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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 "libavutil/audioconvert.h"
  26. #include "avcodec.h"
  27. #include "get_bits.h"
  28. #include "dsputil.h"
  29. #include "mathops.h"
  30. #include "dct32.h"
  31. /*
  32. * TODO:
  33. * - test lsf / mpeg25 extensively.
  34. */
  35. #include "mpegaudio.h"
  36. #include "mpegaudiodecheader.h"
  37. #if CONFIG_FLOAT
  38. # define SHR(a,b) ((a)*(1.0f/(1<<(b))))
  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. # define OUT_FMT AV_SAMPLE_FMT_FLT
  46. #else
  47. # define SHR(a,b) ((a)>>(b))
  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 ## _fixed
  55. # define OUT_FMT AV_SAMPLE_FMT_S16
  56. #endif
  57. /****************/
  58. #define HEADER_SIZE 4
  59. #include "mpegaudiodata.h"
  60. #include "mpegaudiodectab.h"
  61. static void RENAME(compute_antialias)(MPADecodeContext *s, GranuleDef *g);
  62. static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
  63. int *dither_state, OUT_INT *samples, int incr);
  64. /* vlc structure for decoding layer 3 huffman tables */
  65. static VLC huff_vlc[16];
  66. static VLC_TYPE huff_vlc_tables[
  67. 0+128+128+128+130+128+154+166+
  68. 142+204+190+170+542+460+662+414
  69. ][2];
  70. static const int huff_vlc_tables_sizes[16] = {
  71. 0, 128, 128, 128, 130, 128, 154, 166,
  72. 142, 204, 190, 170, 542, 460, 662, 414
  73. };
  74. static VLC huff_quad_vlc[2];
  75. static VLC_TYPE huff_quad_vlc_tables[128+16][2];
  76. static const int huff_quad_vlc_tables_sizes[2] = {
  77. 128, 16
  78. };
  79. /* computed from band_size_long */
  80. static uint16_t band_index_long[9][23];
  81. #include "mpegaudio_tablegen.h"
  82. /* intensity stereo coef table */
  83. static INTFLOAT is_table[2][16];
  84. static INTFLOAT is_table_lsf[2][2][16];
  85. static int32_t csa_table[8][4];
  86. static float csa_table_float[8][4];
  87. static INTFLOAT mdct_win[8][36];
  88. static int16_t division_tab3[1<<6 ];
  89. static int16_t division_tab5[1<<8 ];
  90. static int16_t division_tab9[1<<11];
  91. static int16_t * const division_tabs[4] = {
  92. division_tab3, division_tab5, NULL, division_tab9
  93. };
  94. /* lower 2 bits: modulo 3, higher bits: shift */
  95. static uint16_t scale_factor_modshift[64];
  96. /* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
  97. static int32_t scale_factor_mult[15][3];
  98. /* mult table for layer 2 group quantization */
  99. #define SCALE_GEN(v) \
  100. { FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
  101. static const int32_t scale_factor_mult2[3][3] = {
  102. SCALE_GEN(4.0 / 3.0), /* 3 steps */
  103. SCALE_GEN(4.0 / 5.0), /* 5 steps */
  104. SCALE_GEN(4.0 / 9.0), /* 9 steps */
  105. };
  106. DECLARE_ALIGNED(16, MPA_INT, RENAME(ff_mpa_synth_window))[512+256];
  107. /**
  108. * Convert region offsets to region sizes and truncate
  109. * size to big_values.
  110. */
  111. static void ff_region_offset2size(GranuleDef *g){
  112. int i, k, j=0;
  113. g->region_size[2] = (576 / 2);
  114. for(i=0;i<3;i++) {
  115. k = FFMIN(g->region_size[i], g->big_values);
  116. g->region_size[i] = k - j;
  117. j = k;
  118. }
  119. }
  120. static void ff_init_short_region(MPADecodeContext *s, GranuleDef *g){
  121. if (g->block_type == 2)
  122. g->region_size[0] = (36 / 2);
  123. else {
  124. if (s->sample_rate_index <= 2)
  125. g->region_size[0] = (36 / 2);
  126. else if (s->sample_rate_index != 8)
  127. g->region_size[0] = (54 / 2);
  128. else
  129. g->region_size[0] = (108 / 2);
  130. }
  131. g->region_size[1] = (576 / 2);
  132. }
  133. static void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2){
  134. int l;
  135. g->region_size[0] =
  136. band_index_long[s->sample_rate_index][ra1 + 1] >> 1;
  137. /* should not overflow */
  138. l = FFMIN(ra1 + ra2 + 2, 22);
  139. g->region_size[1] =
  140. band_index_long[s->sample_rate_index][l] >> 1;
  141. }
  142. static void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g){
  143. if (g->block_type == 2) {
  144. if (g->switch_point) {
  145. /* if switched mode, we handle the 36 first samples as
  146. long blocks. For 8000Hz, we handle the 48 first
  147. exponents as long blocks (XXX: check this!) */
  148. if (s->sample_rate_index <= 2)
  149. g->long_end = 8;
  150. else if (s->sample_rate_index != 8)
  151. g->long_end = 6;
  152. else
  153. g->long_end = 4; /* 8000 Hz */
  154. g->short_start = 2 + (s->sample_rate_index != 8);
  155. } else {
  156. g->long_end = 0;
  157. g->short_start = 0;
  158. }
  159. } else {
  160. g->short_start = 13;
  161. g->long_end = 22;
  162. }
  163. }
  164. /* layer 1 unscaling */
  165. /* n = number of bits of the mantissa minus 1 */
  166. static inline int l1_unscale(int n, int mant, int scale_factor)
  167. {
  168. int shift, mod;
  169. int64_t val;
  170. shift = scale_factor_modshift[scale_factor];
  171. mod = shift & 3;
  172. shift >>= 2;
  173. val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]);
  174. shift += n;
  175. /* NOTE: at this point, 1 <= shift >= 21 + 15 */
  176. return (int)((val + (1LL << (shift - 1))) >> shift);
  177. }
  178. static inline int l2_unscale_group(int steps, int mant, int scale_factor)
  179. {
  180. int shift, mod, val;
  181. shift = scale_factor_modshift[scale_factor];
  182. mod = shift & 3;
  183. shift >>= 2;
  184. val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
  185. /* NOTE: at this point, 0 <= shift <= 21 */
  186. if (shift > 0)
  187. val = (val + (1 << (shift - 1))) >> shift;
  188. return val;
  189. }
  190. /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
  191. static inline int l3_unscale(int value, int exponent)
  192. {
  193. unsigned int m;
  194. int e;
  195. e = table_4_3_exp [4*value + (exponent&3)];
  196. m = table_4_3_value[4*value + (exponent&3)];
  197. e -= (exponent >> 2);
  198. assert(e>=1);
  199. if (e > 31)
  200. return 0;
  201. m = (m + (1 << (e-1))) >> e;
  202. return m;
  203. }
  204. /* all integer n^(4/3) computation code */
  205. #define DEV_ORDER 13
  206. #define POW_FRAC_BITS 24
  207. #define POW_FRAC_ONE (1 << POW_FRAC_BITS)
  208. #define POW_FIX(a) ((int)((a) * POW_FRAC_ONE))
  209. #define POW_MULL(a,b) (((int64_t)(a) * (int64_t)(b)) >> POW_FRAC_BITS)
  210. static int dev_4_3_coefs[DEV_ORDER];
  211. static av_cold void int_pow_init(void)
  212. {
  213. int i, a;
  214. a = POW_FIX(1.0);
  215. for(i=0;i<DEV_ORDER;i++) {
  216. a = POW_MULL(a, POW_FIX(4.0 / 3.0) - i * POW_FIX(1.0)) / (i + 1);
  217. dev_4_3_coefs[i] = a;
  218. }
  219. }
  220. static av_cold int decode_init(AVCodecContext * avctx)
  221. {
  222. MPADecodeContext *s = avctx->priv_data;
  223. static int init=0;
  224. int i, j, k;
  225. s->avctx = avctx;
  226. s->apply_window_mp3 = apply_window_mp3_c;
  227. #if HAVE_MMX && CONFIG_FLOAT
  228. ff_mpegaudiodec_init_mmx(s);
  229. #endif
  230. #if CONFIG_FLOAT
  231. ff_dct_init(&s->dct, 5, DCT_II);
  232. #endif
  233. if (HAVE_ALTIVEC && CONFIG_FLOAT) ff_mpegaudiodec_init_altivec(s);
  234. avctx->sample_fmt= OUT_FMT;
  235. s->error_recognition= avctx->error_recognition;
  236. if (!init && !avctx->parse_only) {
  237. int offset;
  238. /* scale factors table for layer 1/2 */
  239. for(i=0;i<64;i++) {
  240. int shift, mod;
  241. /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
  242. shift = (i / 3);
  243. mod = i % 3;
  244. scale_factor_modshift[i] = mod | (shift << 2);
  245. }
  246. /* scale factor multiply for layer 1 */
  247. for(i=0;i<15;i++) {
  248. int n, norm;
  249. n = i + 2;
  250. norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
  251. scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
  252. scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
  253. scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
  254. av_dlog(avctx, "%d: norm=%x s=%x %x %x\n",
  255. i, norm,
  256. scale_factor_mult[i][0],
  257. scale_factor_mult[i][1],
  258. scale_factor_mult[i][2]);
  259. }
  260. RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));
  261. /* huffman decode tables */
  262. offset = 0;
  263. for(i=1;i<16;i++) {
  264. const HuffTable *h = &mpa_huff_tables[i];
  265. int xsize, x, y;
  266. uint8_t tmp_bits [512];
  267. uint16_t tmp_codes[512];
  268. memset(tmp_bits , 0, sizeof(tmp_bits ));
  269. memset(tmp_codes, 0, sizeof(tmp_codes));
  270. xsize = h->xsize;
  271. j = 0;
  272. for(x=0;x<xsize;x++) {
  273. for(y=0;y<xsize;y++){
  274. tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ];
  275. tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];
  276. }
  277. }
  278. /* XXX: fail test */
  279. huff_vlc[i].table = huff_vlc_tables+offset;
  280. huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
  281. init_vlc(&huff_vlc[i], 7, 512,
  282. tmp_bits, 1, 1, tmp_codes, 2, 2,
  283. INIT_VLC_USE_NEW_STATIC);
  284. offset += huff_vlc_tables_sizes[i];
  285. }
  286. assert(offset == FF_ARRAY_ELEMS(huff_vlc_tables));
  287. offset = 0;
  288. for(i=0;i<2;i++) {
  289. huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
  290. huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
  291. init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
  292. mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
  293. INIT_VLC_USE_NEW_STATIC);
  294. offset += huff_quad_vlc_tables_sizes[i];
  295. }
  296. assert(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));
  297. for(i=0;i<9;i++) {
  298. k = 0;
  299. for(j=0;j<22;j++) {
  300. band_index_long[i][j] = k;
  301. k += band_size_long[i][j];
  302. }
  303. band_index_long[i][22] = k;
  304. }
  305. /* compute n ^ (4/3) and store it in mantissa/exp format */
  306. int_pow_init();
  307. mpegaudio_tableinit();
  308. for (i = 0; i < 4; i++)
  309. if (ff_mpa_quant_bits[i] < 0)
  310. for (j = 0; j < (1<<(-ff_mpa_quant_bits[i]+1)); j++) {
  311. int val1, val2, val3, steps;
  312. int val = j;
  313. steps = ff_mpa_quant_steps[i];
  314. val1 = val % steps;
  315. val /= steps;
  316. val2 = val % steps;
  317. val3 = val / steps;
  318. division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);
  319. }
  320. for(i=0;i<7;i++) {
  321. float f;
  322. INTFLOAT v;
  323. if (i != 6) {
  324. f = tan((double)i * M_PI / 12.0);
  325. v = FIXR(f / (1.0 + f));
  326. } else {
  327. v = FIXR(1.0);
  328. }
  329. is_table[0][i] = v;
  330. is_table[1][6 - i] = v;
  331. }
  332. /* invalid values */
  333. for(i=7;i<16;i++)
  334. is_table[0][i] = is_table[1][i] = 0.0;
  335. for(i=0;i<16;i++) {
  336. double f;
  337. int e, k;
  338. for(j=0;j<2;j++) {
  339. e = -(j + 1) * ((i + 1) >> 1);
  340. f = pow(2.0, e / 4.0);
  341. k = i & 1;
  342. is_table_lsf[j][k ^ 1][i] = FIXR(f);
  343. is_table_lsf[j][k][i] = FIXR(1.0);
  344. av_dlog(avctx, "is_table_lsf %d %d: %x %x\n",
  345. i, j, is_table_lsf[j][0][i], is_table_lsf[j][1][i]);
  346. }
  347. }
  348. for(i=0;i<8;i++) {
  349. float ci, cs, ca;
  350. ci = ci_table[i];
  351. cs = 1.0 / sqrt(1.0 + ci * ci);
  352. ca = cs * ci;
  353. csa_table[i][0] = FIXHR(cs/4);
  354. csa_table[i][1] = FIXHR(ca/4);
  355. csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);
  356. csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);
  357. csa_table_float[i][0] = cs;
  358. csa_table_float[i][1] = ca;
  359. csa_table_float[i][2] = ca + cs;
  360. csa_table_float[i][3] = ca - cs;
  361. }
  362. /* compute mdct windows */
  363. for(i=0;i<36;i++) {
  364. for(j=0; j<4; j++){
  365. double d;
  366. if(j==2 && i%3 != 1)
  367. continue;
  368. d= sin(M_PI * (i + 0.5) / 36.0);
  369. if(j==1){
  370. if (i>=30) d= 0;
  371. else if(i>=24) d= sin(M_PI * (i - 18 + 0.5) / 12.0);
  372. else if(i>=18) d= 1;
  373. }else if(j==3){
  374. if (i< 6) d= 0;
  375. else if(i< 12) d= sin(M_PI * (i - 6 + 0.5) / 12.0);
  376. else if(i< 18) d= 1;
  377. }
  378. //merge last stage of imdct into the window coefficients
  379. d*= 0.5 / cos(M_PI*(2*i + 19)/72);
  380. if(j==2)
  381. mdct_win[j][i/3] = FIXHR((d / (1<<5)));
  382. else
  383. mdct_win[j][i ] = FIXHR((d / (1<<5)));
  384. }
  385. }
  386. /* NOTE: we do frequency inversion adter the MDCT by changing
  387. the sign of the right window coefs */
  388. for(j=0;j<4;j++) {
  389. for(i=0;i<36;i+=2) {
  390. mdct_win[j + 4][i] = mdct_win[j][i];
  391. mdct_win[j + 4][i + 1] = -mdct_win[j][i + 1];
  392. }
  393. }
  394. init = 1;
  395. }
  396. if (avctx->codec_id == CODEC_ID_MP3ADU)
  397. s->adu_mode = 1;
  398. return 0;
  399. }
  400. #if CONFIG_FLOAT
  401. static inline float round_sample(float *sum)
  402. {
  403. float sum1=*sum;
  404. *sum = 0;
  405. return sum1;
  406. }
  407. /* signed 16x16 -> 32 multiply add accumulate */
  408. #define MACS(rt, ra, rb) rt+=(ra)*(rb)
  409. /* signed 16x16 -> 32 multiply */
  410. #define MULS(ra, rb) ((ra)*(rb))
  411. #define MLSS(rt, ra, rb) rt-=(ra)*(rb)
  412. #else
  413. static inline int round_sample(int64_t *sum)
  414. {
  415. int sum1;
  416. sum1 = (int)((*sum) >> OUT_SHIFT);
  417. *sum &= (1<<OUT_SHIFT)-1;
  418. return av_clip_int16(sum1);
  419. }
  420. # define MULS(ra, rb) MUL64(ra, rb)
  421. # define MACS(rt, ra, rb) MAC64(rt, ra, rb)
  422. # define MLSS(rt, ra, rb) MLS64(rt, ra, rb)
  423. #endif
  424. #define SUM8(op, sum, w, p) \
  425. { \
  426. op(sum, (w)[0 * 64], (p)[0 * 64]); \
  427. op(sum, (w)[1 * 64], (p)[1 * 64]); \
  428. op(sum, (w)[2 * 64], (p)[2 * 64]); \
  429. op(sum, (w)[3 * 64], (p)[3 * 64]); \
  430. op(sum, (w)[4 * 64], (p)[4 * 64]); \
  431. op(sum, (w)[5 * 64], (p)[5 * 64]); \
  432. op(sum, (w)[6 * 64], (p)[6 * 64]); \
  433. op(sum, (w)[7 * 64], (p)[7 * 64]); \
  434. }
  435. #define SUM8P2(sum1, op1, sum2, op2, w1, w2, p) \
  436. { \
  437. INTFLOAT tmp;\
  438. tmp = p[0 * 64];\
  439. op1(sum1, (w1)[0 * 64], tmp);\
  440. op2(sum2, (w2)[0 * 64], tmp);\
  441. tmp = p[1 * 64];\
  442. op1(sum1, (w1)[1 * 64], tmp);\
  443. op2(sum2, (w2)[1 * 64], tmp);\
  444. tmp = p[2 * 64];\
  445. op1(sum1, (w1)[2 * 64], tmp);\
  446. op2(sum2, (w2)[2 * 64], tmp);\
  447. tmp = p[3 * 64];\
  448. op1(sum1, (w1)[3 * 64], tmp);\
  449. op2(sum2, (w2)[3 * 64], tmp);\
  450. tmp = p[4 * 64];\
  451. op1(sum1, (w1)[4 * 64], tmp);\
  452. op2(sum2, (w2)[4 * 64], tmp);\
  453. tmp = p[5 * 64];\
  454. op1(sum1, (w1)[5 * 64], tmp);\
  455. op2(sum2, (w2)[5 * 64], tmp);\
  456. tmp = p[6 * 64];\
  457. op1(sum1, (w1)[6 * 64], tmp);\
  458. op2(sum2, (w2)[6 * 64], tmp);\
  459. tmp = p[7 * 64];\
  460. op1(sum1, (w1)[7 * 64], tmp);\
  461. op2(sum2, (w2)[7 * 64], tmp);\
  462. }
  463. void av_cold RENAME(ff_mpa_synth_init)(MPA_INT *window)
  464. {
  465. int i, j;
  466. /* max = 18760, max sum over all 16 coefs : 44736 */
  467. for(i=0;i<257;i++) {
  468. INTFLOAT v;
  469. v = ff_mpa_enwindow[i];
  470. #if CONFIG_FLOAT
  471. v *= 1.0 / (1LL<<(16 + FRAC_BITS));
  472. #endif
  473. window[i] = v;
  474. if ((i & 63) != 0)
  475. v = -v;
  476. if (i != 0)
  477. window[512 - i] = v;
  478. }
  479. // Needed for avoiding shuffles in ASM implementations
  480. for(i=0; i < 8; i++)
  481. for(j=0; j < 16; j++)
  482. window[512+16*i+j] = window[64*i+32-j];
  483. for(i=0; i < 8; i++)
  484. for(j=0; j < 16; j++)
  485. window[512+128+16*i+j] = window[64*i+48-j];
  486. }
  487. static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
  488. int *dither_state, OUT_INT *samples, int incr)
  489. {
  490. register const MPA_INT *w, *w2, *p;
  491. int j;
  492. OUT_INT *samples2;
  493. #if CONFIG_FLOAT
  494. float sum, sum2;
  495. #else
  496. int64_t sum, sum2;
  497. #endif
  498. /* copy to avoid wrap */
  499. memcpy(synth_buf + 512, synth_buf, 32 * sizeof(*synth_buf));
  500. samples2 = samples + 31 * incr;
  501. w = window;
  502. w2 = window + 31;
  503. sum = *dither_state;
  504. p = synth_buf + 16;
  505. SUM8(MACS, sum, w, p);
  506. p = synth_buf + 48;
  507. SUM8(MLSS, sum, w + 32, p);
  508. *samples = round_sample(&sum);
  509. samples += incr;
  510. w++;
  511. /* we calculate two samples at the same time to avoid one memory
  512. access per two sample */
  513. for(j=1;j<16;j++) {
  514. sum2 = 0;
  515. p = synth_buf + 16 + j;
  516. SUM8P2(sum, MACS, sum2, MLSS, w, w2, p);
  517. p = synth_buf + 48 - j;
  518. SUM8P2(sum, MLSS, sum2, MLSS, w + 32, w2 + 32, p);
  519. *samples = round_sample(&sum);
  520. samples += incr;
  521. sum += sum2;
  522. *samples2 = round_sample(&sum);
  523. samples2 -= incr;
  524. w++;
  525. w2--;
  526. }
  527. p = synth_buf + 32;
  528. SUM8(MLSS, sum, w + 32, p);
  529. *samples = round_sample(&sum);
  530. *dither_state= sum;
  531. }
  532. /* 32 sub band synthesis filter. Input: 32 sub band samples, Output:
  533. 32 samples. */
  534. /* XXX: optimize by avoiding ring buffer usage */
  535. #if !CONFIG_FLOAT
  536. void ff_mpa_synth_filter_fixed(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
  537. MPA_INT *window, int *dither_state,
  538. OUT_INT *samples, int incr,
  539. INTFLOAT sb_samples[SBLIMIT])
  540. {
  541. register MPA_INT *synth_buf;
  542. int offset;
  543. offset = *synth_buf_offset;
  544. synth_buf = synth_buf_ptr + offset;
  545. ff_dct32_fixed(synth_buf, sb_samples);
  546. apply_window_mp3_c(synth_buf, window, dither_state, samples, incr);
  547. offset = (offset - 32) & 511;
  548. *synth_buf_offset = offset;
  549. }
  550. #endif
  551. #define C3 FIXHR(0.86602540378443864676/2)
  552. /* 0.5 / cos(pi*(2*i+1)/36) */
  553. static const INTFLOAT icos36[9] = {
  554. FIXR(0.50190991877167369479),
  555. FIXR(0.51763809020504152469), //0
  556. FIXR(0.55168895948124587824),
  557. FIXR(0.61038729438072803416),
  558. FIXR(0.70710678118654752439), //1
  559. FIXR(0.87172339781054900991),
  560. FIXR(1.18310079157624925896),
  561. FIXR(1.93185165257813657349), //2
  562. FIXR(5.73685662283492756461),
  563. };
  564. /* 0.5 / cos(pi*(2*i+1)/36) */
  565. static const INTFLOAT icos36h[9] = {
  566. FIXHR(0.50190991877167369479/2),
  567. FIXHR(0.51763809020504152469/2), //0
  568. FIXHR(0.55168895948124587824/2),
  569. FIXHR(0.61038729438072803416/2),
  570. FIXHR(0.70710678118654752439/2), //1
  571. FIXHR(0.87172339781054900991/2),
  572. FIXHR(1.18310079157624925896/4),
  573. FIXHR(1.93185165257813657349/4), //2
  574. // FIXHR(5.73685662283492756461),
  575. };
  576. /* 12 points IMDCT. We compute it "by hand" by factorizing obvious
  577. cases. */
  578. static void imdct12(INTFLOAT *out, INTFLOAT *in)
  579. {
  580. INTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
  581. in0= in[0*3];
  582. in1= in[1*3] + in[0*3];
  583. in2= in[2*3] + in[1*3];
  584. in3= in[3*3] + in[2*3];
  585. in4= in[4*3] + in[3*3];
  586. in5= in[5*3] + in[4*3];
  587. in5 += in3;
  588. in3 += in1;
  589. in2= MULH3(in2, C3, 2);
  590. in3= MULH3(in3, C3, 4);
  591. t1 = in0 - in4;
  592. t2 = MULH3(in1 - in5, icos36h[4], 2);
  593. out[ 7]=
  594. out[10]= t1 + t2;
  595. out[ 1]=
  596. out[ 4]= t1 - t2;
  597. in0 += SHR(in4, 1);
  598. in4 = in0 + in2;
  599. in5 += 2*in1;
  600. in1 = MULH3(in5 + in3, icos36h[1], 1);
  601. out[ 8]=
  602. out[ 9]= in4 + in1;
  603. out[ 2]=
  604. out[ 3]= in4 - in1;
  605. in0 -= in2;
  606. in5 = MULH3(in5 - in3, icos36h[7], 2);
  607. out[ 0]=
  608. out[ 5]= in0 - in5;
  609. out[ 6]=
  610. out[11]= in0 + in5;
  611. }
  612. /* cos(pi*i/18) */
  613. #define C1 FIXHR(0.98480775301220805936/2)
  614. #define C2 FIXHR(0.93969262078590838405/2)
  615. #define C3 FIXHR(0.86602540378443864676/2)
  616. #define C4 FIXHR(0.76604444311897803520/2)
  617. #define C5 FIXHR(0.64278760968653932632/2)
  618. #define C6 FIXHR(0.5/2)
  619. #define C7 FIXHR(0.34202014332566873304/2)
  620. #define C8 FIXHR(0.17364817766693034885/2)
  621. /* using Lee like decomposition followed by hand coded 9 points DCT */
  622. static void imdct36(INTFLOAT *out, INTFLOAT *buf, INTFLOAT *in, INTFLOAT *win)
  623. {
  624. int i, j;
  625. INTFLOAT t0, t1, t2, t3, s0, s1, s2, s3;
  626. INTFLOAT tmp[18], *tmp1, *in1;
  627. for(i=17;i>=1;i--)
  628. in[i] += in[i-1];
  629. for(i=17;i>=3;i-=2)
  630. in[i] += in[i-2];
  631. for(j=0;j<2;j++) {
  632. tmp1 = tmp + j;
  633. in1 = in + j;
  634. t2 = in1[2*4] + in1[2*8] - in1[2*2];
  635. t3 = in1[2*0] + SHR(in1[2*6],1);
  636. t1 = in1[2*0] - in1[2*6];
  637. tmp1[ 6] = t1 - SHR(t2,1);
  638. tmp1[16] = t1 + t2;
  639. t0 = MULH3(in1[2*2] + in1[2*4] , C2, 2);
  640. t1 = MULH3(in1[2*4] - in1[2*8] , -2*C8, 1);
  641. t2 = MULH3(in1[2*2] + in1[2*8] , -C4, 2);
  642. tmp1[10] = t3 - t0 - t2;
  643. tmp1[ 2] = t3 + t0 + t1;
  644. tmp1[14] = t3 + t2 - t1;
  645. tmp1[ 4] = MULH3(in1[2*5] + in1[2*7] - in1[2*1], -C3, 2);
  646. t2 = MULH3(in1[2*1] + in1[2*5], C1, 2);
  647. t3 = MULH3(in1[2*5] - in1[2*7], -2*C7, 1);
  648. t0 = MULH3(in1[2*3], C3, 2);
  649. t1 = MULH3(in1[2*1] + in1[2*7], -C5, 2);
  650. tmp1[ 0] = t2 + t3 + t0;
  651. tmp1[12] = t2 + t1 - t0;
  652. tmp1[ 8] = t3 - t1 - t0;
  653. }
  654. i = 0;
  655. for(j=0;j<4;j++) {
  656. t0 = tmp[i];
  657. t1 = tmp[i + 2];
  658. s0 = t1 + t0;
  659. s2 = t1 - t0;
  660. t2 = tmp[i + 1];
  661. t3 = tmp[i + 3];
  662. s1 = MULH3(t3 + t2, icos36h[j], 2);
  663. s3 = MULLx(t3 - t2, icos36[8 - j], FRAC_BITS);
  664. t0 = s0 + s1;
  665. t1 = s0 - s1;
  666. out[(9 + j)*SBLIMIT] = MULH3(t1, win[9 + j], 1) + buf[9 + j];
  667. out[(8 - j)*SBLIMIT] = MULH3(t1, win[8 - j], 1) + buf[8 - j];
  668. buf[9 + j] = MULH3(t0, win[18 + 9 + j], 1);
  669. buf[8 - j] = MULH3(t0, win[18 + 8 - j], 1);
  670. t0 = s2 + s3;
  671. t1 = s2 - s3;
  672. out[(9 + 8 - j)*SBLIMIT] = MULH3(t1, win[9 + 8 - j], 1) + buf[9 + 8 - j];
  673. out[( j)*SBLIMIT] = MULH3(t1, win[ j], 1) + buf[ j];
  674. buf[9 + 8 - j] = MULH3(t0, win[18 + 9 + 8 - j], 1);
  675. buf[ + j] = MULH3(t0, win[18 + j], 1);
  676. i += 4;
  677. }
  678. s0 = tmp[16];
  679. s1 = MULH3(tmp[17], icos36h[4], 2);
  680. t0 = s0 + s1;
  681. t1 = s0 - s1;
  682. out[(9 + 4)*SBLIMIT] = MULH3(t1, win[9 + 4], 1) + buf[9 + 4];
  683. out[(8 - 4)*SBLIMIT] = MULH3(t1, win[8 - 4], 1) + buf[8 - 4];
  684. buf[9 + 4] = MULH3(t0, win[18 + 9 + 4], 1);
  685. buf[8 - 4] = MULH3(t0, win[18 + 8 - 4], 1);
  686. }
  687. /* return the number of decoded frames */
  688. static int mp_decode_layer1(MPADecodeContext *s)
  689. {
  690. int bound, i, v, n, ch, j, mant;
  691. uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
  692. uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
  693. if (s->mode == MPA_JSTEREO)
  694. bound = (s->mode_ext + 1) * 4;
  695. else
  696. bound = SBLIMIT;
  697. /* allocation bits */
  698. for(i=0;i<bound;i++) {
  699. for(ch=0;ch<s->nb_channels;ch++) {
  700. allocation[ch][i] = get_bits(&s->gb, 4);
  701. }
  702. }
  703. for(i=bound;i<SBLIMIT;i++) {
  704. allocation[0][i] = get_bits(&s->gb, 4);
  705. }
  706. /* scale factors */
  707. for(i=0;i<bound;i++) {
  708. for(ch=0;ch<s->nb_channels;ch++) {
  709. if (allocation[ch][i])
  710. scale_factors[ch][i] = get_bits(&s->gb, 6);
  711. }
  712. }
  713. for(i=bound;i<SBLIMIT;i++) {
  714. if (allocation[0][i]) {
  715. scale_factors[0][i] = get_bits(&s->gb, 6);
  716. scale_factors[1][i] = get_bits(&s->gb, 6);
  717. }
  718. }
  719. /* compute samples */
  720. for(j=0;j<12;j++) {
  721. for(i=0;i<bound;i++) {
  722. for(ch=0;ch<s->nb_channels;ch++) {
  723. n = allocation[ch][i];
  724. if (n) {
  725. mant = get_bits(&s->gb, n + 1);
  726. v = l1_unscale(n, mant, scale_factors[ch][i]);
  727. } else {
  728. v = 0;
  729. }
  730. s->sb_samples[ch][j][i] = v;
  731. }
  732. }
  733. for(i=bound;i<SBLIMIT;i++) {
  734. n = allocation[0][i];
  735. if (n) {
  736. mant = get_bits(&s->gb, n + 1);
  737. v = l1_unscale(n, mant, scale_factors[0][i]);
  738. s->sb_samples[0][j][i] = v;
  739. v = l1_unscale(n, mant, scale_factors[1][i]);
  740. s->sb_samples[1][j][i] = v;
  741. } else {
  742. s->sb_samples[0][j][i] = 0;
  743. s->sb_samples[1][j][i] = 0;
  744. }
  745. }
  746. }
  747. return 12;
  748. }
  749. static int mp_decode_layer2(MPADecodeContext *s)
  750. {
  751. int sblimit; /* number of used subbands */
  752. const unsigned char *alloc_table;
  753. int table, bit_alloc_bits, i, j, ch, bound, v;
  754. unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
  755. unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
  756. unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
  757. int scale, qindex, bits, steps, k, l, m, b;
  758. /* select decoding table */
  759. table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
  760. s->sample_rate, s->lsf);
  761. sblimit = ff_mpa_sblimit_table[table];
  762. alloc_table = ff_mpa_alloc_tables[table];
  763. if (s->mode == MPA_JSTEREO)
  764. bound = (s->mode_ext + 1) * 4;
  765. else
  766. bound = sblimit;
  767. av_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
  768. /* sanity check */
  769. if( bound > sblimit ) bound = sblimit;
  770. /* parse bit allocation */
  771. j = 0;
  772. for(i=0;i<bound;i++) {
  773. bit_alloc_bits = alloc_table[j];
  774. for(ch=0;ch<s->nb_channels;ch++) {
  775. bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
  776. }
  777. j += 1 << bit_alloc_bits;
  778. }
  779. for(i=bound;i<sblimit;i++) {
  780. bit_alloc_bits = alloc_table[j];
  781. v = get_bits(&s->gb, bit_alloc_bits);
  782. bit_alloc[0][i] = v;
  783. bit_alloc[1][i] = v;
  784. j += 1 << bit_alloc_bits;
  785. }
  786. /* scale codes */
  787. for(i=0;i<sblimit;i++) {
  788. for(ch=0;ch<s->nb_channels;ch++) {
  789. if (bit_alloc[ch][i])
  790. scale_code[ch][i] = get_bits(&s->gb, 2);
  791. }
  792. }
  793. /* scale factors */
  794. for(i=0;i<sblimit;i++) {
  795. for(ch=0;ch<s->nb_channels;ch++) {
  796. if (bit_alloc[ch][i]) {
  797. sf = scale_factors[ch][i];
  798. switch(scale_code[ch][i]) {
  799. default:
  800. case 0:
  801. sf[0] = get_bits(&s->gb, 6);
  802. sf[1] = get_bits(&s->gb, 6);
  803. sf[2] = get_bits(&s->gb, 6);
  804. break;
  805. case 2:
  806. sf[0] = get_bits(&s->gb, 6);
  807. sf[1] = sf[0];
  808. sf[2] = sf[0];
  809. break;
  810. case 1:
  811. sf[0] = get_bits(&s->gb, 6);
  812. sf[2] = get_bits(&s->gb, 6);
  813. sf[1] = sf[0];
  814. break;
  815. case 3:
  816. sf[0] = get_bits(&s->gb, 6);
  817. sf[2] = get_bits(&s->gb, 6);
  818. sf[1] = sf[2];
  819. break;
  820. }
  821. }
  822. }
  823. }
  824. /* samples */
  825. for(k=0;k<3;k++) {
  826. for(l=0;l<12;l+=3) {
  827. j = 0;
  828. for(i=0;i<bound;i++) {
  829. bit_alloc_bits = alloc_table[j];
  830. for(ch=0;ch<s->nb_channels;ch++) {
  831. b = bit_alloc[ch][i];
  832. if (b) {
  833. scale = scale_factors[ch][i][k];
  834. qindex = alloc_table[j+b];
  835. bits = ff_mpa_quant_bits[qindex];
  836. if (bits < 0) {
  837. int v2;
  838. /* 3 values at the same time */
  839. v = get_bits(&s->gb, -bits);
  840. v2 = division_tabs[qindex][v];
  841. steps = ff_mpa_quant_steps[qindex];
  842. s->sb_samples[ch][k * 12 + l + 0][i] =
  843. l2_unscale_group(steps, v2 & 15, scale);
  844. s->sb_samples[ch][k * 12 + l + 1][i] =
  845. l2_unscale_group(steps, (v2 >> 4) & 15, scale);
  846. s->sb_samples[ch][k * 12 + l + 2][i] =
  847. l2_unscale_group(steps, v2 >> 8 , scale);
  848. } else {
  849. for(m=0;m<3;m++) {
  850. v = get_bits(&s->gb, bits);
  851. v = l1_unscale(bits - 1, v, scale);
  852. s->sb_samples[ch][k * 12 + l + m][i] = v;
  853. }
  854. }
  855. } else {
  856. s->sb_samples[ch][k * 12 + l + 0][i] = 0;
  857. s->sb_samples[ch][k * 12 + l + 1][i] = 0;
  858. s->sb_samples[ch][k * 12 + l + 2][i] = 0;
  859. }
  860. }
  861. /* next subband in alloc table */
  862. j += 1 << bit_alloc_bits;
  863. }
  864. /* XXX: find a way to avoid this duplication of code */
  865. for(i=bound;i<sblimit;i++) {
  866. bit_alloc_bits = alloc_table[j];
  867. b = bit_alloc[0][i];
  868. if (b) {
  869. int mant, scale0, scale1;
  870. scale0 = scale_factors[0][i][k];
  871. scale1 = scale_factors[1][i][k];
  872. qindex = alloc_table[j+b];
  873. bits = ff_mpa_quant_bits[qindex];
  874. if (bits < 0) {
  875. /* 3 values at the same time */
  876. v = get_bits(&s->gb, -bits);
  877. steps = ff_mpa_quant_steps[qindex];
  878. mant = v % steps;
  879. v = v / steps;
  880. s->sb_samples[0][k * 12 + l + 0][i] =
  881. l2_unscale_group(steps, mant, scale0);
  882. s->sb_samples[1][k * 12 + l + 0][i] =
  883. l2_unscale_group(steps, mant, scale1);
  884. mant = v % steps;
  885. v = v / steps;
  886. s->sb_samples[0][k * 12 + l + 1][i] =
  887. l2_unscale_group(steps, mant, scale0);
  888. s->sb_samples[1][k * 12 + l + 1][i] =
  889. l2_unscale_group(steps, mant, scale1);
  890. s->sb_samples[0][k * 12 + l + 2][i] =
  891. l2_unscale_group(steps, v, scale0);
  892. s->sb_samples[1][k * 12 + l + 2][i] =
  893. l2_unscale_group(steps, v, scale1);
  894. } else {
  895. for(m=0;m<3;m++) {
  896. mant = get_bits(&s->gb, bits);
  897. s->sb_samples[0][k * 12 + l + m][i] =
  898. l1_unscale(bits - 1, mant, scale0);
  899. s->sb_samples[1][k * 12 + l + m][i] =
  900. l1_unscale(bits - 1, mant, scale1);
  901. }
  902. }
  903. } else {
  904. s->sb_samples[0][k * 12 + l + 0][i] = 0;
  905. s->sb_samples[0][k * 12 + l + 1][i] = 0;
  906. s->sb_samples[0][k * 12 + l + 2][i] = 0;
  907. s->sb_samples[1][k * 12 + l + 0][i] = 0;
  908. s->sb_samples[1][k * 12 + l + 1][i] = 0;
  909. s->sb_samples[1][k * 12 + l + 2][i] = 0;
  910. }
  911. /* next subband in alloc table */
  912. j += 1 << bit_alloc_bits;
  913. }
  914. /* fill remaining samples to zero */
  915. for(i=sblimit;i<SBLIMIT;i++) {
  916. for(ch=0;ch<s->nb_channels;ch++) {
  917. s->sb_samples[ch][k * 12 + l + 0][i] = 0;
  918. s->sb_samples[ch][k * 12 + l + 1][i] = 0;
  919. s->sb_samples[ch][k * 12 + l + 2][i] = 0;
  920. }
  921. }
  922. }
  923. }
  924. return 3 * 12;
  925. }
  926. #define SPLIT(dst,sf,n)\
  927. if(n==3){\
  928. int m= (sf*171)>>9;\
  929. dst= sf - 3*m;\
  930. sf=m;\
  931. }else if(n==4){\
  932. dst= sf&3;\
  933. sf>>=2;\
  934. }else if(n==5){\
  935. int m= (sf*205)>>10;\
  936. dst= sf - 5*m;\
  937. sf=m;\
  938. }else if(n==6){\
  939. int m= (sf*171)>>10;\
  940. dst= sf - 6*m;\
  941. sf=m;\
  942. }else{\
  943. dst=0;\
  944. }
  945. static av_always_inline void lsf_sf_expand(int *slen,
  946. int sf, int n1, int n2, int n3)
  947. {
  948. SPLIT(slen[3], sf, n3)
  949. SPLIT(slen[2], sf, n2)
  950. SPLIT(slen[1], sf, n1)
  951. slen[0] = sf;
  952. }
  953. static void exponents_from_scale_factors(MPADecodeContext *s,
  954. GranuleDef *g,
  955. int16_t *exponents)
  956. {
  957. const uint8_t *bstab, *pretab;
  958. int len, i, j, k, l, v0, shift, gain, gains[3];
  959. int16_t *exp_ptr;
  960. exp_ptr = exponents;
  961. gain = g->global_gain - 210;
  962. shift = g->scalefac_scale + 1;
  963. bstab = band_size_long[s->sample_rate_index];
  964. pretab = mpa_pretab[g->preflag];
  965. for(i=0;i<g->long_end;i++) {
  966. v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
  967. len = bstab[i];
  968. for(j=len;j>0;j--)
  969. *exp_ptr++ = v0;
  970. }
  971. if (g->short_start < 13) {
  972. bstab = band_size_short[s->sample_rate_index];
  973. gains[0] = gain - (g->subblock_gain[0] << 3);
  974. gains[1] = gain - (g->subblock_gain[1] << 3);
  975. gains[2] = gain - (g->subblock_gain[2] << 3);
  976. k = g->long_end;
  977. for(i=g->short_start;i<13;i++) {
  978. len = bstab[i];
  979. for(l=0;l<3;l++) {
  980. v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
  981. for(j=len;j>0;j--)
  982. *exp_ptr++ = v0;
  983. }
  984. }
  985. }
  986. }
  987. /* handle n = 0 too */
  988. static inline int get_bitsz(GetBitContext *s, int n)
  989. {
  990. if (n == 0)
  991. return 0;
  992. else
  993. return get_bits(s, n);
  994. }
  995. static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos, int *end_pos2){
  996. if(s->in_gb.buffer && *pos >= s->gb.size_in_bits){
  997. s->gb= s->in_gb;
  998. s->in_gb.buffer=NULL;
  999. assert((get_bits_count(&s->gb) & 7) == 0);
  1000. skip_bits_long(&s->gb, *pos - *end_pos);
  1001. *end_pos2=
  1002. *end_pos= *end_pos2 + get_bits_count(&s->gb) - *pos;
  1003. *pos= get_bits_count(&s->gb);
  1004. }
  1005. }
  1006. /* Following is a optimized code for
  1007. INTFLOAT v = *src
  1008. if(get_bits1(&s->gb))
  1009. v = -v;
  1010. *dst = v;
  1011. */
  1012. #if CONFIG_FLOAT
  1013. #define READ_FLIP_SIGN(dst,src)\
  1014. v = AV_RN32A(src) ^ (get_bits1(&s->gb)<<31);\
  1015. AV_WN32A(dst, v);
  1016. #else
  1017. #define READ_FLIP_SIGN(dst,src)\
  1018. v= -get_bits1(&s->gb);\
  1019. *(dst) = (*(src) ^ v) - v;
  1020. #endif
  1021. static int huffman_decode(MPADecodeContext *s, GranuleDef *g,
  1022. int16_t *exponents, int end_pos2)
  1023. {
  1024. int s_index;
  1025. int i;
  1026. int last_pos, bits_left;
  1027. VLC *vlc;
  1028. int end_pos= FFMIN(end_pos2, s->gb.size_in_bits);
  1029. /* low frequencies (called big values) */
  1030. s_index = 0;
  1031. for(i=0;i<3;i++) {
  1032. int j, k, l, linbits;
  1033. j = g->region_size[i];
  1034. if (j == 0)
  1035. continue;
  1036. /* select vlc table */
  1037. k = g->table_select[i];
  1038. l = mpa_huff_data[k][0];
  1039. linbits = mpa_huff_data[k][1];
  1040. vlc = &huff_vlc[l];
  1041. if(!l){
  1042. memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid)*2*j);
  1043. s_index += 2*j;
  1044. continue;
  1045. }
  1046. /* read huffcode and compute each couple */
  1047. for(;j>0;j--) {
  1048. int exponent, x, y;
  1049. int v;
  1050. int pos= get_bits_count(&s->gb);
  1051. if (pos >= end_pos){
  1052. // av_log(NULL, AV_LOG_ERROR, "pos: %d %d %d %d\n", pos, end_pos, end_pos2, s_index);
  1053. switch_buffer(s, &pos, &end_pos, &end_pos2);
  1054. // av_log(NULL, AV_LOG_ERROR, "new pos: %d %d\n", pos, end_pos);
  1055. if(pos >= end_pos)
  1056. break;
  1057. }
  1058. y = get_vlc2(&s->gb, vlc->table, 7, 3);
  1059. if(!y){
  1060. g->sb_hybrid[s_index ] =
  1061. g->sb_hybrid[s_index+1] = 0;
  1062. s_index += 2;
  1063. continue;
  1064. }
  1065. exponent= exponents[s_index];
  1066. av_dlog(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n",
  1067. i, g->region_size[i] - j, x, y, exponent);
  1068. if(y&16){
  1069. x = y >> 5;
  1070. y = y & 0x0f;
  1071. if (x < 15){
  1072. READ_FLIP_SIGN(g->sb_hybrid+s_index, RENAME(expval_table)[ exponent ]+x)
  1073. }else{
  1074. x += get_bitsz(&s->gb, linbits);
  1075. v = l3_unscale(x, exponent);
  1076. if (get_bits1(&s->gb))
  1077. v = -v;
  1078. g->sb_hybrid[s_index] = v;
  1079. }
  1080. if (y < 15){
  1081. READ_FLIP_SIGN(g->sb_hybrid+s_index+1, RENAME(expval_table)[ exponent ]+y)
  1082. }else{
  1083. y += get_bitsz(&s->gb, linbits);
  1084. v = l3_unscale(y, exponent);
  1085. if (get_bits1(&s->gb))
  1086. v = -v;
  1087. g->sb_hybrid[s_index+1] = v;
  1088. }
  1089. }else{
  1090. x = y >> 5;
  1091. y = y & 0x0f;
  1092. x += y;
  1093. if (x < 15){
  1094. READ_FLIP_SIGN(g->sb_hybrid+s_index+!!y, RENAME(expval_table)[ exponent ]+x)
  1095. }else{
  1096. x += get_bitsz(&s->gb, linbits);
  1097. v = l3_unscale(x, exponent);
  1098. if (get_bits1(&s->gb))
  1099. v = -v;
  1100. g->sb_hybrid[s_index+!!y] = v;
  1101. }
  1102. g->sb_hybrid[s_index+ !y] = 0;
  1103. }
  1104. s_index+=2;
  1105. }
  1106. }
  1107. /* high frequencies */
  1108. vlc = &huff_quad_vlc[g->count1table_select];
  1109. last_pos=0;
  1110. while (s_index <= 572) {
  1111. int pos, code;
  1112. pos = get_bits_count(&s->gb);
  1113. if (pos >= end_pos) {
  1114. if (pos > end_pos2 && last_pos){
  1115. /* some encoders generate an incorrect size for this
  1116. part. We must go back into the data */
  1117. s_index -= 4;
  1118. skip_bits_long(&s->gb, last_pos - pos);
  1119. av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
  1120. if(s->error_recognition >= FF_ER_COMPLIANT)
  1121. s_index=0;
  1122. break;
  1123. }
  1124. // av_log(NULL, AV_LOG_ERROR, "pos2: %d %d %d %d\n", pos, end_pos, end_pos2, s_index);
  1125. switch_buffer(s, &pos, &end_pos, &end_pos2);
  1126. // av_log(NULL, AV_LOG_ERROR, "new pos2: %d %d %d\n", pos, end_pos, s_index);
  1127. if(pos >= end_pos)
  1128. break;
  1129. }
  1130. last_pos= pos;
  1131. code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
  1132. av_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
  1133. g->sb_hybrid[s_index+0]=
  1134. g->sb_hybrid[s_index+1]=
  1135. g->sb_hybrid[s_index+2]=
  1136. g->sb_hybrid[s_index+3]= 0;
  1137. while(code){
  1138. static const int idxtab[16]={3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0};
  1139. int v;
  1140. int pos= s_index+idxtab[code];
  1141. code ^= 8>>idxtab[code];
  1142. READ_FLIP_SIGN(g->sb_hybrid+pos, RENAME(exp_table)+exponents[pos])
  1143. }
  1144. s_index+=4;
  1145. }
  1146. /* skip extension bits */
  1147. bits_left = end_pos2 - get_bits_count(&s->gb);
  1148. //av_log(NULL, AV_LOG_ERROR, "left:%d buf:%p\n", bits_left, s->in_gb.buffer);
  1149. if (bits_left < 0 && s->error_recognition >= FF_ER_COMPLIANT) {
  1150. av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
  1151. s_index=0;
  1152. }else if(bits_left > 0 && s->error_recognition >= FF_ER_AGGRESSIVE){
  1153. av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
  1154. s_index=0;
  1155. }
  1156. memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid)*(576 - s_index));
  1157. skip_bits_long(&s->gb, bits_left);
  1158. i= get_bits_count(&s->gb);
  1159. switch_buffer(s, &i, &end_pos, &end_pos2);
  1160. return 0;
  1161. }
  1162. /* Reorder short blocks from bitstream order to interleaved order. It
  1163. would be faster to do it in parsing, but the code would be far more
  1164. complicated */
  1165. static void reorder_block(MPADecodeContext *s, GranuleDef *g)
  1166. {
  1167. int i, j, len;
  1168. INTFLOAT *ptr, *dst, *ptr1;
  1169. INTFLOAT tmp[576];
  1170. if (g->block_type != 2)
  1171. return;
  1172. if (g->switch_point) {
  1173. if (s->sample_rate_index != 8) {
  1174. ptr = g->sb_hybrid + 36;
  1175. } else {
  1176. ptr = g->sb_hybrid + 48;
  1177. }
  1178. } else {
  1179. ptr = g->sb_hybrid;
  1180. }
  1181. for(i=g->short_start;i<13;i++) {
  1182. len = band_size_short[s->sample_rate_index][i];
  1183. ptr1 = ptr;
  1184. dst = tmp;
  1185. for(j=len;j>0;j--) {
  1186. *dst++ = ptr[0*len];
  1187. *dst++ = ptr[1*len];
  1188. *dst++ = ptr[2*len];
  1189. ptr++;
  1190. }
  1191. ptr+=2*len;
  1192. memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
  1193. }
  1194. }
  1195. #define ISQRT2 FIXR(0.70710678118654752440)
  1196. static void compute_stereo(MPADecodeContext *s,
  1197. GranuleDef *g0, GranuleDef *g1)
  1198. {
  1199. int i, j, k, l;
  1200. int sf_max, sf, len, non_zero_found;
  1201. INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;
  1202. int non_zero_found_short[3];
  1203. /* intensity stereo */
  1204. if (s->mode_ext & MODE_EXT_I_STEREO) {
  1205. if (!s->lsf) {
  1206. is_tab = is_table;
  1207. sf_max = 7;
  1208. } else {
  1209. is_tab = is_table_lsf[g1->scalefac_compress & 1];
  1210. sf_max = 16;
  1211. }
  1212. tab0 = g0->sb_hybrid + 576;
  1213. tab1 = g1->sb_hybrid + 576;
  1214. non_zero_found_short[0] = 0;
  1215. non_zero_found_short[1] = 0;
  1216. non_zero_found_short[2] = 0;
  1217. k = (13 - g1->short_start) * 3 + g1->long_end - 3;
  1218. for(i = 12;i >= g1->short_start;i--) {
  1219. /* for last band, use previous scale factor */
  1220. if (i != 11)
  1221. k -= 3;
  1222. len = band_size_short[s->sample_rate_index][i];
  1223. for(l=2;l>=0;l--) {
  1224. tab0 -= len;
  1225. tab1 -= len;
  1226. if (!non_zero_found_short[l]) {
  1227. /* test if non zero band. if so, stop doing i-stereo */
  1228. for(j=0;j<len;j++) {
  1229. if (tab1[j] != 0) {
  1230. non_zero_found_short[l] = 1;
  1231. goto found1;
  1232. }
  1233. }
  1234. sf = g1->scale_factors[k + l];
  1235. if (sf >= sf_max)
  1236. goto found1;
  1237. v1 = is_tab[0][sf];
  1238. v2 = is_tab[1][sf];
  1239. for(j=0;j<len;j++) {
  1240. tmp0 = tab0[j];
  1241. tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
  1242. tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
  1243. }
  1244. } else {
  1245. found1:
  1246. if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1247. /* lower part of the spectrum : do ms stereo
  1248. if enabled */
  1249. for(j=0;j<len;j++) {
  1250. tmp0 = tab0[j];
  1251. tmp1 = tab1[j];
  1252. tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
  1253. tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
  1254. }
  1255. }
  1256. }
  1257. }
  1258. }
  1259. non_zero_found = non_zero_found_short[0] |
  1260. non_zero_found_short[1] |
  1261. non_zero_found_short[2];
  1262. for(i = g1->long_end - 1;i >= 0;i--) {
  1263. len = band_size_long[s->sample_rate_index][i];
  1264. tab0 -= len;
  1265. tab1 -= len;
  1266. /* test if non zero band. if so, stop doing i-stereo */
  1267. if (!non_zero_found) {
  1268. for(j=0;j<len;j++) {
  1269. if (tab1[j] != 0) {
  1270. non_zero_found = 1;
  1271. goto found2;
  1272. }
  1273. }
  1274. /* for last band, use previous scale factor */
  1275. k = (i == 21) ? 20 : i;
  1276. sf = g1->scale_factors[k];
  1277. if (sf >= sf_max)
  1278. goto found2;
  1279. v1 = is_tab[0][sf];
  1280. v2 = is_tab[1][sf];
  1281. for(j=0;j<len;j++) {
  1282. tmp0 = tab0[j];
  1283. tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
  1284. tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
  1285. }
  1286. } else {
  1287. found2:
  1288. if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1289. /* lower part of the spectrum : do ms stereo
  1290. if enabled */
  1291. for(j=0;j<len;j++) {
  1292. tmp0 = tab0[j];
  1293. tmp1 = tab1[j];
  1294. tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
  1295. tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
  1296. }
  1297. }
  1298. }
  1299. }
  1300. } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1301. /* ms stereo ONLY */
  1302. /* NOTE: the 1/sqrt(2) normalization factor is included in the
  1303. global gain */
  1304. tab0 = g0->sb_hybrid;
  1305. tab1 = g1->sb_hybrid;
  1306. for(i=0;i<576;i++) {
  1307. tmp0 = tab0[i];
  1308. tmp1 = tab1[i];
  1309. tab0[i] = tmp0 + tmp1;
  1310. tab1[i] = tmp0 - tmp1;
  1311. }
  1312. }
  1313. }
  1314. #if !CONFIG_FLOAT
  1315. static void compute_antialias_fixed(MPADecodeContext *s, GranuleDef *g)
  1316. {
  1317. int32_t *ptr, *csa;
  1318. int n, i;
  1319. /* we antialias only "long" bands */
  1320. if (g->block_type == 2) {
  1321. if (!g->switch_point)
  1322. return;
  1323. /* XXX: check this for 8000Hz case */
  1324. n = 1;
  1325. } else {
  1326. n = SBLIMIT - 1;
  1327. }
  1328. ptr = g->sb_hybrid + 18;
  1329. for(i = n;i > 0;i--) {
  1330. int tmp0, tmp1, tmp2;
  1331. csa = &csa_table[0][0];
  1332. #define INT_AA(j) \
  1333. tmp0 = ptr[-1-j];\
  1334. tmp1 = ptr[ j];\
  1335. tmp2= MULH(tmp0 + tmp1, csa[0+4*j]);\
  1336. ptr[-1-j] = 4*(tmp2 - MULH(tmp1, csa[2+4*j]));\
  1337. ptr[ j] = 4*(tmp2 + MULH(tmp0, csa[3+4*j]));
  1338. INT_AA(0)
  1339. INT_AA(1)
  1340. INT_AA(2)
  1341. INT_AA(3)
  1342. INT_AA(4)
  1343. INT_AA(5)
  1344. INT_AA(6)
  1345. INT_AA(7)
  1346. ptr += 18;
  1347. }
  1348. }
  1349. #endif
  1350. static void compute_imdct(MPADecodeContext *s,
  1351. GranuleDef *g,
  1352. INTFLOAT *sb_samples,
  1353. INTFLOAT *mdct_buf)
  1354. {
  1355. INTFLOAT *win, *win1, *out_ptr, *ptr, *buf, *ptr1;
  1356. INTFLOAT out2[12];
  1357. int i, j, mdct_long_end, sblimit;
  1358. /* find last non zero block */
  1359. ptr = g->sb_hybrid + 576;
  1360. ptr1 = g->sb_hybrid + 2 * 18;
  1361. while (ptr >= ptr1) {
  1362. int32_t *p;
  1363. ptr -= 6;
  1364. p= (int32_t*)ptr;
  1365. if(p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
  1366. break;
  1367. }
  1368. sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
  1369. if (g->block_type == 2) {
  1370. /* XXX: check for 8000 Hz */
  1371. if (g->switch_point)
  1372. mdct_long_end = 2;
  1373. else
  1374. mdct_long_end = 0;
  1375. } else {
  1376. mdct_long_end = sblimit;
  1377. }
  1378. buf = mdct_buf;
  1379. ptr = g->sb_hybrid;
  1380. for(j=0;j<mdct_long_end;j++) {
  1381. /* apply window & overlap with previous buffer */
  1382. out_ptr = sb_samples + j;
  1383. /* select window */
  1384. if (g->switch_point && j < 2)
  1385. win1 = mdct_win[0];
  1386. else
  1387. win1 = mdct_win[g->block_type];
  1388. /* select frequency inversion */
  1389. win = win1 + ((4 * 36) & -(j & 1));
  1390. imdct36(out_ptr, buf, ptr, win);
  1391. out_ptr += 18*SBLIMIT;
  1392. ptr += 18;
  1393. buf += 18;
  1394. }
  1395. for(j=mdct_long_end;j<sblimit;j++) {
  1396. /* select frequency inversion */
  1397. win = mdct_win[2] + ((4 * 36) & -(j & 1));
  1398. out_ptr = sb_samples + j;
  1399. for(i=0; i<6; i++){
  1400. *out_ptr = buf[i];
  1401. out_ptr += SBLIMIT;
  1402. }
  1403. imdct12(out2, ptr + 0);
  1404. for(i=0;i<6;i++) {
  1405. *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[i + 6*1];
  1406. buf[i + 6*2] = MULH3(out2[i + 6], win[i + 6], 1);
  1407. out_ptr += SBLIMIT;
  1408. }
  1409. imdct12(out2, ptr + 1);
  1410. for(i=0;i<6;i++) {
  1411. *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[i + 6*2];
  1412. buf[i + 6*0] = MULH3(out2[i + 6], win[i + 6], 1);
  1413. out_ptr += SBLIMIT;
  1414. }
  1415. imdct12(out2, ptr + 2);
  1416. for(i=0;i<6;i++) {
  1417. buf[i + 6*0] = MULH3(out2[i ], win[i ], 1) + buf[i + 6*0];
  1418. buf[i + 6*1] = MULH3(out2[i + 6], win[i + 6], 1);
  1419. buf[i + 6*2] = 0;
  1420. }
  1421. ptr += 18;
  1422. buf += 18;
  1423. }
  1424. /* zero bands */
  1425. for(j=sblimit;j<SBLIMIT;j++) {
  1426. /* overlap */
  1427. out_ptr = sb_samples + j;
  1428. for(i=0;i<18;i++) {
  1429. *out_ptr = buf[i];
  1430. buf[i] = 0;
  1431. out_ptr += SBLIMIT;
  1432. }
  1433. buf += 18;
  1434. }
  1435. }
  1436. /* main layer3 decoding function */
  1437. static int mp_decode_layer3(MPADecodeContext *s)
  1438. {
  1439. int nb_granules, main_data_begin, private_bits;
  1440. int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
  1441. GranuleDef *g;
  1442. int16_t exponents[576]; //FIXME try INTFLOAT
  1443. /* read side info */
  1444. if (s->lsf) {
  1445. main_data_begin = get_bits(&s->gb, 8);
  1446. private_bits = get_bits(&s->gb, s->nb_channels);
  1447. nb_granules = 1;
  1448. } else {
  1449. main_data_begin = get_bits(&s->gb, 9);
  1450. if (s->nb_channels == 2)
  1451. private_bits = get_bits(&s->gb, 3);
  1452. else
  1453. private_bits = get_bits(&s->gb, 5);
  1454. nb_granules = 2;
  1455. for(ch=0;ch<s->nb_channels;ch++) {
  1456. s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
  1457. s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
  1458. }
  1459. }
  1460. for(gr=0;gr<nb_granules;gr++) {
  1461. for(ch=0;ch<s->nb_channels;ch++) {
  1462. av_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
  1463. g = &s->granules[ch][gr];
  1464. g->part2_3_length = get_bits(&s->gb, 12);
  1465. g->big_values = get_bits(&s->gb, 9);
  1466. if(g->big_values > 288){
  1467. av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
  1468. return -1;
  1469. }
  1470. g->global_gain = get_bits(&s->gb, 8);
  1471. /* if MS stereo only is selected, we precompute the
  1472. 1/sqrt(2) renormalization factor */
  1473. if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
  1474. MODE_EXT_MS_STEREO)
  1475. g->global_gain -= 2;
  1476. if (s->lsf)
  1477. g->scalefac_compress = get_bits(&s->gb, 9);
  1478. else
  1479. g->scalefac_compress = get_bits(&s->gb, 4);
  1480. blocksplit_flag = get_bits1(&s->gb);
  1481. if (blocksplit_flag) {
  1482. g->block_type = get_bits(&s->gb, 2);
  1483. if (g->block_type == 0){
  1484. av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
  1485. return -1;
  1486. }
  1487. g->switch_point = get_bits1(&s->gb);
  1488. for(i=0;i<2;i++)
  1489. g->table_select[i] = get_bits(&s->gb, 5);
  1490. for(i=0;i<3;i++)
  1491. g->subblock_gain[i] = get_bits(&s->gb, 3);
  1492. ff_init_short_region(s, g);
  1493. } else {
  1494. int region_address1, region_address2;
  1495. g->block_type = 0;
  1496. g->switch_point = 0;
  1497. for(i=0;i<3;i++)
  1498. g->table_select[i] = get_bits(&s->gb, 5);
  1499. /* compute huffman coded region sizes */
  1500. region_address1 = get_bits(&s->gb, 4);
  1501. region_address2 = get_bits(&s->gb, 3);
  1502. av_dlog(s->avctx, "region1=%d region2=%d\n",
  1503. region_address1, region_address2);
  1504. ff_init_long_region(s, g, region_address1, region_address2);
  1505. }
  1506. ff_region_offset2size(g);
  1507. ff_compute_band_indexes(s, g);
  1508. g->preflag = 0;
  1509. if (!s->lsf)
  1510. g->preflag = get_bits1(&s->gb);
  1511. g->scalefac_scale = get_bits1(&s->gb);
  1512. g->count1table_select = get_bits1(&s->gb);
  1513. av_dlog(s->avctx, "block_type=%d switch_point=%d\n",
  1514. g->block_type, g->switch_point);
  1515. }
  1516. }
  1517. if (!s->adu_mode) {
  1518. const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
  1519. assert((get_bits_count(&s->gb) & 7) == 0);
  1520. /* now we get bits from the main_data_begin offset */
  1521. av_dlog(s->avctx, "seekback: %d\n", main_data_begin);
  1522. //av_log(NULL, AV_LOG_ERROR, "backstep:%d, lastbuf:%d\n", main_data_begin, s->last_buf_size);
  1523. memcpy(s->last_buf + s->last_buf_size, ptr, EXTRABYTES);
  1524. s->in_gb= s->gb;
  1525. init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);
  1526. skip_bits_long(&s->gb, 8*(s->last_buf_size - main_data_begin));
  1527. }
  1528. for(gr=0;gr<nb_granules;gr++) {
  1529. for(ch=0;ch<s->nb_channels;ch++) {
  1530. g = &s->granules[ch][gr];
  1531. if(get_bits_count(&s->gb)<0){
  1532. av_log(s->avctx, AV_LOG_DEBUG, "mdb:%d, lastbuf:%d skipping granule %d\n",
  1533. main_data_begin, s->last_buf_size, gr);
  1534. skip_bits_long(&s->gb, g->part2_3_length);
  1535. memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
  1536. if(get_bits_count(&s->gb) >= s->gb.size_in_bits && s->in_gb.buffer){
  1537. skip_bits_long(&s->in_gb, get_bits_count(&s->gb) - s->gb.size_in_bits);
  1538. s->gb= s->in_gb;
  1539. s->in_gb.buffer=NULL;
  1540. }
  1541. continue;
  1542. }
  1543. bits_pos = get_bits_count(&s->gb);
  1544. if (!s->lsf) {
  1545. uint8_t *sc;
  1546. int slen, slen1, slen2;
  1547. /* MPEG1 scale factors */
  1548. slen1 = slen_table[0][g->scalefac_compress];
  1549. slen2 = slen_table[1][g->scalefac_compress];
  1550. av_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
  1551. if (g->block_type == 2) {
  1552. n = g->switch_point ? 17 : 18;
  1553. j = 0;
  1554. if(slen1){
  1555. for(i=0;i<n;i++)
  1556. g->scale_factors[j++] = get_bits(&s->gb, slen1);
  1557. }else{
  1558. for(i=0;i<n;i++)
  1559. g->scale_factors[j++] = 0;
  1560. }
  1561. if(slen2){
  1562. for(i=0;i<18;i++)
  1563. g->scale_factors[j++] = get_bits(&s->gb, slen2);
  1564. for(i=0;i<3;i++)
  1565. g->scale_factors[j++] = 0;
  1566. }else{
  1567. for(i=0;i<21;i++)
  1568. g->scale_factors[j++] = 0;
  1569. }
  1570. } else {
  1571. sc = s->granules[ch][0].scale_factors;
  1572. j = 0;
  1573. for(k=0;k<4;k++) {
  1574. n = (k == 0 ? 6 : 5);
  1575. if ((g->scfsi & (0x8 >> k)) == 0) {
  1576. slen = (k < 2) ? slen1 : slen2;
  1577. if(slen){
  1578. for(i=0;i<n;i++)
  1579. g->scale_factors[j++] = get_bits(&s->gb, slen);
  1580. }else{
  1581. for(i=0;i<n;i++)
  1582. g->scale_factors[j++] = 0;
  1583. }
  1584. } else {
  1585. /* simply copy from last granule */
  1586. for(i=0;i<n;i++) {
  1587. g->scale_factors[j] = sc[j];
  1588. j++;
  1589. }
  1590. }
  1591. }
  1592. g->scale_factors[j++] = 0;
  1593. }
  1594. } else {
  1595. int tindex, tindex2, slen[4], sl, sf;
  1596. /* LSF scale factors */
  1597. if (g->block_type == 2) {
  1598. tindex = g->switch_point ? 2 : 1;
  1599. } else {
  1600. tindex = 0;
  1601. }
  1602. sf = g->scalefac_compress;
  1603. if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
  1604. /* intensity stereo case */
  1605. sf >>= 1;
  1606. if (sf < 180) {
  1607. lsf_sf_expand(slen, sf, 6, 6, 0);
  1608. tindex2 = 3;
  1609. } else if (sf < 244) {
  1610. lsf_sf_expand(slen, sf - 180, 4, 4, 0);
  1611. tindex2 = 4;
  1612. } else {
  1613. lsf_sf_expand(slen, sf - 244, 3, 0, 0);
  1614. tindex2 = 5;
  1615. }
  1616. } else {
  1617. /* normal case */
  1618. if (sf < 400) {
  1619. lsf_sf_expand(slen, sf, 5, 4, 4);
  1620. tindex2 = 0;
  1621. } else if (sf < 500) {
  1622. lsf_sf_expand(slen, sf - 400, 5, 4, 0);
  1623. tindex2 = 1;
  1624. } else {
  1625. lsf_sf_expand(slen, sf - 500, 3, 0, 0);
  1626. tindex2 = 2;
  1627. g->preflag = 1;
  1628. }
  1629. }
  1630. j = 0;
  1631. for(k=0;k<4;k++) {
  1632. n = lsf_nsf_table[tindex2][tindex][k];
  1633. sl = slen[k];
  1634. if(sl){
  1635. for(i=0;i<n;i++)
  1636. g->scale_factors[j++] = get_bits(&s->gb, sl);
  1637. }else{
  1638. for(i=0;i<n;i++)
  1639. g->scale_factors[j++] = 0;
  1640. }
  1641. }
  1642. /* XXX: should compute exact size */
  1643. for(;j<40;j++)
  1644. g->scale_factors[j] = 0;
  1645. }
  1646. exponents_from_scale_factors(s, g, exponents);
  1647. /* read Huffman coded residue */
  1648. huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
  1649. } /* ch */
  1650. if (s->nb_channels == 2)
  1651. compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
  1652. for(ch=0;ch<s->nb_channels;ch++) {
  1653. g = &s->granules[ch][gr];
  1654. reorder_block(s, g);
  1655. RENAME(compute_antialias)(s, g);
  1656. compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
  1657. }
  1658. } /* gr */
  1659. if(get_bits_count(&s->gb)<0)
  1660. skip_bits_long(&s->gb, -get_bits_count(&s->gb));
  1661. return nb_granules * 18;
  1662. }
  1663. static int mp_decode_frame(MPADecodeContext *s,
  1664. OUT_INT *samples, const uint8_t *buf, int buf_size)
  1665. {
  1666. int i, nb_frames, ch;
  1667. OUT_INT *samples_ptr;
  1668. init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE)*8);
  1669. /* skip error protection field */
  1670. if (s->error_protection)
  1671. skip_bits(&s->gb, 16);
  1672. switch(s->layer) {
  1673. case 1:
  1674. s->avctx->frame_size = 384;
  1675. nb_frames = mp_decode_layer1(s);
  1676. break;
  1677. case 2:
  1678. s->avctx->frame_size = 1152;
  1679. nb_frames = mp_decode_layer2(s);
  1680. break;
  1681. case 3:
  1682. s->avctx->frame_size = s->lsf ? 576 : 1152;
  1683. default:
  1684. nb_frames = mp_decode_layer3(s);
  1685. s->last_buf_size=0;
  1686. if(s->in_gb.buffer){
  1687. align_get_bits(&s->gb);
  1688. i= get_bits_left(&s->gb)>>3;
  1689. if(i >= 0 && i <= BACKSTEP_SIZE){
  1690. memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);
  1691. s->last_buf_size=i;
  1692. }else
  1693. av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
  1694. s->gb= s->in_gb;
  1695. s->in_gb.buffer= NULL;
  1696. }
  1697. align_get_bits(&s->gb);
  1698. assert((get_bits_count(&s->gb) & 7) == 0);
  1699. i= get_bits_left(&s->gb)>>3;
  1700. if(i<0 || i > BACKSTEP_SIZE || nb_frames<0){
  1701. if(i<0)
  1702. av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
  1703. i= FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
  1704. }
  1705. assert(i <= buf_size - HEADER_SIZE && i>= 0);
  1706. memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
  1707. s->last_buf_size += i;
  1708. break;
  1709. }
  1710. /* apply the synthesis filter */
  1711. for(ch=0;ch<s->nb_channels;ch++) {
  1712. samples_ptr = samples + ch;
  1713. for(i=0;i<nb_frames;i++) {
  1714. RENAME(ff_mpa_synth_filter)(
  1715. #if CONFIG_FLOAT
  1716. s,
  1717. #endif
  1718. s->synth_buf[ch], &(s->synth_buf_offset[ch]),
  1719. RENAME(ff_mpa_synth_window), &s->dither_state,
  1720. samples_ptr, s->nb_channels,
  1721. s->sb_samples[ch][i]);
  1722. samples_ptr += 32 * s->nb_channels;
  1723. }
  1724. }
  1725. return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
  1726. }
  1727. static int decode_frame(AVCodecContext * avctx,
  1728. void *data, int *data_size,
  1729. AVPacket *avpkt)
  1730. {
  1731. const uint8_t *buf = avpkt->data;
  1732. int buf_size = avpkt->size;
  1733. MPADecodeContext *s = avctx->priv_data;
  1734. uint32_t header;
  1735. int out_size;
  1736. OUT_INT *out_samples = data;
  1737. if(buf_size < HEADER_SIZE)
  1738. return -1;
  1739. header = AV_RB32(buf);
  1740. if(ff_mpa_check_header(header) < 0){
  1741. av_log(avctx, AV_LOG_ERROR, "Header missing\n");
  1742. return -1;
  1743. }
  1744. if (ff_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
  1745. /* free format: prepare to compute frame size */
  1746. s->frame_size = -1;
  1747. return -1;
  1748. }
  1749. /* update codec info */
  1750. avctx->channels = s->nb_channels;
  1751. avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
  1752. if (!avctx->bit_rate)
  1753. avctx->bit_rate = s->bit_rate;
  1754. avctx->sub_id = s->layer;
  1755. if(*data_size < 1152*avctx->channels*sizeof(OUT_INT))
  1756. return -1;
  1757. *data_size = 0;
  1758. if(s->frame_size<=0 || s->frame_size > buf_size){
  1759. av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
  1760. return -1;
  1761. }else if(s->frame_size < buf_size){
  1762. av_log(avctx, AV_LOG_ERROR, "incorrect frame size\n");
  1763. buf_size= s->frame_size;
  1764. }
  1765. out_size = mp_decode_frame(s, out_samples, buf, buf_size);
  1766. if(out_size>=0){
  1767. *data_size = out_size;
  1768. avctx->sample_rate = s->sample_rate;
  1769. //FIXME maybe move the other codec info stuff from above here too
  1770. }else
  1771. av_log(avctx, AV_LOG_DEBUG, "Error while decoding MPEG audio frame.\n"); //FIXME return -1 / but also return the number of bytes consumed
  1772. s->frame_size = 0;
  1773. return buf_size;
  1774. }
  1775. static void flush(AVCodecContext *avctx){
  1776. MPADecodeContext *s = avctx->priv_data;
  1777. memset(s->synth_buf, 0, sizeof(s->synth_buf));
  1778. s->last_buf_size= 0;
  1779. }
  1780. #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
  1781. static int decode_frame_adu(AVCodecContext * avctx,
  1782. void *data, int *data_size,
  1783. AVPacket *avpkt)
  1784. {
  1785. const uint8_t *buf = avpkt->data;
  1786. int buf_size = avpkt->size;
  1787. MPADecodeContext *s = avctx->priv_data;
  1788. uint32_t header;
  1789. int len, out_size;
  1790. OUT_INT *out_samples = data;
  1791. len = buf_size;
  1792. // Discard too short frames
  1793. if (buf_size < HEADER_SIZE) {
  1794. *data_size = 0;
  1795. return buf_size;
  1796. }
  1797. if (len > MPA_MAX_CODED_FRAME_SIZE)
  1798. len = MPA_MAX_CODED_FRAME_SIZE;
  1799. // Get header and restore sync word
  1800. header = AV_RB32(buf) | 0xffe00000;
  1801. if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame
  1802. *data_size = 0;
  1803. return buf_size;
  1804. }
  1805. ff_mpegaudio_decode_header((MPADecodeHeader *)s, header);
  1806. /* update codec info */
  1807. avctx->sample_rate = s->sample_rate;
  1808. avctx->channels = s->nb_channels;
  1809. if (!avctx->bit_rate)
  1810. avctx->bit_rate = s->bit_rate;
  1811. avctx->sub_id = s->layer;
  1812. s->frame_size = len;
  1813. if (avctx->parse_only) {
  1814. out_size = buf_size;
  1815. } else {
  1816. out_size = mp_decode_frame(s, out_samples, buf, buf_size);
  1817. }
  1818. *data_size = out_size;
  1819. return buf_size;
  1820. }
  1821. #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
  1822. #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
  1823. /**
  1824. * Context for MP3On4 decoder
  1825. */
  1826. typedef struct MP3On4DecodeContext {
  1827. int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
  1828. int syncword; ///< syncword patch
  1829. const uint8_t *coff; ///< channels offsets in output buffer
  1830. MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
  1831. } MP3On4DecodeContext;
  1832. #include "mpeg4audio.h"
  1833. /* Next 3 arrays are indexed by channel config number (passed via codecdata) */
  1834. static const uint8_t mp3Frames[8] = {0,1,1,2,3,3,4,5}; /* number of mp3 decoder instances */
  1835. /* offsets into output buffer, assume output order is FL FR BL BR C LFE */
  1836. static const uint8_t chan_offset[8][5] = {
  1837. {0},
  1838. {0}, // C
  1839. {0}, // FLR
  1840. {2,0}, // C FLR
  1841. {2,0,3}, // C FLR BS
  1842. {4,0,2}, // C FLR BLRS
  1843. {4,0,2,5}, // C FLR BLRS LFE
  1844. {4,0,2,6,5}, // C FLR BLRS BLR LFE
  1845. };
  1846. static int decode_init_mp3on4(AVCodecContext * avctx)
  1847. {
  1848. MP3On4DecodeContext *s = avctx->priv_data;
  1849. MPEG4AudioConfig cfg;
  1850. int i;
  1851. if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) {
  1852. av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
  1853. return -1;
  1854. }
  1855. ff_mpeg4audio_get_config(&cfg, avctx->extradata, avctx->extradata_size);
  1856. if (!cfg.chan_config || cfg.chan_config > 7) {
  1857. av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
  1858. return -1;
  1859. }
  1860. s->frames = mp3Frames[cfg.chan_config];
  1861. s->coff = chan_offset[cfg.chan_config];
  1862. avctx->channels = ff_mpeg4audio_channels[cfg.chan_config];
  1863. if (cfg.sample_rate < 16000)
  1864. s->syncword = 0xffe00000;
  1865. else
  1866. s->syncword = 0xfff00000;
  1867. /* Init the first mp3 decoder in standard way, so that all tables get builded
  1868. * We replace avctx->priv_data with the context of the first decoder so that
  1869. * decode_init() does not have to be changed.
  1870. * Other decoders will be initialized here copying data from the first context
  1871. */
  1872. // Allocate zeroed memory for the first decoder context
  1873. s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
  1874. // Put decoder context in place to make init_decode() happy
  1875. avctx->priv_data = s->mp3decctx[0];
  1876. decode_init(avctx);
  1877. // Restore mp3on4 context pointer
  1878. avctx->priv_data = s;
  1879. s->mp3decctx[0]->adu_mode = 1; // Set adu mode
  1880. /* Create a separate codec/context for each frame (first is already ok).
  1881. * Each frame is 1 or 2 channels - up to 5 frames allowed
  1882. */
  1883. for (i = 1; i < s->frames; i++) {
  1884. s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
  1885. s->mp3decctx[i]->adu_mode = 1;
  1886. s->mp3decctx[i]->avctx = avctx;
  1887. }
  1888. return 0;
  1889. }
  1890. static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
  1891. {
  1892. MP3On4DecodeContext *s = avctx->priv_data;
  1893. int i;
  1894. for (i = 0; i < s->frames; i++)
  1895. av_free(s->mp3decctx[i]);
  1896. return 0;
  1897. }
  1898. static int decode_frame_mp3on4(AVCodecContext * avctx,
  1899. void *data, int *data_size,
  1900. AVPacket *avpkt)
  1901. {
  1902. const uint8_t *buf = avpkt->data;
  1903. int buf_size = avpkt->size;
  1904. MP3On4DecodeContext *s = avctx->priv_data;
  1905. MPADecodeContext *m;
  1906. int fsize, len = buf_size, out_size = 0;
  1907. uint32_t header;
  1908. OUT_INT *out_samples = data;
  1909. OUT_INT decoded_buf[MPA_FRAME_SIZE * MPA_MAX_CHANNELS];
  1910. OUT_INT *outptr, *bp;
  1911. int fr, j, n;
  1912. if(*data_size < MPA_FRAME_SIZE * MPA_MAX_CHANNELS * s->frames * sizeof(OUT_INT))
  1913. return -1;
  1914. *data_size = 0;
  1915. // Discard too short frames
  1916. if (buf_size < HEADER_SIZE)
  1917. return -1;
  1918. // If only one decoder interleave is not needed
  1919. outptr = s->frames == 1 ? out_samples : decoded_buf;
  1920. avctx->bit_rate = 0;
  1921. for (fr = 0; fr < s->frames; fr++) {
  1922. fsize = AV_RB16(buf) >> 4;
  1923. fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
  1924. m = s->mp3decctx[fr];
  1925. assert (m != NULL);
  1926. header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
  1927. if (ff_mpa_check_header(header) < 0) // Bad header, discard block
  1928. break;
  1929. ff_mpegaudio_decode_header((MPADecodeHeader *)m, header);
  1930. out_size += mp_decode_frame(m, outptr, buf, fsize);
  1931. buf += fsize;
  1932. len -= fsize;
  1933. if(s->frames > 1) {
  1934. n = m->avctx->frame_size*m->nb_channels;
  1935. /* interleave output data */
  1936. bp = out_samples + s->coff[fr];
  1937. if(m->nb_channels == 1) {
  1938. for(j = 0; j < n; j++) {
  1939. *bp = decoded_buf[j];
  1940. bp += avctx->channels;
  1941. }
  1942. } else {
  1943. for(j = 0; j < n; j++) {
  1944. bp[0] = decoded_buf[j++];
  1945. bp[1] = decoded_buf[j];
  1946. bp += avctx->channels;
  1947. }
  1948. }
  1949. }
  1950. avctx->bit_rate += m->bit_rate;
  1951. }
  1952. /* update codec info */
  1953. avctx->sample_rate = s->mp3decctx[0]->sample_rate;
  1954. *data_size = out_size;
  1955. return buf_size;
  1956. }
  1957. #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
  1958. #if !CONFIG_FLOAT
  1959. #if CONFIG_MP1_DECODER
  1960. AVCodec ff_mp1_decoder =
  1961. {
  1962. "mp1",
  1963. AVMEDIA_TYPE_AUDIO,
  1964. CODEC_ID_MP1,
  1965. sizeof(MPADecodeContext),
  1966. decode_init,
  1967. NULL,
  1968. NULL,
  1969. decode_frame,
  1970. CODEC_CAP_PARSE_ONLY,
  1971. .flush= flush,
  1972. .long_name= NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
  1973. };
  1974. #endif
  1975. #if CONFIG_MP2_DECODER
  1976. AVCodec ff_mp2_decoder =
  1977. {
  1978. "mp2",
  1979. AVMEDIA_TYPE_AUDIO,
  1980. CODEC_ID_MP2,
  1981. sizeof(MPADecodeContext),
  1982. decode_init,
  1983. NULL,
  1984. NULL,
  1985. decode_frame,
  1986. CODEC_CAP_PARSE_ONLY,
  1987. .flush= flush,
  1988. .long_name= NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
  1989. };
  1990. #endif
  1991. #if CONFIG_MP3_DECODER
  1992. AVCodec ff_mp3_decoder =
  1993. {
  1994. "mp3",
  1995. AVMEDIA_TYPE_AUDIO,
  1996. CODEC_ID_MP3,
  1997. sizeof(MPADecodeContext),
  1998. decode_init,
  1999. NULL,
  2000. NULL,
  2001. decode_frame,
  2002. CODEC_CAP_PARSE_ONLY,
  2003. .flush= flush,
  2004. .long_name= NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
  2005. };
  2006. #endif
  2007. #if CONFIG_MP3ADU_DECODER
  2008. AVCodec ff_mp3adu_decoder =
  2009. {
  2010. "mp3adu",
  2011. AVMEDIA_TYPE_AUDIO,
  2012. CODEC_ID_MP3ADU,
  2013. sizeof(MPADecodeContext),
  2014. decode_init,
  2015. NULL,
  2016. NULL,
  2017. decode_frame_adu,
  2018. CODEC_CAP_PARSE_ONLY,
  2019. .flush= flush,
  2020. .long_name= NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
  2021. };
  2022. #endif
  2023. #if CONFIG_MP3ON4_DECODER
  2024. AVCodec ff_mp3on4_decoder =
  2025. {
  2026. "mp3on4",
  2027. AVMEDIA_TYPE_AUDIO,
  2028. CODEC_ID_MP3ON4,
  2029. sizeof(MP3On4DecodeContext),
  2030. decode_init_mp3on4,
  2031. NULL,
  2032. decode_close_mp3on4,
  2033. decode_frame_mp3on4,
  2034. .flush= flush,
  2035. .long_name= NULL_IF_CONFIG_SMALL("MP3onMP4"),
  2036. };
  2037. #endif
  2038. #endif