
  979. l2_unscale_group(steps, v, scale1);
  980. } else {
  981. for(m=0;m<3;m++) {
  982. mant = get_bits(&s->gb, bits);
  983. s->sb_samples[0][k * 12 + l + m][i] =
  984. l1_unscale(bits - 1, mant, scale0);
  985. s->sb_samples[1][k * 12 + l + m][i] =
  986. l1_unscale(bits - 1, mant, scale1);
  987. }
  988. }
  989. } else {
  990. s->sb_samples[0][k * 12 + l + 0][i] = 0;
  991. s->sb_samples[0][k * 12 + l + 1][i] = 0;
  992. s->sb_samples[0][k * 12 + l + 2][i] = 0;
  993. s->sb_samples[1][k * 12 + l + 0][i] = 0;
  994. s->sb_samples[1][k * 12 + l + 1][i] = 0;
  995. s->sb_samples[1][k * 12 + l + 2][i] = 0;
  996. }
  997. /* next subband in alloc table */
  998. j += 1 << bit_alloc_bits;
  999. }
  1000. /* fill remaining samples to zero */
  1001. for(i=sblimit;i<SBLIMIT;i++) {
  1002. for(ch=0;ch<s->nb_channels;ch++) {
  1003. s->sb_samples[ch][k * 12 + l + 0][i] = 0;
  1004. s->sb_samples[ch][k * 12 + l + 1][i] = 0;
  1005. s->sb_samples[ch][k * 12 + l + 2][i] = 0;
  1006. }
  1007. }
  1008. }
  1009. }
  1010. return 3 * 12;
  1011. }
  1012. #define SPLIT(dst,sf,n)\
  1013. if(n==3){\
  1014. int m= (sf*171)>>9;\
  1015. dst= sf - 3*m;\
  1016. sf=m;\
  1017. }else if(n==4){\
  1018. dst= sf&3;\
  1019. sf>>=2;\
  1020. }else if(n==5){\
  1021. int m= (sf*205)>>10;\
  1022. dst= sf - 5*m;\
  1023. sf=m;\
  1024. }else if(n==6){\
  1025. int m= (sf*171)>>10;\
  1026. dst= sf - 6*m;\
  1027. sf=m;\
  1028. }else{\
  1029. dst=0;\
  1030. }
  1031. static av_always_inline void lsf_sf_expand(int *slen,
  1032. int sf, int n1, int n2, int n3)
  1033. {
  1034. SPLIT(slen[3], sf, n3)
  1035. SPLIT(slen[2], sf, n2)
  1036. SPLIT(slen[1], sf, n1)
  1037. slen[0] = sf;
  1038. }
  1039. static void exponents_from_scale_factors(MPADecodeContext *s,
  1040. GranuleDef *g,
  1041. int16_t *exponents)
  1042. {
  1043. const uint8_t *bstab, *pretab;
  1044. int len, i, j, k, l, v0, shift, gain, gains[3];
  1045. int16_t *exp_ptr;
  1046. exp_ptr = exponents;
  1047. gain = g->global_gain - 210;
  1048. shift = g->scalefac_scale + 1;
  1049. bstab = band_size_long[s->sample_rate_index];
  1050. pretab = mpa_pretab[g->preflag];
  1051. for(i=0;i<g->long_end;i++) {
  1052. v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
  1053. len = bstab[i];
  1054. for(j=len;j>0;j--)
  1055. *exp_ptr++ = v0;
  1056. }
  1057. if (g->short_start < 13) {
  1058. bstab = band_size_short[s->sample_rate_index];
  1059. gains[0] = gain - (g->subblock_gain[0] << 3);
  1060. gains[1] = gain - (g->subblock_gain[1] << 3);
  1061. gains[2] = gain - (g->subblock_gain[2] << 3);
  1062. k = g->long_end;
  1063. for(i=g->short_start;i<13;i++) {
  1064. len = bstab[i];
  1065. for(l=0;l<3;l++) {
  1066. v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
  1067. for(j=len;j>0;j--)
  1068. *exp_ptr++ = v0;
  1069. }
  1070. }
  1071. }
  1072. }
  1073. /* handle n = 0 too */
  1074. static inline int get_bitsz(GetBitContext *s, int n)
  1075. {
  1076. if (n == 0)
  1077. return 0;
  1078. else
  1079. return get_bits(s, n);
  1080. }
  1081. static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos, int *end_pos2){
  1082. if(s->in_gb.buffer && *pos >= s->gb.size_in_bits){
  1083. s->gb= s->in_gb;
  1084. s->in_gb.buffer=NULL;
  1085. assert((get_bits_count(&s->gb) & 7) == 0);
  1086. skip_bits_long(&s->gb, *pos - *end_pos);
  1087. *end_pos2=
  1088. *end_pos= *end_pos2 + get_bits_count(&s->gb) - *pos;
  1089. *pos= get_bits_count(&s->gb);
  1090. }
  1091. }
  1092. /* Following is a optimized code for
  1093. INTFLOAT v = *src
  1094. if(get_bits1(&s->gb))
  1095. v = -v;
  1096. *dst = v;
  1097. */
  1098. #if CONFIG_FLOAT
  1099. #define READ_FLIP_SIGN(dst,src)\
  1100. v = AV_RN32A(src) ^ (get_bits1(&s->gb)<<31);\
  1101. AV_WN32A(dst, v);
  1102. #else
  1103. #define READ_FLIP_SIGN(dst,src)\
  1104. v= -get_bits1(&s->gb);\
  1105. *(dst) = (*(src) ^ v) - v;
  1106. #endif
  1107. static int huffman_decode(MPADecodeContext *s, GranuleDef *g,
  1108. int16_t *exponents, int end_pos2)
  1109. {
  1110. int s_index;
  1111. int i;
  1112. int last_pos, bits_left;
  1113. VLC *vlc;
  1114. int end_pos= FFMIN(end_pos2, s->gb.size_in_bits);
  1115. /* low frequencies (called big values) */
  1116. s_index = 0;
  1117. for(i=0;i<3;i++) {
  1118. int j, k, l, linbits;
  1119. j = g->region_size[i];
  1120. if (j == 0)
  1121. continue;
  1122. /* select vlc table */
  1123. k = g->table_select[i];
  1124. l = mpa_huff_data[k][0];
  1125. linbits = mpa_huff_data[k][1];
  1126. vlc = &huff_vlc[l];
  1127. if(!l){
  1128. memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid)*2*j);
  1129. s_index += 2*j;
  1130. continue;
  1131. }
  1132. /* read huffcode and compute each couple */
  1133. for(;j>0;j--) {
  1134. int exponent, x, y;
  1135. int v;
  1136. int pos= get_bits_count(&s->gb);
  1137. if (pos >= end_pos){
  1138. // av_log(NULL, AV_LOG_ERROR, "pos: %d %d %d %d\n", pos, end_pos, end_pos2, s_index);
  1139. switch_buffer(s, &pos, &end_pos, &end_pos2);
  1140. // av_log(NULL, AV_LOG_ERROR, "new pos: %d %d\n", pos, end_pos);
  1141. if(pos >= end_pos)
  1142. break;
  1143. }
  1144. y = get_vlc2(&s->gb, vlc->table, 7, 3);
  1145. if(!y){
  1146. g->sb_hybrid[s_index ] =
  1147. g->sb_hybrid[s_index+1] = 0;
  1148. s_index += 2;
  1149. continue;
  1150. }
  1151. exponent= exponents[s_index];
  1152. dprintf(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n",
  1153. i, g->region_size[i] - j, x, y, exponent);
  1154. if(y&16){
  1155. x = y >> 5;
  1156. y = y & 0x0f;
  1157. if (x < 15){
  1158. READ_FLIP_SIGN(g->sb_hybrid+s_index, RENAME(expval_table)[ exponent ]+x)
  1159. }else{
  1160. x += get_bitsz(&s->gb, linbits);
  1161. v = l3_unscale(x, exponent);
  1162. if (get_bits1(&s->gb))
  1163. v = -v;
  1164. g->sb_hybrid[s_index] = v;
  1165. }
  1166. if (y < 15){
  1167. READ_FLIP_SIGN(g->sb_hybrid+s_index+1, RENAME(expval_table)[ exponent ]+y)
  1168. }else{
  1169. y += get_bitsz(&s->gb, linbits);
  1170. v = l3_unscale(y, exponent);
  1171. if (get_bits1(&s->gb))
  1172. v = -v;
  1173. g->sb_hybrid[s_index+1] = v;
  1174. }
  1175. }else{
  1176. x = y >> 5;
  1177. y = y & 0x0f;
  1178. x += y;
  1179. if (x < 15){
  1180. READ_FLIP_SIGN(g->sb_hybrid+s_index+!!y, RENAME(expval_table)[ exponent ]+x)
  1181. }else{
  1182. x += get_bitsz(&s->gb, linbits);
  1183. v = l3_unscale(x, exponent);
  1184. if (get_bits1(&s->gb))
  1185. v = -v;
  1186. g->sb_hybrid[s_index+!!y] = v;
  1187. }
  1188. g->sb_hybrid[s_index+ !y] = 0;
  1189. }
  1190. s_index+=2;
  1191. }
  1192. }
  1193. /* high frequencies */
  1194. vlc = &huff_quad_vlc[g->count1table_select];
  1195. last_pos=0;
  1196. while (s_index <= 572) {
  1197. int pos, code;
  1198. pos = get_bits_count(&s->gb);
  1199. if (pos >= end_pos) {
  1200. if (pos > end_pos2 && last_pos){
  1201. /* some encoders generate an incorrect size for this
  1202. part. We must go back into the data */
  1203. s_index -= 4;
  1204. skip_bits_long(&s->gb, last_pos - pos);
  1205. av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
  1206. if(s->error_recognition >= FF_ER_COMPLIANT)
  1207. s_index=0;
  1208. break;
  1209. }
  1210. // av_log(NULL, AV_LOG_ERROR, "pos2: %d %d %d %d\n", pos, end_pos, end_pos2, s_index);
  1211. switch_buffer(s, &pos, &end_pos, &end_pos2);
  1212. // av_log(NULL, AV_LOG_ERROR, "new pos2: %d %d %d\n", pos, end_pos, s_index);
  1213. if(pos >= end_pos)
  1214. break;
  1215. }
  1216. last_pos= pos;
  1217. code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
  1218. dprintf(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
  1219. g->sb_hybrid[s_index+0]=
  1220. g->sb_hybrid[s_index+1]=
  1221. g->sb_hybrid[s_index+2]=
  1222. g->sb_hybrid[s_index+3]= 0;
  1223. while(code){
  1224. static const int idxtab[16]={3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0};
  1225. int v;
  1226. int pos= s_index+idxtab[code];
  1227. code ^= 8>>idxtab[code];
  1228. READ_FLIP_SIGN(g->sb_hybrid+pos, RENAME(exp_table)+exponents[pos])
  1229. }
  1230. s_index+=4;
  1231. }
  1232. /* skip extension bits */
  1233. bits_left = end_pos2 - get_bits_count(&s->gb);
  1234. //av_log(NULL, AV_LOG_ERROR, "left:%d buf:%p\n", bits_left, s->in_gb.buffer);
  1235. if (bits_left < 0 && s->error_recognition >= FF_ER_COMPLIANT) {
  1236. av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
  1237. s_index=0;
  1238. }else if(bits_left > 0 && s->error_recognition >= FF_ER_AGGRESSIVE){
  1239. av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
  1240. s_index=0;
  1241. }
  1242. memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid)*(576 - s_index));
  1243. skip_bits_long(&s->gb, bits_left);
  1244. i= get_bits_count(&s->gb);
  1245. switch_buffer(s, &i, &end_pos, &end_pos2);
  1246. return 0;
  1247. }
  1248. /* Reorder short blocks from bitstream order to interleaved order. It
  1249. would be faster to do it in parsing, but the code would be far more
  1250. complicated */
  1251. static void reorder_block(MPADecodeContext *s, GranuleDef *g)
  1252. {
  1253. int i, j, len;
  1254. INTFLOAT *ptr, *dst, *ptr1;
  1255. INTFLOAT tmp[576];
  1256. if (g->block_type != 2)
  1257. return;
  1258. if (g->switch_point) {
  1259. if (s->sample_rate_index != 8) {
  1260. ptr = g->sb_hybrid + 36;
  1261. } else {
  1262. ptr = g->sb_hybrid + 48;
  1263. }
  1264. } else {
  1265. ptr = g->sb_hybrid;
  1266. }
  1267. for(i=g->short_start;i<13;i++) {
  1268. len = band_size_short[s->sample_rate_index][i];
  1269. ptr1 = ptr;
  1270. dst = tmp;
  1271. for(j=len;j>0;j--) {
  1272. *dst++ = ptr[0*len];
  1273. *dst++ = ptr[1*len];
  1274. *dst++ = ptr[2*len];
  1275. ptr++;
  1276. }
  1277. ptr+=2*len;
  1278. memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
  1279. }
  1280. }
  1281. #define ISQRT2 FIXR(0.70710678118654752440)
  1282. static void compute_stereo(MPADecodeContext *s,
  1283. GranuleDef *g0, GranuleDef *g1)
  1284. {
  1285. int i, j, k, l;
  1286. int sf_max, sf, len, non_zero_found;
  1287. INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;
  1288. int non_zero_found_short[3];
  1289. /* intensity stereo */
  1290. if (s->mode_ext & MODE_EXT_I_STEREO) {
  1291. if (!s->lsf) {
  1292. is_tab = is_table;
  1293. sf_max = 7;
  1294. } else {
  1295. is_tab = is_table_lsf[g1->scalefac_compress & 1];
  1296. sf_max = 16;
  1297. }
  1298. tab0 = g0->sb_hybrid + 576;
  1299. tab1 = g1->sb_hybrid + 576;
  1300. non_zero_found_short[0] = 0;
  1301. non_zero_found_short[1] = 0;
  1302. non_zero_found_short[2] = 0;
  1303. k = (13 - g1->short_start) * 3 + g1->long_end - 3;
  1304. for(i = 12;i >= g1->short_start;i--) {
  1305. /* for last band, use previous scale factor */
  1306. if (i != 11)
  1307. k -= 3;
  1308. len = band_size_short[s->sample_rate_index][i];
  1309. for(l=2;l>=0;l--) {
  1310. tab0 -= len;
  1311. tab1 -= len;
  1312. if (!non_zero_found_short[l]) {
  1313. /* test if non zero band. if so, stop doing i-stereo */
  1314. for(j=0;j<len;j++) {
  1315. if (tab1[j] != 0) {
  1316. non_zero_found_short[l] = 1;
  1317. goto found1;
  1318. }
  1319. }
  1320. sf = g1->scale_factors[k + l];
  1321. if (sf >= sf_max)
  1322. goto found1;
  1323. v1 = is_tab[0][sf];
  1324. v2 = is_tab[1][sf];
  1325. for(j=0;j<len;j++) {
  1326. tmp0 = tab0[j];
  1327. tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
  1328. tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
  1329. }
  1330. } else {
  1331. found1:
  1332. if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1333. /* lower part of the spectrum : do ms stereo
  1334. if enabled */
  1335. for(j=0;j<len;j++) {
  1336. tmp0 = tab0[j];
  1337. tmp1 = tab1[j];
  1338. tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
  1339. tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
  1340. }
  1341. }
  1342. }
  1343. }
  1344. }
  1345. non_zero_found = non_zero_found_short[0] |
  1346. non_zero_found_short[1] |
  1347. non_zero_found_short[2];
  1348. for(i = g1->long_end - 1;i >= 0;i--) {
  1349. len = band_size_long[s->sample_rate_index][i];
  1350. tab0 -= len;
  1351. tab1 -= len;
  1352. /* test if non zero band. if so, stop doing i-stereo */
  1353. if (!non_zero_found) {
  1354. for(j=0;j<len;j++) {
  1355. if (tab1[j] != 0) {
  1356. non_zero_found = 1;
  1357. goto found2;
  1358. }
  1359. }
  1360. /* for last band, use previous scale factor */
  1361. k = (i == 21) ? 20 : i;
  1362. sf = g1->scale_factors[k];
  1363. if (sf >= sf_max)
  1364. goto found2;
  1365. v1 = is_tab[0][sf];
  1366. v2 = is_tab[1][sf];
  1367. for(j=0;j<len;j++) {
  1368. tmp0 = tab0[j];
  1369. tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
  1370. tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
  1371. }
  1372. } else {
  1373. found2:
  1374. if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1375. /* lower part of the spectrum : do ms stereo
  1376. if enabled */
  1377. for(j=0;j<len;j++) {
  1378. tmp0 = tab0[j];
  1379. tmp1 = tab1[j];
  1380. tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
  1381. tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
  1382. }
  1383. }
  1384. }
  1385. }
  1386. } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1387. /* ms stereo ONLY */
  1388. /* NOTE: the 1/sqrt(2) normalization factor is included in the
  1389. global gain */
  1390. tab0 = g0->sb_hybrid;
  1391. tab1 = g1->sb_hybrid;
  1392. for(i=0;i<576;i++) {
  1393. tmp0 = tab0[i];
  1394. tmp1 = tab1[i];
  1395. tab0[i] = tmp0 + tmp1;
  1396. tab1[i] = tmp0 - tmp1;
  1397. }
  1398. }
  1399. }
  1400. #if !CONFIG_FLOAT
  1401. static void compute_antialias_integer(MPADecodeContext *s,
  1402. GranuleDef *g)
  1403. {
  1404. int32_t *ptr, *csa;
  1405. int n, i;
  1406. /* we antialias only "long" bands */
  1407. if (g->block_type == 2) {
  1408. if (!g->switch_point)
  1409. return;
  1410. /* XXX: check this for 8000Hz case */
  1411. n = 1;
  1412. } else {
  1413. n = SBLIMIT - 1;
  1414. }
  1415. ptr = g->sb_hybrid + 18;
  1416. for(i = n;i > 0;i--) {
  1417. int tmp0, tmp1, tmp2;
  1418. csa = &csa_table[0][0];
  1419. #define INT_AA(j) \
  1420. tmp0 = ptr[-1-j];\
  1421. tmp1 = ptr[ j];\
  1422. tmp2= MULH(tmp0 + tmp1, csa[0+4*j]);\
  1423. ptr[-1-j] = 4*(tmp2 - MULH(tmp1, csa[2+4*j]));\
  1424. ptr[ j] = 4*(tmp2 + MULH(tmp0, csa[3+4*j]));
  1425. INT_AA(0)
  1426. INT_AA(1)
  1427. INT_AA(2)
  1428. INT_AA(3)
  1429. INT_AA(4)
  1430. INT_AA(5)
  1431. INT_AA(6)
  1432. INT_AA(7)
  1433. ptr += 18;
  1434. }
  1435. }
  1436. #endif
  1437. static void compute_imdct(MPADecodeContext *s,
  1438. GranuleDef *g,
  1439. INTFLOAT *sb_samples,
  1440. INTFLOAT *mdct_buf)
  1441. {
  1442. INTFLOAT *win, *win1, *out_ptr, *ptr, *buf, *ptr1;
  1443. INTFLOAT out2[12];
  1444. int i, j, mdct_long_end, sblimit;
  1445. /* find last non zero block */
  1446. ptr = g->sb_hybrid + 576;
  1447. ptr1 = g->sb_hybrid + 2 * 18;
  1448. while (ptr >= ptr1) {
  1449. int32_t *p;
  1450. ptr -= 6;
  1451. p= (int32_t*)ptr;
  1452. if(p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
  1453. break;
  1454. }
  1455. sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
  1456. if (g->block_type == 2) {
  1457. /* XXX: check for 8000 Hz */
  1458. if (g->switch_point)
  1459. mdct_long_end = 2;
  1460. else
  1461. mdct_long_end = 0;
  1462. } else {
  1463. mdct_long_end = sblimit;
  1464. }
  1465. buf = mdct_buf;
  1466. ptr = g->sb_hybrid;
  1467. for(j=0;j<mdct_long_end;j++) {
  1468. /* apply window & overlap with previous buffer */
  1469. out_ptr = sb_samples + j;
  1470. /* select window */
  1471. if (g->switch_point && j < 2)
  1472. win1 = mdct_win[0];
  1473. else
  1474. win1 = mdct_win[g->block_type];
  1475. /* select frequency inversion */
  1476. win = win1 + ((4 * 36) & -(j & 1));
  1477. imdct36(out_ptr, buf, ptr, win);
  1478. out_ptr += 18*SBLIMIT;
  1479. ptr += 18;
  1480. buf += 18;
  1481. }
  1482. for(j=mdct_long_end;j<sblimit;j++) {
  1483. /* select frequency inversion */
  1484. win = mdct_win[2] + ((4 * 36) & -(j & 1));
  1485. out_ptr = sb_samples + j;
  1486. for(i=0; i<6; i++){
  1487. *out_ptr = buf[i];
  1488. out_ptr += SBLIMIT;
  1489. }
  1490. imdct12(out2, ptr + 0);
  1491. for(i=0;i<6;i++) {
  1492. *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[i + 6*1];
  1493. buf[i + 6*2] = MULH3(out2[i + 6], win[i + 6], 1);
  1494. out_ptr += SBLIMIT;
  1495. }
  1496. imdct12(out2, ptr + 1);
  1497. for(i=0;i<6;i++) {
  1498. *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[i + 6*2];
  1499. buf[i + 6*0] = MULH3(out2[i + 6], win[i + 6], 1);
  1500. out_ptr += SBLIMIT;
  1501. }
  1502. imdct12(out2, ptr + 2);
  1503. for(i=0;i<6;i++) {
  1504. buf[i + 6*0] = MULH3(out2[i ], win[i ], 1) + buf[i + 6*0];
  1505. buf[i + 6*1] = MULH3(out2[i + 6], win[i + 6], 1);
  1506. buf[i + 6*2] = 0;
  1507. }
  1508. ptr += 18;
  1509. buf += 18;
  1510. }
  1511. /* zero bands */
  1512. for(j=sblimit;j<SBLIMIT;j++) {
  1513. /* overlap */
  1514. out_ptr = sb_samples + j;
  1515. for(i=0;i<18;i++) {
  1516. *out_ptr = buf[i];
  1517. buf[i] = 0;
  1518. out_ptr += SBLIMIT;
  1519. }
  1520. buf += 18;
  1521. }
  1522. }
  1523. /* main layer3 decoding function */
  1524. static int mp_decode_layer3(MPADecodeContext *s)
  1525. {
  1526. int nb_granules, main_data_begin, private_bits;
  1527. int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
  1528. GranuleDef *g;
  1529. int16_t exponents[576]; //FIXME try INTFLOAT
  1530. /* read side info */
  1531. if (s->lsf) {
  1532. main_data_begin = get_bits(&s->gb, 8);
  1533. private_bits = get_bits(&s->gb, s->nb_channels);
  1534. nb_granules = 1;
  1535. } else {
  1536. main_data_begin = get_bits(&s->gb, 9);
  1537. if (s->nb_channels == 2)
  1538. private_bits = get_bits(&s->gb, 3);
  1539. else
  1540. private_bits = get_bits(&s->gb, 5);
  1541. nb_granules = 2;
  1542. for(ch=0;ch<s->nb_channels;ch++) {
  1543. s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
  1544. s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
  1545. }
  1546. }
  1547. for(gr=0;gr<nb_granules;gr++) {
  1548. for(ch=0;ch<s->nb_channels;ch++) {
  1549. dprintf(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
  1550. g = &s->granules[ch][gr];
  1551. g->part2_3_length = get_bits(&s->gb, 12);
  1552. g->big_values = get_bits(&s->gb, 9);
  1553. if(g->big_values > 288){
  1554. av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
  1555. return -1;
  1556. }
  1557. g->global_gain = get_bits(&s->gb, 8);
  1558. /* if MS stereo only is selected, we precompute the
  1559. 1/sqrt(2) renormalization factor */
  1560. if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
  1561. MODE_EXT_MS_STEREO)
  1562. g->global_gain -= 2;
  1563. if (s->lsf)
  1564. g->scalefac_compress = get_bits(&s->gb, 9);
  1565. else
  1566. g->scalefac_compress = get_bits(&s->gb, 4);
  1567. blocksplit_flag = get_bits1(&s->gb);
  1568. if (blocksplit_flag) {
  1569. g->block_type = get_bits(&s->gb, 2);
  1570. if (g->block_type == 0){
  1571. av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
  1572. return -1;
  1573. }
  1574. g->switch_point = get_bits1(&s->gb);
  1575. for(i=0;i<2;i++)
  1576. g->table_select[i] = get_bits(&s->gb, 5);
  1577. for(i=0;i<3;i++)
  1578. g->subblock_gain[i] = get_bits(&s->gb, 3);
  1579. ff_init_short_region(s, g);
  1580. } else {
  1581. int region_address1, region_address2;
  1582. g->block_type = 0;
  1583. g->switch_point = 0;
  1584. for(i=0;i<3;i++)
  1585. g->table_select[i] = get_bits(&s->gb, 5);
  1586. /* compute huffman coded region sizes */
  1587. region_address1 = get_bits(&s->gb, 4);
  1588. region_address2 = get_bits(&s->gb, 3);
  1589. dprintf(s->avctx, "region1=%d region2=%d\n",
  1590. region_address1, region_address2);
  1591. ff_init_long_region(s, g, region_address1, region_address2);
  1592. }
  1593. ff_region_offset2size(g);
  1594. ff_compute_band_indexes(s, g);
  1595. g->preflag = 0;
  1596. if (!s->lsf)
  1597. g->preflag = get_bits1(&s->gb);
  1598. g->scalefac_scale = get_bits1(&s->gb);
  1599. g->count1table_select = get_bits1(&s->gb);
  1600. dprintf(s->avctx, "block_type=%d switch_point=%d\n",
  1601. g->block_type, g->switch_point);
  1602. }
  1603. }
  1604. if (!s->adu_mode) {
  1605. const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
  1606. assert((get_bits_count(&s->gb) & 7) == 0);
  1607. /* now we get bits from the main_data_begin offset */
  1608. dprintf(s->avctx, "seekback: %d\n", main_data_begin);
  1609. //av_log(NULL, AV_LOG_ERROR, "backstep:%d, lastbuf:%d\n", main_data_begin, s->last_buf_size);
  1610. memcpy(s->last_buf + s->last_buf_size, ptr, EXTRABYTES);
  1611. s->in_gb= s->gb;
  1612. init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);
  1613. skip_bits_long(&s->gb, 8*(s->last_buf_size - main_data_begin));
  1614. }
  1615. for(gr=0;gr<nb_granules;gr++) {
  1616. for(ch=0;ch<s->nb_channels;ch++) {
  1617. g = &s->granules[ch][gr];
  1618. if(get_bits_count(&s->gb)<0){
  1619. av_log(s->avctx, AV_LOG_DEBUG, "mdb:%d, lastbuf:%d skipping granule %d\n",
  1620. main_data_begin, s->last_buf_size, gr);
  1621. skip_bits_long(&s->gb, g->part2_3_length);
  1622. memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
  1623. if(get_bits_count(&s->gb) >= s->gb.size_in_bits && s->in_gb.buffer){
  1624. skip_bits_long(&s->in_gb, get_bits_count(&s->gb) - s->gb.size_in_bits);
  1625. s->gb= s->in_gb;
  1626. s->in_gb.buffer=NULL;
  1627. }
  1628. continue;
  1629. }
  1630. bits_pos = get_bits_count(&s->gb);
  1631. if (!s->lsf) {
  1632. uint8_t *sc;
  1633. int slen, slen1, slen2;
  1634. /* MPEG1 scale factors */
  1635. slen1 = slen_table[0][g->scalefac_compress];
  1636. slen2 = slen_table[1][g->scalefac_compress];
  1637. dprintf(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
  1638. if (g->block_type == 2) {
  1639. n = g->switch_point ? 17 : 18;
  1640. j = 0;
  1641. if(slen1){
  1642. for(i=0;i<n;i++)
  1643. g->scale_factors[j++] = get_bits(&s->gb, slen1);
  1644. }else{
  1645. for(i=0;i<n;i++)
  1646. g->scale_factors[j++] = 0;
  1647. }
  1648. if(slen2){
  1649. for(i=0;i<18;i++)
  1650. g->scale_factors[j++] = get_bits(&s->gb, slen2);
  1651. for(i=0;i<3;i++)
  1652. g->scale_factors[j++] = 0;
  1653. }else{
  1654. for(i=0;i<21;i++)
  1655. g->scale_factors[j++] = 0;
  1656. }
  1657. } else {
  1658. sc = s->granules[ch][0].scale_factors;
  1659. j = 0;
  1660. for(k=0;k<4;k++) {
  1661. n = (k == 0 ? 6 : 5);
  1662. if ((g->scfsi & (0x8 >> k)) == 0) {
  1663. slen = (k < 2) ? slen1 : slen2;
  1664. if(slen){
  1665. for(i=0;i<n;i++)
  1666. g->scale_factors[j++] = get_bits(&s->gb, slen);
  1667. }else{
  1668. for(i=0;i<n;i++)
  1669. g->scale_factors[j++] = 0;
  1670. }
  1671. } else {
  1672. /* simply copy from last granule */
  1673. for(i=0;i<n;i++) {
  1674. g->scale_factors[j] = sc[j];
  1675. j++;
  1676. }
  1677. }
  1678. }
  1679. g->scale_factors[j++] = 0;
  1680. }
  1681. } else {
  1682. int tindex, tindex2, slen[4], sl, sf;
  1683. /* LSF scale factors */
  1684. if (g->block_type == 2) {
  1685. tindex = g->switch_point ? 2 : 1;
  1686. } else {
  1687. tindex = 0;
  1688. }
  1689. sf = g->scalefac_compress;
  1690. if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
  1691. /* intensity stereo case */
  1692. sf >>= 1;
  1693. if (sf < 180) {
  1694. lsf_sf_expand(slen, sf, 6, 6, 0);
  1695. tindex2 = 3;
  1696. } else if (sf < 244) {
  1697. lsf_sf_expand(slen, sf - 180, 4, 4, 0);
  1698. tindex2 = 4;
  1699. } else {
  1700. lsf_sf_expand(slen, sf - 244, 3, 0, 0);
  1701. tindex2 = 5;
  1702. }
  1703. } else {
  1704. /* normal case */
  1705. if (sf < 400) {
  1706. lsf_sf_expand(slen, sf, 5, 4, 4);
  1707. tindex2 = 0;
  1708. } else if (sf < 500) {
  1709. lsf_sf_expand(slen, sf - 400, 5, 4, 0);
  1710. tindex2 = 1;
  1711. } else {
  1712. lsf_sf_expand(slen, sf - 500, 3, 0, 0);
  1713. tindex2 = 2;
  1714. g->preflag = 1;
  1715. }
  1716. }
  1717. j = 0;
  1718. for(k=0;k<4;k++) {
  1719. n = lsf_nsf_table[tindex2][tindex][k];
  1720. sl = slen[k];
  1721. if(sl){
  1722. for(i=0;i<n;i++)
  1723. g->scale_factors[j++] = get_bits(&s->gb, sl);
  1724. }else{
  1725. for(i=0;i<n;i++)
  1726. g->scale_factors[j++] = 0;
  1727. }
  1728. }
  1729. /* XXX: should compute exact size */
  1730. for(;j<40;j++)
  1731. g->scale_factors[j] = 0;
  1732. }
  1733. exponents_from_scale_factors(s, g, exponents);
  1734. /* read Huffman coded residue */
  1735. huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
  1736. } /* ch */
  1737. if (s->nb_channels == 2)
  1738. compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
  1739. for(ch=0;ch<s->nb_channels;ch++) {
  1740. g = &s->granules[ch][gr];
  1741. reorder_block(s, g);
  1742. compute_antialias(s, g);
  1743. compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
  1744. }
  1745. } /* gr */
  1746. if(get_bits_count(&s->gb)<0)
  1747. skip_bits_long(&s->gb, -get_bits_count(&s->gb));
  1748. return nb_granules * 18;
  1749. }
  1750. static int mp_decode_frame(MPADecodeContext *s,
  1751. OUT_INT *samples, const uint8_t *buf, int buf_size)
  1752. {
  1753. int i, nb_frames, ch;
  1754. OUT_INT *samples_ptr;
  1755. init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE)*8);
  1756. /* skip error protection field */
  1757. if (s->error_protection)
  1758. skip_bits(&s->gb, 16);
  1759. dprintf(s->avctx, "frame %d:\n", s->frame_count);
  1760. switch(s->layer) {
  1761. case 1:
  1762. s->avctx->frame_size = 384;
  1763. nb_frames = mp_decode_layer1(s);
  1764. break;
  1765. case 2:
  1766. s->avctx->frame_size = 1152;
  1767. nb_frames = mp_decode_layer2(s);
  1768. break;
  1769. case 3:
  1770. s->avctx->frame_size = s->lsf ? 576 : 1152;
  1771. default:
  1772. nb_frames = mp_decode_layer3(s);
  1773. s->last_buf_size=0;
  1774. if(s->in_gb.buffer){
  1775. align_get_bits(&s->gb);
  1776. i= get_bits_left(&s->gb)>>3;
  1777. if(i >= 0 && i <= BACKSTEP_SIZE){
  1778. memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);
  1779. s->last_buf_size=i;
  1780. }else
  1781. av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
  1782. s->gb= s->in_gb;
  1783. s->in_gb.buffer= NULL;
  1784. }
  1785. align_get_bits(&s->gb);
  1786. assert((get_bits_count(&s->gb) & 7) == 0);
  1787. i= get_bits_left(&s->gb)>>3;
  1788. if(i<0 || i > BACKSTEP_SIZE || nb_frames<0){
  1789. if(i<0)
  1790. av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
  1791. i= FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
  1792. }
  1793. assert(i <= buf_size - HEADER_SIZE && i>= 0);
  1794. memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
  1795. s->last_buf_size += i;
  1796. break;
  1797. }
  1798. /* apply the synthesis filter */
  1799. for(ch=0;ch<s->nb_channels;ch++) {
  1800. samples_ptr = samples + ch;
  1801. for(i=0;i<nb_frames;i++) {
  1802. RENAME(ff_mpa_synth_filter)(
  1803. #if CONFIG_FLOAT
  1804. s,
  1805. #endif
  1806. s->synth_buf[ch], &(s->synth_buf_offset[ch]),
  1807. RENAME(ff_mpa_synth_window), &s->dither_state,
  1808. samples_ptr, s->nb_channels,
  1809. s->sb_samples[ch][i]);
  1810. samples_ptr += 32 * s->nb_channels;
  1811. }
  1812. }
  1813. return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
  1814. }
  1815. static int decode_frame(AVCodecContext * avctx,
  1816. void *data, int *data_size,
  1817. AVPacket *avpkt)
  1818. {
  1819. const uint8_t *buf = avpkt->data;
  1820. int buf_size = avpkt->size;
  1821. MPADecodeContext *s = avctx->priv_data;
  1822. uint32_t header;
  1823. int out_size;
  1824. OUT_INT *out_samples = data;
  1825. if(buf_size < HEADER_SIZE)
  1826. return -1;
  1827. header = AV_RB32(buf);
  1828. if(ff_mpa_check_header(header) < 0){
  1829. av_log(avctx, AV_LOG_ERROR, "Header missing\n");
  1830. return -1;
  1831. }
  1832. if (ff_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
  1833. /* free format: prepare to compute frame size */
  1834. s->frame_size = -1;
  1835. return -1;
  1836. }
  1837. /* update codec info */
  1838. avctx->channels = s->nb_channels;
  1839. avctx->bit_rate = s->bit_rate;
  1840. avctx->sub_id = s->layer;
  1841. if(*data_size < 1152*avctx->channels*sizeof(OUT_INT))
  1842. return -1;
  1843. *data_size = 0;
  1844. if(s->frame_size<=0 || s->frame_size > buf_size){
  1845. av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
  1846. return -1;
  1847. }else if(s->frame_size < buf_size){
  1848. av_log(avctx, AV_LOG_ERROR, "incorrect frame size\n");
  1849. buf_size= s->frame_size;
  1850. }
  1851. out_size = mp_decode_frame(s, out_samples, buf, buf_size);
  1852. if(out_size>=0){
  1853. *data_size = out_size;
  1854. avctx->sample_rate = s->sample_rate;
  1855. //FIXME maybe move the other codec info stuff from above here too
  1856. }else
  1857. av_log(avctx, AV_LOG_DEBUG, "Error while decoding MPEG audio frame.\n"); //FIXME return -1 / but also return the number of bytes consumed
  1858. s->frame_size = 0;
  1859. return buf_size;
  1860. }
  1861. static void flush(AVCodecContext *avctx){
  1862. MPADecodeContext *s = avctx->priv_data;
  1863. memset(s->synth_buf, 0, sizeof(s->synth_buf));
  1864. s->last_buf_size= 0;
  1865. }
  1866. #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
  1867. static int decode_frame_adu(AVCodecContext * avctx,
  1868. void *data, int *data_size,
  1869. AVPacket *avpkt)
  1870. {
  1871. const uint8_t *buf = avpkt->data;
  1872. int buf_size = avpkt->size;
  1873. MPADecodeContext *s = avctx->priv_data;
  1874. uint32_t header;
  1875. int len, out_size;
  1876. OUT_INT *out_samples = data;
  1877. len = buf_size;
  1878. // Discard too short frames
  1879. if (buf_size < HEADER_SIZE) {
  1880. *data_size = 0;
  1881. return buf_size;
  1882. }
  1883. if (len > MPA_MAX_CODED_FRAME_SIZE)
  1884. len = MPA_MAX_CODED_FRAME_SIZE;
  1885. // Get header and restore sync word
  1886. header = AV_RB32(buf) | 0xffe00000;
  1887. if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame
  1888. *data_size = 0;
  1889. return buf_size;
  1890. }
  1891. ff_mpegaudio_decode_header((MPADecodeHeader *)s, header);
  1892. /* update codec info */
  1893. avctx->sample_rate = s->sample_rate;
  1894. avctx->channels = s->nb_channels;
  1895. avctx->bit_rate = s->bit_rate;
  1896. avctx->sub_id = s->layer;
  1897. s->frame_size = len;
  1898. if (avctx->parse_only) {
  1899. out_size = buf_size;
  1900. } else {
  1901. out_size = mp_decode_frame(s, out_samples, buf, buf_size);
  1902. }
  1903. *data_size = out_size;
  1904. return buf_size;
  1905. }
  1906. #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
  1907. #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
  1908. /**
  1909. * Context for MP3On4 decoder
  1910. */
  1911. typedef struct MP3On4DecodeContext {
  1912. int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
  1913. int syncword; ///< syncword patch
  1914. const uint8_t *coff; ///< channels offsets in output buffer
  1915. MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
  1916. } MP3On4DecodeContext;
  1917. #include "mpeg4audio.h"
  1918. /* Next 3 arrays are indexed by channel config number (passed via codecdata) */
  1919. static const uint8_t mp3Frames[8] = {0,1,1,2,3,3,4,5}; /* number of mp3 decoder instances */
  1920. /* offsets into output buffer, assume output order is FL FR BL BR C LFE */
  1921. static const uint8_t chan_offset[8][5] = {
  1922. {0},
  1923. {0}, // C
  1924. {0}, // FLR
  1925. {2,0}, // C FLR
  1926. {2,0,3}, // C FLR BS
  1927. {4,0,2}, // C FLR BLRS
  1928. {4,0,2,5}, // C FLR BLRS LFE
  1929. {4,0,2,6,5}, // C FLR BLRS BLR LFE
  1930. };
  1931. static int decode_init_mp3on4(AVCodecContext * avctx)
  1932. {
  1933. MP3On4DecodeContext *s = avctx->priv_data;
  1934. MPEG4AudioConfig cfg;
  1935. int i;
  1936. if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) {
  1937. av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
  1938. return -1;
  1939. }
  1940. ff_mpeg4audio_get_config(&cfg, avctx->extradata, avctx->extradata_size);
  1941. if (!cfg.chan_config || cfg.chan_config > 7) {
  1942. av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
  1943. return -1;
  1944. }
  1945. s->frames = mp3Frames[cfg.chan_config];
  1946. s->coff = chan_offset[cfg.chan_config];
  1947. avctx->channels = ff_mpeg4audio_channels[cfg.chan_config];
  1948. if (cfg.sample_rate < 16000)
  1949. s->syncword = 0xffe00000;
  1950. else
  1951. s->syncword = 0xfff00000;
  1952. /* Init the first mp3 decoder in standard way, so that all tables get builded
  1953. * We replace avctx->priv_data with the context of the first decoder so that
  1954. * decode_init() does not have to be changed.
  1955. * Other decoders will be initialized here copying data from the first context
  1956. */
  1957. // Allocate zeroed memory for the first decoder context
  1958. s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
  1959. // Put decoder context in place to make init_decode() happy
  1960. avctx->priv_data = s->mp3decctx[0];
  1961. decode_init(avctx);
  1962. // Restore mp3on4 context pointer
  1963. avctx->priv_data = s;
  1964. s->mp3decctx[0]->adu_mode = 1; // Set adu mode
  1965. /* Create a separate codec/context for each frame (first is already ok).
  1966. * Each frame is 1 or 2 channels - up to 5 frames allowed
  1967. */
  1968. for (i = 1; i < s->frames; i++) {
  1969. s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
  1970. s->mp3decctx[i]->adu_mode = 1;
  1971. s->mp3decctx[i]->avctx = avctx;
  1972. }
  1973. return 0;
  1974. }
  1975. static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
  1976. {
  1977. MP3On4DecodeContext *s = avctx->priv_data;
  1978. int i;
  1979. for (i = 0; i < s->frames; i++)
  1980. if (s->mp3decctx[i])
  1981. av_free(s->mp3decctx[i]);
  1982. return 0;
  1983. }
  1984. static int decode_frame_mp3on4(AVCodecContext * avctx,
  1985. void *data, int *data_size,
  1986. AVPacket *avpkt)
  1987. {
  1988. const uint8_t *buf = avpkt->data;
  1989. int buf_size = avpkt->size;
  1990. MP3On4DecodeContext *s = avctx->priv_data;
  1991. MPADecodeContext *m;
  1992. int fsize, len = buf_size, out_size = 0;
  1993. uint32_t header;
  1994. OUT_INT *out_samples = data;
  1995. OUT_INT decoded_buf[MPA_FRAME_SIZE * MPA_MAX_CHANNELS];
  1996. OUT_INT *outptr, *bp;
  1997. int fr, j, n;
  1998. if(*data_size < MPA_FRAME_SIZE * MPA_MAX_CHANNELS * s->frames * sizeof(OUT_INT))
  1999. return -1;
  2000. *data_size = 0;
  2001. // Discard too short frames
  2002. if (buf_size < HEADER_SIZE)
  2003. return -1;
  2004. // If only one decoder interleave is not needed
  2005. outptr = s->frames == 1 ? out_samples : decoded_buf;
  2006. avctx->bit_rate = 0;
  2007. for (fr = 0; fr < s->frames; fr++) {
  2008. fsize = AV_RB16(buf) >> 4;
  2009. fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
  2010. m = s->mp3decctx[fr];
  2011. assert (m != NULL);
  2012. header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
  2013. if (ff_mpa_check_header(header) < 0) // Bad header, discard block
  2014. break;
  2015. ff_mpegaudio_decode_header((MPADecodeHeader *)m, header);
  2016. out_size += mp_decode_frame(m, outptr, buf, fsize);
  2017. buf += fsize;
  2018. len -= fsize;
  2019. if(s->frames > 1) {
  2020. n = m->avctx->frame_size*m->nb_channels;
  2021. /* interleave output data */
  2022. bp = out_samples + s->coff[fr];
  2023. if(m->nb_channels == 1) {
  2024. for(j = 0; j < n; j++) {
  2025. *bp = decoded_buf[j];
  2026. bp += avctx->channels;
  2027. }
  2028. } else {
  2029. for(j = 0; j < n; j++) {
  2030. bp[0] = decoded_buf[j++];
  2031. bp[1] = decoded_buf[j];
  2032. bp += avctx->channels;
  2033. }
  2034. }
  2035. }
  2036. avctx->bit_rate += m->bit_rate;
  2037. }
  2038. /* update codec info */
  2039. avctx->sample_rate = s->mp3decctx[0]->sample_rate;
  2040. *data_size = out_size;
  2041. return buf_size;
  2042. }
  2043. #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
  2044. #if !CONFIG_FLOAT
  2045. #if CONFIG_MP1_DECODER
  2046. AVCodec mp1_decoder =
  2047. {
  2048. "mp1",
  2049. AVMEDIA_TYPE_AUDIO,
  2050. CODEC_ID_MP1,
  2051. sizeof(MPADecodeContext),
  2052. decode_init,
  2053. NULL,
  2054. NULL,
  2055. decode_frame,
  2056. CODEC_CAP_PARSE_ONLY,
  2057. .flush= flush,
  2058. .long_name= NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
  2059. };
  2060. #endif
  2061. #if CONFIG_MP2_DECODER
  2062. AVCodec mp2_decoder =
  2063. {
  2064. "mp2",
  2065. AVMEDIA_TYPE_AUDIO,
  2066. CODEC_ID_MP2,
  2067. sizeof(MPADecodeContext),
  2068. decode_init,
  2069. NULL,
  2070. NULL,
  2071. decode_frame,
  2072. CODEC_CAP_PARSE_ONLY,
  2073. .flush= flush,
  2074. .long_name= NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
  2075. };
  2076. #endif
  2077. #if CONFIG_MP3_DECODER
  2078. AVCodec mp3_decoder =
  2079. {
  2080. "mp3",
  2081. AVMEDIA_TYPE_AUDIO,
  2082. CODEC_ID_MP3,
  2083. sizeof(MPADecodeContext),
  2084. decode_init,
  2085. NULL,
  2086. NULL,
  2087. decode_frame,
  2088. CODEC_CAP_PARSE_ONLY,
  2089. .flush= flush,
  2090. .long_name= NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
  2091. };
  2092. #endif
  2093. #if CONFIG_MP3ADU_DECODER
  2094. AVCodec mp3adu_decoder =
  2095. {
  2096. "mp3adu",
  2097. AVMEDIA_TYPE_AUDIO,
  2098. CODEC_ID_MP3ADU,
  2099. sizeof(MPADecodeContext),
  2100. decode_init,
  2101. NULL,
  2102. NULL,
  2103. decode_frame_adu,
  2104. CODEC_CAP_PARSE_ONLY,
  2105. .flush= flush,
  2106. .long_name= NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
  2107. };
  2108. #endif
  2109. #if CONFIG_MP3ON4_DECODER
  2110. AVCodec mp3on4_decoder =
  2111. {
  2112. "mp3on4",
  2113. AVMEDIA_TYPE_AUDIO,
  2114. CODEC_ID_MP3ON4,
  2115. sizeof(MP3On4DecodeContext),
  2116. decode_init_mp3on4,
  2117. NULL,
  2118. decode_close_mp3on4,
  2119. decode_frame_mp3on4,
  2120. .flush= flush,
  2121. .long_name= NULL_IF_CONFIG_SMALL("MP3onMP4"),
  2122. };
  2123. #endif
  2124. #endif