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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. #include "libavformat/id3v1.h"
  29. /*
  30. * TODO:
  31. * - in low precision mode, use more 16 bit multiplies in synth filter
  32. * - test lsf / mpeg25 extensively.
  33. */
  34. #include "mpegaudio.h"
  35. #include "mpegaudiodecheader.h"
  36. #include "mathops.h"
  37. #if CONFIG_FLOAT
  38. # define SHR(a,b) ((a)*(1.0f/(1<<(b))))
  39. # define compute_antialias compute_antialias_float
  40. # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
  41. # define FIXR(x) ((float)(x))
  42. # define FIXHR(x) ((float)(x))
  43. # define MULH3(x, y, s) ((s)*(y)*(x))
  44. # define MULLx(x, y, s) ((y)*(x))
  45. # define RENAME(a) a ## _float
  46. #else
  47. # define SHR(a,b) ((a)>>(b))
  48. # define compute_antialias compute_antialias_integer
  49. /* WARNING: only correct for posititive numbers */
  50. # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
  51. # define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
  52. # define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
  53. # define MULH3(x, y, s) MULH((s)*(x), y)
  54. # define MULLx(x, y, s) MULL(x,y,s)
  55. # define RENAME(a) a
  56. #endif
  57. /****************/
  58. #define HEADER_SIZE 4
  59. #include "mpegaudiodata.h"
  60. #include "mpegaudiodectab.h"
  61. #if CONFIG_FLOAT
  62. # include "fft.h"
  63. #else
  64. # include "dct32.c"
  65. #endif
  66. static void compute_antialias(MPADecodeContext *s, GranuleDef *g);
  67. static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
  68. int *dither_state, OUT_INT *samples, int incr);
  69. /* vlc structure for decoding layer 3 huffman tables */
  70. static VLC huff_vlc[16];
  71. static VLC_TYPE huff_vlc_tables[
  72. 0+128+128+128+130+128+154+166+
  73. 142+204+190+170+542+460+662+414
  74. ][2];
  75. static const int huff_vlc_tables_sizes[16] = {
  76. 0, 128, 128, 128, 130, 128, 154, 166,
  77. 142, 204, 190, 170, 542, 460, 662, 414
  78. };
  79. static VLC huff_quad_vlc[2];
  80. static VLC_TYPE huff_quad_vlc_tables[128+16][2];
  81. static const int huff_quad_vlc_tables_sizes[2] = {
  82. 128, 16
  83. };
  84. /* computed from band_size_long */
  85. static uint16_t band_index_long[9][23];
  86. #include "mpegaudio_tablegen.h"
  87. /* intensity stereo coef table */
  88. static INTFLOAT is_table[2][16];
  89. static INTFLOAT is_table_lsf[2][2][16];
  90. static int32_t csa_table[8][4];
  91. static float csa_table_float[8][4];
  92. static INTFLOAT mdct_win[8][36];
  93. static int16_t division_tab3[1<<6 ];
  94. static int16_t division_tab5[1<<8 ];
  95. static int16_t division_tab9[1<<11];
  96. static int16_t * const division_tabs[4] = {
  97. division_tab3, division_tab5, NULL, division_tab9
  98. };
  99. /* lower 2 bits: modulo 3, higher bits: shift */
  100. static uint16_t scale_factor_modshift[64];
  101. /* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
  102. static int32_t scale_factor_mult[15][3];
  103. /* mult table for layer 2 group quantization */
  104. #define SCALE_GEN(v) \
  105. { FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
  106. static const int32_t scale_factor_mult2[3][3] = {
  107. SCALE_GEN(4.0 / 3.0), /* 3 steps */
  108. SCALE_GEN(4.0 / 5.0), /* 5 steps */
  109. SCALE_GEN(4.0 / 9.0), /* 9 steps */
  110. };
  111. DECLARE_ALIGNED(16, MPA_INT, RENAME(ff_mpa_synth_window))[512+256];
  112. /**
  113. * Convert region offsets to region sizes and truncate
  114. * size to big_values.
  115. */
  116. static void ff_region_offset2size(GranuleDef *g){
  117. int i, k, j=0;
  118. g->region_size[2] = (576 / 2);
  119. for(i=0;i<3;i++) {
  120. k = FFMIN(g->region_size[i], g->big_values);
  121. g->region_size[i] = k - j;
  122. j = k;
  123. }
  124. }
  125. static void ff_init_short_region(MPADecodeContext *s, GranuleDef *g){
  126. if (g->block_type == 2)
  127. g->region_size[0] = (36 / 2);
  128. else {
  129. if (s->sample_rate_index <= 2)
  130. g->region_size[0] = (36 / 2);
  131. else if (s->sample_rate_index != 8)
  132. g->region_size[0] = (54 / 2);
  133. else
  134. g->region_size[0] = (108 / 2);
  135. }
  136. g->region_size[1] = (576 / 2);
  137. }
  138. static void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2){
  139. int l;
  140. g->region_size[0] =
  141. band_index_long[s->sample_rate_index][ra1 + 1] >> 1;
  142. /* should not overflow */
  143. l = FFMIN(ra1 + ra2 + 2, 22);
  144. g->region_size[1] =
  145. band_index_long[s->sample_rate_index][l] >> 1;
  146. }
  147. static void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g){
  148. if (g->block_type == 2) {
  149. if (g->switch_point) {
  150. /* if switched mode, we handle the 36 first samples as
  151. long blocks. For 8000Hz, we handle the 48 first
  152. exponents as long blocks (XXX: check this!) */
  153. if (s->sample_rate_index <= 2)
  154. g->long_end = 8;
  155. else if (s->sample_rate_index != 8)
  156. g->long_end = 6;
  157. else
  158. g->long_end = 4; /* 8000 Hz */
  159. g->short_start = 2 + (s->sample_rate_index != 8);
  160. } else {
  161. g->long_end = 0;
  162. g->short_start = 0;
  163. }
  164. } else {
  165. g->short_start = 13;
  166. g->long_end = 22;
  167. }
  168. }
  169. /* layer 1 unscaling */
  170. /* n = number of bits of the mantissa minus 1 */
  171. static inline int l1_unscale(int n, int mant, int scale_factor)
  172. {
  173. int shift, mod;
  174. int64_t val;
  175. shift = scale_factor_modshift[scale_factor];
  176. mod = shift & 3;
  177. shift >>= 2;
  178. val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]);
  179. shift += n;
  180. /* NOTE: at this point, 1 <= shift >= 21 + 15 */
  181. return (int)((val + (1LL << (shift - 1))) >> shift);
  182. }
  183. static inline int l2_unscale_group(int steps, int mant, int scale_factor)
  184. {
  185. int shift, mod, val;
  186. shift = scale_factor_modshift[scale_factor];
  187. mod = shift & 3;
  188. shift >>= 2;
  189. val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
  190. /* NOTE: at this point, 0 <= shift <= 21 */
  191. if (shift > 0)
  192. val = (val + (1 << (shift - 1))) >> shift;
  193. return val;
  194. }
  195. /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
  196. static inline int l3_unscale(int value, int exponent)
  197. {
  198. unsigned int m;
  199. int e;
  200. e = table_4_3_exp [4*value + (exponent&3)];
  201. m = table_4_3_value[4*value + (exponent&3)];
  202. e -= (exponent >> 2);
  203. assert(e>=1);
  204. if (e > 31)
  205. return 0;
  206. m = (m + (1 << (e-1))) >> e;
  207. return m;
  208. }
  209. /* all integer n^(4/3) computation code */
  210. #define DEV_ORDER 13
  211. #define POW_FRAC_BITS 24
  212. #define POW_FRAC_ONE (1 << POW_FRAC_BITS)
  213. #define POW_FIX(a) ((int)((a) * POW_FRAC_ONE))
  214. #define POW_MULL(a,b) (((int64_t)(a) * (int64_t)(b)) >> POW_FRAC_BITS)
  215. static int dev_4_3_coefs[DEV_ORDER];
  216. #if 0 /* unused */
  217. static int pow_mult3[3] = {
  218. POW_FIX(1.0),
  219. POW_FIX(1.25992104989487316476),
  220. POW_FIX(1.58740105196819947474),
  221. };
  222. #endif
  223. static av_cold void int_pow_init(void)
  224. {
  225. int i, a;
  226. a = POW_FIX(1.0);
  227. for(i=0;i<DEV_ORDER;i++) {
  228. a = POW_MULL(a, POW_FIX(4.0 / 3.0) - i * POW_FIX(1.0)) / (i + 1);
  229. dev_4_3_coefs[i] = a;
  230. }
  231. }
  232. #if 0 /* unused, remove? */
  233. /* return the mantissa and the binary exponent */
  234. static int int_pow(int i, int *exp_ptr)
  235. {
  236. int e, er, eq, j;
  237. int a, a1;
  238. /* renormalize */
  239. a = i;
  240. e = POW_FRAC_BITS;
  241. while (a < (1 << (POW_FRAC_BITS - 1))) {
  242. a = a << 1;
  243. e--;
  244. }
  245. a -= (1 << POW_FRAC_BITS);
  246. a1 = 0;
  247. for(j = DEV_ORDER - 1; j >= 0; j--)
  248. a1 = POW_MULL(a, dev_4_3_coefs[j] + a1);
  249. a = (1 << POW_FRAC_BITS) + a1;
  250. /* exponent compute (exact) */
  251. e = e * 4;
  252. er = e % 3;
  253. eq = e / 3;
  254. a = POW_MULL(a, pow_mult3[er]);
  255. while (a >= 2 * POW_FRAC_ONE) {
  256. a = a >> 1;
  257. eq++;
  258. }
  259. /* convert to float */
  260. while (a < POW_FRAC_ONE) {
  261. a = a << 1;
  262. eq--;
  263. }
  264. /* now POW_FRAC_ONE <= a < 2 * POW_FRAC_ONE */
  265. #if POW_FRAC_BITS > FRAC_BITS
  266. a = (a + (1 << (POW_FRAC_BITS - FRAC_BITS - 1))) >> (POW_FRAC_BITS - FRAC_BITS);
  267. /* correct overflow */
  268. if (a >= 2 * (1 << FRAC_BITS)) {
  269. a = a >> 1;
  270. eq++;
  271. }
  272. #endif
  273. *exp_ptr = eq;
  274. return a;
  275. }
  276. #endif
  277. static av_cold int decode_init(AVCodecContext * avctx)
  278. {
  279. MPADecodeContext *s = avctx->priv_data;
  280. static int init=0;
  281. int i, j, k;
  282. s->avctx = avctx;
  283. s->apply_window_mp3 = apply_window_mp3_c;
  284. #if HAVE_MMX && CONFIG_FLOAT
  285. ff_mpegaudiodec_init_mmx(s);
  286. #endif
  287. #if CONFIG_FLOAT
  288. ff_dct_init(&s->dct, 5, DCT_II);
  289. #endif
  290. if (HAVE_ALTIVEC && CONFIG_FLOAT) ff_mpegaudiodec_init_altivec(s);
  291. avctx->sample_fmt= OUT_FMT;
  292. s->error_recognition= avctx->error_recognition;
  293. if (!init && !avctx->parse_only) {
  294. int offset;
  295. /* scale factors table for layer 1/2 */
  296. for(i=0;i<64;i++) {
  297. int shift, mod;
  298. /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
  299. shift = (i / 3);
  300. mod = i % 3;
  301. scale_factor_modshift[i] = mod | (shift << 2);
  302. }
  303. /* scale factor multiply for layer 1 */
  304. for(i=0;i<15;i++) {
  305. int n, norm;
  306. n = i + 2;
  307. norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
  308. scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
  309. scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
  310. scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
  311. dprintf(avctx, "%d: norm=%x s=%x %x %x\n",
  312. i, norm,
  313. scale_factor_mult[i][0],
  314. scale_factor_mult[i][1],
  315. scale_factor_mult[i][2]);
  316. }
  317. RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));
  318. /* huffman decode tables */
  319. offset = 0;
  320. for(i=1;i<16;i++) {
  321. const HuffTable *h = &mpa_huff_tables[i];
  322. int xsize, x, y;
  323. uint8_t tmp_bits [512];
  324. uint16_t tmp_codes[512];
  325. memset(tmp_bits , 0, sizeof(tmp_bits ));
  326. memset(tmp_codes, 0, sizeof(tmp_codes));
  327. xsize = h->xsize;
  328. j = 0;
  329. for(x=0;x<xsize;x++) {
  330. for(y=0;y<xsize;y++){
  331. tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ];
  332. tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];
  333. }
  334. }
  335. /* XXX: fail test */
  336. huff_vlc[i].table = huff_vlc_tables+offset;
  337. huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
  338. init_vlc(&huff_vlc[i], 7, 512,
  339. tmp_bits, 1, 1, tmp_codes, 2, 2,
  340. INIT_VLC_USE_NEW_STATIC);
  341. offset += huff_vlc_tables_sizes[i];
  342. }
  343. assert(offset == FF_ARRAY_ELEMS(huff_vlc_tables));
  344. offset = 0;
  345. for(i=0;i<2;i++) {
  346. huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
  347. huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
  348. init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
  349. mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
  350. INIT_VLC_USE_NEW_STATIC);
  351. offset += huff_quad_vlc_tables_sizes[i];
  352. }
  353. assert(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));
  354. for(i=0;i<9;i++) {
  355. k = 0;
  356. for(j=0;j<22;j++) {
  357. band_index_long[i][j] = k;
  358. k += band_size_long[i][j];
  359. }
  360. band_index_long[i][22] = k;
  361. }
  362. /* compute n ^ (4/3) and store it in mantissa/exp format */
  363. int_pow_init();
  364. mpegaudio_tableinit();
  365. for (i = 0; i < 4; i++)
  366. if (ff_mpa_quant_bits[i] < 0)
  367. for (j = 0; j < (1<<(-ff_mpa_quant_bits[i]+1)); j++) {
  368. int val1, val2, val3, steps;
  369. int val = j;
  370. steps = ff_mpa_quant_steps[i];
  371. val1 = val % steps;
  372. val /= steps;
  373. val2 = val % steps;
  374. val3 = val / steps;
  375. division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);
  376. }
  377. for(i=0;i<7;i++) {
  378. float f;
  379. INTFLOAT v;
  380. if (i != 6) {
  381. f = tan((double)i * M_PI / 12.0);
  382. v = FIXR(f / (1.0 + f));
  383. } else {
  384. v = FIXR(1.0);
  385. }
  386. is_table[0][i] = v;
  387. is_table[1][6 - i] = v;
  388. }
  389. /* invalid values */
  390. for(i=7;i<16;i++)
  391. is_table[0][i] = is_table[1][i] = 0.0;
  392. for(i=0;i<16;i++) {
  393. double f;
  394. int e, k;
  395. for(j=0;j<2;j++) {
  396. e = -(j + 1) * ((i + 1) >> 1);
  397. f = pow(2.0, e / 4.0);
  398. k = i & 1;
  399. is_table_lsf[j][k ^ 1][i] = FIXR(f);
  400. is_table_lsf[j][k][i] = FIXR(1.0);
  401. dprintf(avctx, "is_table_lsf %d %d: %x %x\n",
  402. i, j, is_table_lsf[j][0][i], is_table_lsf[j][1][i]);
  403. }
  404. }
  405. for(i=0;i<8;i++) {
  406. float ci, cs, ca;
  407. ci = ci_table[i];
  408. cs = 1.0 / sqrt(1.0 + ci * ci);
  409. ca = cs * ci;
  410. csa_table[i][0] = FIXHR(cs/4);
  411. csa_table[i][1] = FIXHR(ca/4);
  412. csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);
  413. csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);
  414. csa_table_float[i][0] = cs;
  415. csa_table_float[i][1] = ca;
  416. csa_table_float[i][2] = ca + cs;
  417. csa_table_float[i][3] = ca - cs;
  418. }
  419. /* compute mdct windows */
  420. for(i=0;i<36;i++) {
  421. for(j=0; j<4; j++){
  422. double d;
  423. if(j==2 && i%3 != 1)
  424. continue;
  425. d= sin(M_PI * (i + 0.5) / 36.0);
  426. if(j==1){
  427. if (i>=30) d= 0;
  428. else if(i>=24) d= sin(M_PI * (i - 18 + 0.5) / 12.0);
  429. else if(i>=18) d= 1;
  430. }else if(j==3){
  431. if (i< 6) d= 0;
  432. else if(i< 12) d= sin(M_PI * (i - 6 + 0.5) / 12.0);
  433. else if(i< 18) d= 1;
  434. }
  435. //merge last stage of imdct into the window coefficients
  436. d*= 0.5 / cos(M_PI*(2*i + 19)/72);
  437. if(j==2)
  438. mdct_win[j][i/3] = FIXHR((d / (1<<5)));
  439. else
  440. mdct_win[j][i ] = FIXHR((d / (1<<5)));
  441. }
  442. }
  443. /* NOTE: we do frequency inversion adter the MDCT by changing
  444. the sign of the right window coefs */
  445. for(j=0;j<4;j++) {
  446. for(i=0;i<36;i+=2) {
  447. mdct_win[j + 4][i] = mdct_win[j][i];
  448. mdct_win[j + 4][i + 1] = -mdct_win[j][i + 1];
  449. }
  450. }
  451. init = 1;
  452. }
  453. if (avctx->codec_id == CODEC_ID_MP3ADU)
  454. s->adu_mode = 1;
  455. return 0;
  456. }
  457. #if CONFIG_FLOAT
  458. static inline float round_sample(float *sum)
  459. {
  460. float sum1=*sum;
  461. *sum = 0;
  462. return sum1;
  463. }
  464. /* signed 16x16 -> 32 multiply add accumulate */
  465. #define MACS(rt, ra, rb) rt+=(ra)*(rb)
  466. /* signed 16x16 -> 32 multiply */
  467. #define MULS(ra, rb) ((ra)*(rb))
  468. #define MLSS(rt, ra, rb) rt-=(ra)*(rb)
  469. #elif FRAC_BITS <= 15
  470. static inline int round_sample(int *sum)
  471. {
  472. int sum1;
  473. sum1 = (*sum) >> OUT_SHIFT;
  474. *sum &= (1<<OUT_SHIFT)-1;
  475. return av_clip(sum1, OUT_MIN, OUT_MAX);
  476. }
  477. /* signed 16x16 -> 32 multiply add accumulate */
  478. #define MACS(rt, ra, rb) MAC16(rt, ra, rb)
  479. /* signed 16x16 -> 32 multiply */
  480. #define MULS(ra, rb) MUL16(ra, rb)
  481. #define MLSS(rt, ra, rb) MLS16(rt, ra, rb)
  482. #else
  483. static inline int round_sample(int64_t *sum)
  484. {
  485. int sum1;
  486. sum1 = (int)((*sum) >> OUT_SHIFT);
  487. *sum &= (1<<OUT_SHIFT)-1;
  488. return av_clip(sum1, OUT_MIN, OUT_MAX);
  489. }
  490. # define MULS(ra, rb) MUL64(ra, rb)
  491. # define MACS(rt, ra, rb) MAC64(rt, ra, rb)
  492. # define MLSS(rt, ra, rb) MLS64(rt, ra, rb)
  493. #endif
  494. #define SUM8(op, sum, w, p) \
  495. { \
  496. op(sum, (w)[0 * 64], (p)[0 * 64]); \
  497. op(sum, (w)[1 * 64], (p)[1 * 64]); \
  498. op(sum, (w)[2 * 64], (p)[2 * 64]); \
  499. op(sum, (w)[3 * 64], (p)[3 * 64]); \
  500. op(sum, (w)[4 * 64], (p)[4 * 64]); \
  501. op(sum, (w)[5 * 64], (p)[5 * 64]); \
  502. op(sum, (w)[6 * 64], (p)[6 * 64]); \
  503. op(sum, (w)[7 * 64], (p)[7 * 64]); \
  504. }
  505. #define SUM8P2(sum1, op1, sum2, op2, w1, w2, p) \
  506. { \
  507. INTFLOAT tmp;\
  508. tmp = p[0 * 64];\
  509. op1(sum1, (w1)[0 * 64], tmp);\
  510. op2(sum2, (w2)[0 * 64], tmp);\
  511. tmp = p[1 * 64];\
  512. op1(sum1, (w1)[1 * 64], tmp);\
  513. op2(sum2, (w2)[1 * 64], tmp);\
  514. tmp = p[2 * 64];\
  515. op1(sum1, (w1)[2 * 64], tmp);\
  516. op2(sum2, (w2)[2 * 64], tmp);\
  517. tmp = p[3 * 64];\
  518. op1(sum1, (w1)[3 * 64], tmp);\
  519. op2(sum2, (w2)[3 * 64], tmp);\
  520. tmp = p[4 * 64];\
  521. op1(sum1, (w1)[4 * 64], tmp);\
  522. op2(sum2, (w2)[4 * 64], tmp);\
  523. tmp = p[5 * 64];\
  524. op1(sum1, (w1)[5 * 64], tmp);\
  525. op2(sum2, (w2)[5 * 64], tmp);\
  526. tmp = p[6 * 64];\
  527. op1(sum1, (w1)[6 * 64], tmp);\
  528. op2(sum2, (w2)[6 * 64], tmp);\
  529. tmp = p[7 * 64];\
  530. op1(sum1, (w1)[7 * 64], tmp);\
  531. op2(sum2, (w2)[7 * 64], tmp);\
  532. }
  533. void av_cold RENAME(ff_mpa_synth_init)(MPA_INT *window)
  534. {
  535. int i, j;
  536. /* max = 18760, max sum over all 16 coefs : 44736 */
  537. for(i=0;i<257;i++) {
  538. INTFLOAT v;
  539. v = ff_mpa_enwindow[i];
  540. #if CONFIG_FLOAT
  541. v *= 1.0 / (1LL<<(16 + FRAC_BITS));
  542. #elif WFRAC_BITS < 16
  543. v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
  544. #endif
  545. window[i] = v;
  546. if ((i & 63) != 0)
  547. v = -v;
  548. if (i != 0)
  549. window[512 - i] = v;
  550. }
  551. // Needed for avoiding shuffles in ASM implementations
  552. for(i=0; i < 8; i++)
  553. for(j=0; j < 16; j++)
  554. window[512+16*i+j] = window[64*i+32-j];
  555. for(i=0; i < 8; i++)
  556. for(j=0; j < 16; j++)
  557. window[512+128+16*i+j] = window[64*i+48-j];
  558. }
  559. static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
  560. int *dither_state, OUT_INT *samples, int incr)
  561. {
  562. register const MPA_INT *w, *w2, *p;
  563. int j;
  564. OUT_INT *samples2;
  565. #if CONFIG_FLOAT
  566. float sum, sum2;
  567. #elif FRAC_BITS <= 15
  568. int sum, sum2;
  569. #else
  570. int64_t sum, sum2;
  571. #endif
  572. /* copy to avoid wrap */
  573. memcpy(synth_buf + 512, synth_buf, 32 * sizeof(*synth_buf));
  574. samples2 = samples + 31 * incr;
  575. w = window;
  576. w2 = window + 31;
  577. sum = *dither_state;
  578. p = synth_buf + 16;
  579. SUM8(MACS, sum, w, p);
  580. p = synth_buf + 48;
  581. SUM8(MLSS, sum, w + 32, p);
  582. *samples = round_sample(&sum);
  583. samples += incr;
  584. w++;
  585. /* we calculate two samples at the same time to avoid one memory
  586. access per two sample */
  587. for(j=1;j<16;j++) {
  588. sum2 = 0;
  589. p = synth_buf + 16 + j;
  590. SUM8P2(sum, MACS, sum2, MLSS, w, w2, p);
  591. p = synth_buf + 48 - j;
  592. SUM8P2(sum, MLSS, sum2, MLSS, w + 32, w2 + 32, p);
  593. *samples = round_sample(&sum);
  594. samples += incr;
  595. sum += sum2;
  596. *samples2 = round_sample(&sum);
  597. samples2 -= incr;
  598. w++;
  599. w2--;
  600. }
  601. p = synth_buf + 32;
  602. SUM8(MLSS, sum, w + 32, p);
  603. *samples = round_sample(&sum);
  604. *dither_state= sum;
  605. }
  606. /* 32 sub band synthesis filter. Input: 32 sub band samples, Output:
  607. 32 samples. */
  608. /* XXX: optimize by avoiding ring buffer usage */
  609. #if !CONFIG_FLOAT
  610. void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
  611. MPA_INT *window, int *dither_state,
  612. OUT_INT *samples, int incr,
  613. INTFLOAT sb_samples[SBLIMIT])
  614. {
  615. register MPA_INT *synth_buf;
  616. int offset;
  617. #if FRAC_BITS <= 15
  618. int32_t tmp[32];
  619. int j;
  620. #endif
  621. offset = *synth_buf_offset;
  622. synth_buf = synth_buf_ptr + offset;
  623. #if FRAC_BITS <= 15
  624. dct32(tmp, sb_samples);
  625. for(j=0;j<32;j++) {
  626. /* NOTE: can cause a loss in precision if very high amplitude
  627. sound */
  628. synth_buf[j] = av_clip_int16(tmp[j]);
  629. }
  630. #else
  631. dct32(synth_buf, sb_samples);
  632. #endif
  633. apply_window_mp3_c(synth_buf, window, dither_state, samples, incr);
  634. offset = (offset - 32) & 511;
  635. *synth_buf_offset = offset;
  636. }
  637. #endif
  638. #define C3 FIXHR(0.86602540378443864676/2)
  639. /* 0.5 / cos(pi*(2*i+1)/36) */
  640. static const INTFLOAT icos36[9] = {
  641. FIXR(0.50190991877167369479),
  642. FIXR(0.51763809020504152469), //0
  643. FIXR(0.55168895948124587824),
  644. FIXR(0.61038729438072803416),
  645. FIXR(0.70710678118654752439), //1
  646. FIXR(0.87172339781054900991),
  647. FIXR(1.18310079157624925896),
  648. FIXR(1.93185165257813657349), //2
  649. FIXR(5.73685662283492756461),
  650. };
  651. /* 0.5 / cos(pi*(2*i+1)/36) */
  652. static const INTFLOAT icos36h[9] = {
  653. FIXHR(0.50190991877167369479/2),
  654. FIXHR(0.51763809020504152469/2), //0
  655. FIXHR(0.55168895948124587824/2),
  656. FIXHR(0.61038729438072803416/2),
  657. FIXHR(0.70710678118654752439/2), //1
  658. FIXHR(0.87172339781054900991/2),
  659. FIXHR(1.18310079157624925896/4),
  660. FIXHR(1.93185165257813657349/4), //2
  661. // FIXHR(5.73685662283492756461),
  662. };
  663. /* 12 points IMDCT. We compute it "by hand" by factorizing obvious
  664. cases. */
  665. static void imdct12(INTFLOAT *out, INTFLOAT *in)
  666. {
  667. INTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
  668. in0= in[0*3];
  669. in1= in[1*3] + in[0*3];
  670. in2= in[2*3] + in[1*3];
  671. in3= in[3*3] + in[2*3];
  672. in4= in[4*3] + in[3*3];
  673. in5= in[5*3] + in[4*3];
  674. in5 += in3;
  675. in3 += in1;
  676. in2= MULH3(in2, C3, 2);
  677. in3= MULH3(in3, C3, 4);
  678. t1 = in0 - in4;
  679. t2 = MULH3(in1 - in5, icos36h[4], 2);
  680. out[ 7]=
  681. out[10]= t1 + t2;
  682. out[ 1]=
  683. out[ 4]= t1 - t2;
  684. in0 += SHR(in4, 1);
  685. in4 = in0 + in2;
  686. in5 += 2*in1;
  687. in1 = MULH3(in5 + in3, icos36h[1], 1);
  688. out[ 8]=
  689. out[ 9]= in4 + in1;
  690. out[ 2]=
  691. out[ 3]= in4 - in1;
  692. in0 -= in2;
  693. in5 = MULH3(in5 - in3, icos36h[7], 2);
  694. out[ 0]=
  695. out[ 5]= in0 - in5;
  696. out[ 6]=
  697. out[11]= in0 + in5;
  698. }
  699. /* cos(pi*i/18) */
  700. #define C1 FIXHR(0.98480775301220805936/2)
  701. #define C2 FIXHR(0.93969262078590838405/2)
  702. #define C3 FIXHR(0.86602540378443864676/2)
  703. #define C4 FIXHR(0.76604444311897803520/2)
  704. #define C5 FIXHR(0.64278760968653932632/2)
  705. #define C6 FIXHR(0.5/2)
  706. #define C7 FIXHR(0.34202014332566873304/2)
  707. #define C8 FIXHR(0.17364817766693034885/2)
  708. /* using Lee like decomposition followed by hand coded 9 points DCT */
  709. static void imdct36(INTFLOAT *out, INTFLOAT *buf, INTFLOAT *in, INTFLOAT *win)
  710. {
  711. int i, j;
  712. INTFLOAT t0, t1, t2, t3, s0, s1, s2, s3;
  713. INTFLOAT tmp[18], *tmp1, *in1;
  714. for(i=17;i>=1;i--)
  715. in[i] += in[i-1];
  716. for(i=17;i>=3;i-=2)
  717. in[i] += in[i-2];
  718. for(j=0;j<2;j++) {
  719. tmp1 = tmp + j;
  720. in1 = in + j;
  721. t2 = in1[2*4] + in1[2*8] - in1[2*2];
  722. t3 = in1[2*0] + SHR(in1[2*6],1);
  723. t1 = in1[2*0] - in1[2*6];
  724. tmp1[ 6] = t1 - SHR(t2,1);
  725. tmp1[16] = t1 + t2;
  726. t0 = MULH3(in1[2*2] + in1[2*4] , C2, 2);
  727. t1 = MULH3(in1[2*4] - in1[2*8] , -2*C8, 1);
  728. t2 = MULH3(in1[2*2] + in1[2*8] , -C4, 2);
  729. tmp1[10] = t3 - t0 - t2;
  730. tmp1[ 2] = t3 + t0 + t1;
  731. tmp1[14] = t3 + t2 - t1;
  732. tmp1[ 4] = MULH3(in1[2*5] + in1[2*7] - in1[2*1], -C3, 2);
  733. t2 = MULH3(in1[2*1] + in1[2*5], C1, 2);
  734. t3 = MULH3(in1[2*5] - in1[2*7], -2*C7, 1);
  735. t0 = MULH3(in1[2*3], C3, 2);
  736. t1 = MULH3(in1[2*1] + in1[2*7], -C5, 2);
  737. tmp1[ 0] = t2 + t3 + t0;
  738. tmp1[12] = t2 + t1 - t0;
  739. tmp1[ 8] = t3 - t1 - t0;
  740. }
  741. i = 0;
  742. for(j=0;j<4;j++) {
  743. t0 = tmp[i];
  744. t1 = tmp[i + 2];
  745. s0 = t1 + t0;
  746. s2 = t1 - t0;
  747. t2 = tmp[i + 1];
  748. t3 = tmp[i + 3];
  749. s1 = MULH3(t3 + t2, icos36h[j], 2);
  750. s3 = MULLx(t3 - t2, icos36[8 - j], FRAC_BITS);
  751. t0 = s0 + s1;
  752. t1 = s0 - s1;
  753. out[(9 + j)*SBLIMIT] = MULH3(t1, win[9 + j], 1) + buf[9 + j];
  754. out[(8 - j)*SBLIMIT] = MULH3(t1, win[8 - j], 1) + buf[8 - j];
  755. buf[9 + j] = MULH3(t0, win[18 + 9 + j], 1);
  756. buf[8 - j] = MULH3(t0, win[18 + 8 - j], 1);
  757. t0 = s2 + s3;
  758. t1 = s2 - s3;
  759. out[(9 + 8 - j)*SBLIMIT] = MULH3(t1, win[9 + 8 - j], 1) + buf[9 + 8 - j];
  760. out[( j)*SBLIMIT] = MULH3(t1, win[ j], 1) + buf[ j];
  761. buf[9 + 8 - j] = MULH3(t0, win[18 + 9 + 8 - j], 1);
  762. buf[ + j] = MULH3(t0, win[18 + j], 1);
  763. i += 4;
  764. }
  765. s0 = tmp[16];
  766. s1 = MULH3(tmp[17], icos36h[4], 2);
  767. t0 = s0 + s1;
  768. t1 = s0 - s1;
  769. out[(9 + 4)*SBLIMIT] = MULH3(t1, win[9 + 4], 1) + buf[9 + 4];
  770. out[(8 - 4)*SBLIMIT] = MULH3(t1, win[8 - 4], 1) + buf[8 - 4];
  771. buf[9 + 4] = MULH3(t0, win[18 + 9 + 4], 1);
  772. buf[8 - 4] = MULH3(t0, win[18 + 8 - 4], 1);
  773. }
  774. /* return the number of decoded frames */
  775. static int mp_decode_layer1(MPADecodeContext *s)
  776. {
  777. int bound, i, v, n, ch, j, mant;
  778. uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
  779. uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
  780. if (s->mode == MPA_JSTEREO)
  781. bound = (s->mode_ext + 1) * 4;
  782. else
  783. bound = SBLIMIT;
  784. /* allocation bits */
  785. for(i=0;i<bound;i++) {
  786. for(ch=0;ch<s->nb_channels;ch++) {
  787. allocation[ch][i] = get_bits(&s->gb, 4);
  788. }
  789. }
  790. for(i=bound;i<SBLIMIT;i++) {
  791. allocation[0][i] = get_bits(&s->gb, 4);
  792. }
  793. /* scale factors */
  794. for(i=0;i<bound;i++) {
  795. for(ch=0;ch<s->nb_channels;ch++) {
  796. if (allocation[ch][i])
  797. scale_factors[ch][i] = get_bits(&s->gb, 6);
  798. }
  799. }
  800. for(i=bound;i<SBLIMIT;i++) {
  801. if (allocation[0][i]) {
  802. scale_factors[0][i] = get_bits(&s->gb, 6);
  803. scale_factors[1][i] = get_bits(&s->gb, 6);
  804. }
  805. }
  806. /* compute samples */
  807. for(j=0;j<12;j++) {
  808. for(i=0;i<bound;i++) {
  809. for(ch=0;ch<s->nb_channels;ch++) {
  810. n = allocation[ch][i];
  811. if (n) {
  812. mant = get_bits(&s->gb, n + 1);
  813. v = l1_unscale(n, mant, scale_factors[ch][i]);
  814. } else {
  815. v = 0;
  816. }
  817. s->sb_samples[ch][j][i] = v;
  818. }
  819. }
  820. for(i=bound;i<SBLIMIT;i++) {
  821. n = allocation[0][i];
  822. if (n) {
  823. mant = get_bits(&s->gb, n + 1);
  824. v = l1_unscale(n, mant, scale_factors[0][i]);
  825. s->sb_samples[0][j][i] = v;
  826. v = l1_unscale(n, mant, scale_factors[1][i]);
  827. s->sb_samples[1][j][i] = v;
  828. } else {
  829. s->sb_samples[0][j][i] = 0;
  830. s->sb_samples[1][j][i] = 0;
  831. }
  832. }
  833. }
  834. return 12;
  835. }
  836. static int mp_decode_layer2(MPADecodeContext *s)
  837. {
  838. int sblimit; /* number of used subbands */
  839. const unsigned char *alloc_table;
  840. int table, bit_alloc_bits, i, j, ch, bound, v;
  841. unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
  842. unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
  843. unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
  844. int scale, qindex, bits, steps, k, l, m, b;
  845. /* select decoding table */
  846. table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
  847. s->sample_rate, s->lsf);
  848. sblimit = ff_mpa_sblimit_table[table];
  849. alloc_table = ff_mpa_alloc_tables[table];
  850. if (s->mode == MPA_JSTEREO)
  851. bound = (s->mode_ext + 1) * 4;
  852. else
  853. bound = sblimit;
  854. dprintf(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
  855. /* sanity check */
  856. if( bound > sblimit ) bound = sblimit;
  857. /* parse bit allocation */
  858. j = 0;
  859. for(i=0;i<bound;i++) {
  860. bit_alloc_bits = alloc_table[j];
  861. for(ch=0;ch<s->nb_channels;ch++) {
  862. bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
  863. }
  864. j += 1 << bit_alloc_bits;
  865. }
  866. for(i=bound;i<sblimit;i++) {
  867. bit_alloc_bits = alloc_table[j];
  868. v = get_bits(&s->gb, bit_alloc_bits);
  869. bit_alloc[0][i] = v;
  870. bit_alloc[1][i] = v;
  871. j += 1 << bit_alloc_bits;
  872. }
  873. /* scale codes */
  874. for(i=0;i<sblimit;i++) {
  875. for(ch=0;ch<s->nb_channels;ch++) {
  876. if (bit_alloc[ch][i])
  877. scale_code[ch][i] = get_bits(&s->gb, 2);
  878. }
  879. }
  880. /* scale factors */
  881. for(i=0;i<sblimit;i++) {
  882. for(ch=0;ch<s->nb_channels;ch++) {
  883. if (bit_alloc[ch][i]) {
  884. sf = scale_factors[ch][i];
  885. switch(scale_code[ch][i]) {
  886. default:
  887. case 0:
  888. sf[0] = get_bits(&s->gb, 6);
  889. sf[1] = get_bits(&s->gb, 6);
  890. sf[2] = get_bits(&s->gb, 6);
  891. break;
  892. case 2:
  893. sf[0] = get_bits(&s->gb, 6);
  894. sf[1] = sf[0];
  895. sf[2] = sf[0];
  896. break;
  897. case 1:
  898. sf[0] = get_bits(&s->gb, 6);
  899. sf[2] = get_bits(&s->gb, 6);
  900. sf[1] = sf[0];
  901. break;
  902. case 3:
  903. sf[0] = get_bits(&s->gb, 6);
  904. sf[2] = get_bits(&s->gb, 6);
  905. sf[1] = sf[2];
  906. break;
  907. }
  908. }
  909. }
  910. }
  911. /* samples */
  912. for(k=0;k<3;k++) {
  913. for(l=0;l<12;l+=3) {
  914. j = 0;
  915. for(i=0;i<bound;i++) {
  916. bit_alloc_bits = alloc_table[j];
  917. for(ch=0;ch<s->nb_channels;ch++) {
  918. b = bit_alloc[ch][i];
  919. if (b) {
  920. scale = scale_factors[ch][i][k];
  921. qindex = alloc_table[j+b];
  922. bits = ff_mpa_quant_bits[qindex];
  923. if (bits < 0) {
  924. int v2;
  925. /* 3 values at the same time */
  926. v = get_bits(&s->gb, -bits);
  927. v2 = division_tabs[qindex][v];
  928. steps = ff_mpa_quant_steps[qindex];
  929. s->sb_samples[ch][k * 12 + l + 0][i] =
  930. l2_unscale_group(steps, v2 & 15, scale);
  931. s->sb_samples[ch][k * 12 + l + 1][i] =
  932. l2_unscale_group(steps, (v2 >> 4) & 15, scale);
  933. s->sb_samples[ch][k * 12 + l + 2][i] =
  934. l2_unscale_group(steps, v2 >> 8 , scale);
  935. } else {
  936. for(m=0;m<3;m++) {
  937. v = get_bits(&s->gb, bits);
  938. v = l1_unscale(bits - 1, v, scale);
  939. s->sb_samples[ch][k * 12 + l + m][i] = v;
  940. }
  941. }
  942. } else {
  943. s->sb_samples[ch][k * 12 + l + 0][i] = 0;
  944. s->sb_samples[ch][k * 12 + l + 1][i] = 0;
  945. s->sb_samples[ch][k * 12 + l + 2][i] = 0;
  946. }
  947. }
  948. /* next subband in alloc table */
  949. j += 1 << bit_alloc_bits;
  950. }
  951. /* XXX: find a way to avoid this duplication of code */
  952. for(i=bound;i<sblimit;i++) {
  953. bit_alloc_bits = alloc_table[j];
  954. b = bit_alloc[0][i];
  955. if (b) {
  956. int mant, scale0, scale1;
  957. scale0 = scale_factors[0][i][k];
  958. scale1 = scale_factors[1][i][k];
  959. qindex = alloc_table[j+b];
  960. bits = ff_mpa_quant_bits[qindex];
  961. if (bits < 0) {
  962. /* 3 values at the same time */
  963. v = get_bits(&s->gb, -bits);
  964. steps = ff_mpa_quant_steps[qindex];
  965. mant = v % steps;
  966. v = v / steps;
  967. s->sb_samples[0][k * 12 + l + 0][i] =
  968. l2_unscale_group(steps, mant, scale0);
  969. s->sb_samples[1][k * 12 + l + 0][i] =
  970. l2_unscale_group(steps, mant, scale1);
  971. mant = v % steps;
  972. v = v / steps;
  973. s->sb_samples[0][k * 12 + l + 1][i] =
  974. l2_unscale_group(steps, mant, scale0);
  975. s->sb_samples[1][k * 12 + l + 1][i] =
  976. l2_unscale_group(steps, mant, scale1);
  977. s->sb_samples[0][k * 12 + l + 2][i] =
  978. l2_unscale_group(steps, v, scale0);
  979. s->sb_samples[1][k * 12 + l + 2][i] =
  980. l2_unscale_group(steps, v, scale1);
  981. } else {
  982. for(m=0;m<3;m++) {
  983. mant = get_bits(&s->gb, bits);
  984. s->sb_samples[0][k * 12 + l + m][i] =
  985. l1_unscale(bits - 1, mant, scale0);
  986. s->sb_samples[1][k * 12 + l + m][i] =
  987. l1_unscale(bits - 1, mant, scale1);
  988. }
  989. }
  990. } else {
  991. s->sb_samples[0][k * 12 + l + 0][i] = 0;
  992. s->sb_samples[0][k * 12 + l + 1][i] = 0;
  993. s->sb_samples[0][k * 12 + l + 2][i] = 0;
  994. s->sb_samples[1][k * 12 + l + 0][i] = 0;
  995. s->sb_samples[1][k * 12 + l + 1][i] = 0;
  996. s->sb_samples[1][k * 12 + l + 2][i] = 0;
  997. }
  998. /* next subband in alloc table */
  999. j += 1 << bit_alloc_bits;
  1000. }
  1001. /* fill remaining samples to zero */
  1002. for(i=sblimit;i<SBLIMIT;i++) {
  1003. for(ch=0;ch<s->nb_channels;ch++) {
  1004. s->sb_samples[ch][k * 12 + l + 0][i] = 0;
  1005. s->sb_samples[ch][k * 12 + l + 1][i] = 0;
  1006. s->sb_samples[ch][k * 12 + l + 2][i] = 0;
  1007. }
  1008. }
  1009. }
  1010. }
  1011. return 3 * 12;
  1012. }
  1013. #define SPLIT(dst,sf,n)\
  1014. if(n==3){\
  1015. int m= (sf*171)>>9;\
  1016. dst= sf - 3*m;\
  1017. sf=m;\
  1018. }else if(n==4){\
  1019. dst= sf&3;\
  1020. sf>>=2;\
  1021. }else if(n==5){\
  1022. int m= (sf*205)>>10;\
  1023. dst= sf - 5*m;\
  1024. sf=m;\
  1025. }else if(n==6){\
  1026. int m= (sf*171)>>10;\
  1027. dst= sf - 6*m;\
  1028. sf=m;\
  1029. }else{\
  1030. dst=0;\
  1031. }
  1032. static av_always_inline void lsf_sf_expand(int *slen,
  1033. int sf, int n1, int n2, int n3)
  1034. {
  1035. SPLIT(slen[3], sf, n3)
  1036. SPLIT(slen[2], sf, n2)
  1037. SPLIT(slen[1], sf, n1)
  1038. slen[0] = sf;
  1039. }
  1040. static void exponents_from_scale_factors(MPADecodeContext *s,
  1041. GranuleDef *g,
  1042. int16_t *exponents)
  1043. {
  1044. const uint8_t *bstab, *pretab;
  1045. int len, i, j, k, l, v0, shift, gain, gains[3];
  1046. int16_t *exp_ptr;
  1047. exp_ptr = exponents;
  1048. gain = g->global_gain - 210;
  1049. shift = g->scalefac_scale + 1;
  1050. bstab = band_size_long[s->sample_rate_index];
  1051. pretab = mpa_pretab[g->preflag];
  1052. for(i=0;i<g->long_end;i++) {
  1053. v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
  1054. len = bstab[i];
  1055. for(j=len;j>0;j--)
  1056. *exp_ptr++ = v0;
  1057. }
  1058. if (g->short_start < 13) {
  1059. bstab = band_size_short[s->sample_rate_index];
  1060. gains[0] = gain - (g->subblock_gain[0] << 3);
  1061. gains[1] = gain - (g->subblock_gain[1] << 3);
  1062. gains[2] = gain - (g->subblock_gain[2] << 3);
  1063. k = g->long_end;
  1064. for(i=g->short_start;i<13;i++) {
  1065. len = bstab[i];
  1066. for(l=0;l<3;l++) {
  1067. v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
  1068. for(j=len;j>0;j--)
  1069. *exp_ptr++ = v0;
  1070. }
  1071. }
  1072. }
  1073. }
  1074. /* handle n = 0 too */
  1075. static inline int get_bitsz(GetBitContext *s, int n)
  1076. {
  1077. if (n == 0)
  1078. return 0;
  1079. else
  1080. return get_bits(s, n);
  1081. }
  1082. static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos, int *end_pos2){
  1083. if(s->in_gb.buffer && *pos >= s->gb.size_in_bits){
  1084. s->gb= s->in_gb;
  1085. s->in_gb.buffer=NULL;
  1086. assert((get_bits_count(&s->gb) & 7) == 0);
  1087. skip_bits_long(&s->gb, *pos - *end_pos);
  1088. *end_pos2=
  1089. *end_pos= *end_pos2 + get_bits_count(&s->gb) - *pos;
  1090. *pos= get_bits_count(&s->gb);
  1091. }
  1092. }
  1093. /* Following is a optimized code for
  1094. INTFLOAT v = *src
  1095. if(get_bits1(&s->gb))
  1096. v = -v;
  1097. *dst = v;
  1098. */
  1099. #if CONFIG_FLOAT
  1100. #define READ_FLIP_SIGN(dst,src)\
  1101. v = AV_RN32A(src) ^ (get_bits1(&s->gb)<<31);\
  1102. AV_WN32A(dst, v);
  1103. #else
  1104. #define READ_FLIP_SIGN(dst,src)\
  1105. v= -get_bits1(&s->gb);\
  1106. *(dst) = (*(src) ^ v) - v;
  1107. #endif
  1108. static int huffman_decode(MPADecodeContext *s, GranuleDef *g,
  1109. int16_t *exponents, int end_pos2)
  1110. {
  1111. int s_index;
  1112. int i;
  1113. int last_pos, bits_left;
  1114. VLC *vlc;
  1115. int end_pos= FFMIN(end_pos2, s->gb.size_in_bits);
  1116. /* low frequencies (called big values) */
  1117. s_index = 0;
  1118. for(i=0;i<3;i++) {
  1119. int j, k, l, linbits;
  1120. j = g->region_size[i];
  1121. if (j == 0)
  1122. continue;
  1123. /* select vlc table */
  1124. k = g->table_select[i];
  1125. l = mpa_huff_data[k][0];
  1126. linbits = mpa_huff_data[k][1];
  1127. vlc = &huff_vlc[l];
  1128. if(!l){
  1129. memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid)*2*j);
  1130. s_index += 2*j;
  1131. continue;
  1132. }
  1133. /* read huffcode and compute each couple */
  1134. for(;j>0;j--) {
  1135. int exponent, x, y;
  1136. int v;
  1137. int pos= get_bits_count(&s->gb);
  1138. if (pos >= end_pos){
  1139. // av_log(NULL, AV_LOG_ERROR, "pos: %d %d %d %d\n", pos, end_pos, end_pos2, s_index);
  1140. switch_buffer(s, &pos, &end_pos, &end_pos2);
  1141. // av_log(NULL, AV_LOG_ERROR, "new pos: %d %d\n", pos, end_pos);
  1142. if(pos >= end_pos)
  1143. break;
  1144. }
  1145. y = get_vlc2(&s->gb, vlc->table, 7, 3);
  1146. if(!y){
  1147. g->sb_hybrid[s_index ] =
  1148. g->sb_hybrid[s_index+1] = 0;
  1149. s_index += 2;
  1150. continue;
  1151. }
  1152. exponent= exponents[s_index];
  1153. dprintf(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n",
  1154. i, g->region_size[i] - j, x, y, exponent);
  1155. if(y&16){
  1156. x = y >> 5;
  1157. y = y & 0x0f;
  1158. if (x < 15){
  1159. READ_FLIP_SIGN(g->sb_hybrid+s_index, RENAME(expval_table)[ exponent ]+x)
  1160. }else{
  1161. x += get_bitsz(&s->gb, linbits);
  1162. v = l3_unscale(x, exponent);
  1163. if (get_bits1(&s->gb))
  1164. v = -v;
  1165. g->sb_hybrid[s_index] = v;
  1166. }
  1167. if (y < 15){
  1168. READ_FLIP_SIGN(g->sb_hybrid+s_index+1, RENAME(expval_table)[ exponent ]+y)
  1169. }else{
  1170. y += get_bitsz(&s->gb, linbits);
  1171. v = l3_unscale(y, exponent);
  1172. if (get_bits1(&s->gb))
  1173. v = -v;
  1174. g->sb_hybrid[s_index+1] = v;
  1175. }
  1176. }else{
  1177. x = y >> 5;
  1178. y = y & 0x0f;
  1179. x += y;
  1180. if (x < 15){
  1181. READ_FLIP_SIGN(g->sb_hybrid+s_index+!!y, RENAME(expval_table)[ exponent ]+x)
  1182. }else{
  1183. x += get_bitsz(&s->gb, linbits);
  1184. v = l3_unscale(x, exponent);
  1185. if (get_bits1(&s->gb))
  1186. v = -v;
  1187. g->sb_hybrid[s_index+!!y] = v;
  1188. }
  1189. g->sb_hybrid[s_index+ !y] = 0;
  1190. }
  1191. s_index+=2;
  1192. }
  1193. }
  1194. /* high frequencies */
  1195. vlc = &huff_quad_vlc[g->count1table_select];
  1196. last_pos=0;
  1197. while (s_index <= 572) {
  1198. int pos, code;
  1199. pos = get_bits_count(&s->gb);
  1200. if (pos >= end_pos) {
  1201. if (pos > end_pos2 && last_pos){
  1202. /* some encoders generate an incorrect size for this
  1203. part. We must go back into the data */
  1204. s_index -= 4;
  1205. skip_bits_long(&s->gb, last_pos - pos);
  1206. av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
  1207. if(s->error_recognition >= FF_ER_COMPLIANT)
  1208. s_index=0;
  1209. break;
  1210. }
  1211. // av_log(NULL, AV_LOG_ERROR, "pos2: %d %d %d %d\n", pos, end_pos, end_pos2, s_index);
  1212. switch_buffer(s, &pos, &end_pos, &end_pos2);
  1213. // av_log(NULL, AV_LOG_ERROR, "new pos2: %d %d %d\n", pos, end_pos, s_index);
  1214. if(pos >= end_pos)
  1215. break;
  1216. }
  1217. last_pos= pos;
  1218. code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
  1219. dprintf(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
  1220. g->sb_hybrid[s_index+0]=
  1221. g->sb_hybrid[s_index+1]=
  1222. g->sb_hybrid[s_index+2]=
  1223. g->sb_hybrid[s_index+3]= 0;
  1224. while(code){
  1225. static const int idxtab[16]={3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0};
  1226. int v;
  1227. int pos= s_index+idxtab[code];
  1228. code ^= 8>>idxtab[code];
  1229. READ_FLIP_SIGN(g->sb_hybrid+pos, RENAME(exp_table)+exponents[pos])
  1230. }
  1231. s_index+=4;
  1232. }
  1233. /* skip extension bits */
  1234. bits_left = end_pos2 - get_bits_count(&s->gb);
  1235. //av_log(NULL, AV_LOG_ERROR, "left:%d buf:%p\n", bits_left, s->in_gb.buffer);
  1236. if (bits_left < 0 && s->error_recognition >= FF_ER_COMPLIANT) {
  1237. av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
  1238. s_index=0;
  1239. }else if(bits_left > 0 && s->error_recognition >= FF_ER_AGGRESSIVE){
  1240. av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
  1241. s_index=0;
  1242. }
  1243. memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid)*(576 - s_index));
  1244. skip_bits_long(&s->gb, bits_left);
  1245. i= get_bits_count(&s->gb);
  1246. switch_buffer(s, &i, &end_pos, &end_pos2);
  1247. return 0;
  1248. }
  1249. /* Reorder short blocks from bitstream order to interleaved order. It
  1250. would be faster to do it in parsing, but the code would be far more
  1251. complicated */
  1252. static void reorder_block(MPADecodeContext *s, GranuleDef *g)
  1253. {
  1254. int i, j, len;
  1255. INTFLOAT *ptr, *dst, *ptr1;
  1256. INTFLOAT tmp[576];
  1257. if (g->block_type != 2)
  1258. return;
  1259. if (g->switch_point) {
  1260. if (s->sample_rate_index != 8) {
  1261. ptr = g->sb_hybrid + 36;
  1262. } else {
  1263. ptr = g->sb_hybrid + 48;
  1264. }
  1265. } else {
  1266. ptr = g->sb_hybrid;
  1267. }
  1268. for(i=g->short_start;i<13;i++) {
  1269. len = band_size_short[s->sample_rate_index][i];
  1270. ptr1 = ptr;
  1271. dst = tmp;
  1272. for(j=len;j>0;j--) {
  1273. *dst++ = ptr[0*len];
  1274. *dst++ = ptr[1*len];
  1275. *dst++ = ptr[2*len];
  1276. ptr++;
  1277. }
  1278. ptr+=2*len;
  1279. memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
  1280. }
  1281. }
  1282. #define ISQRT2 FIXR(0.70710678118654752440)
  1283. static void compute_stereo(MPADecodeContext *s,
  1284. GranuleDef *g0, GranuleDef *g1)
  1285. {
  1286. int i, j, k, l;
  1287. int sf_max, sf, len, non_zero_found;
  1288. INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;
  1289. int non_zero_found_short[3];
  1290. /* intensity stereo */
  1291. if (s->mode_ext & MODE_EXT_I_STEREO) {
  1292. if (!s->lsf) {
  1293. is_tab = is_table;
  1294. sf_max = 7;
  1295. } else {
  1296. is_tab = is_table_lsf[g1->scalefac_compress & 1];
  1297. sf_max = 16;
  1298. }
  1299. tab0 = g0->sb_hybrid + 576;
  1300. tab1 = g1->sb_hybrid + 576;
  1301. non_zero_found_short[0] = 0;
  1302. non_zero_found_short[1] = 0;
  1303. non_zero_found_short[2] = 0;
  1304. k = (13 - g1->short_start) * 3 + g1->long_end - 3;
  1305. for(i = 12;i >= g1->short_start;i--) {
  1306. /* for last band, use previous scale factor */
  1307. if (i != 11)
  1308. k -= 3;
  1309. len = band_size_short[s->sample_rate_index][i];
  1310. for(l=2;l>=0;l--) {
  1311. tab0 -= len;
  1312. tab1 -= len;
  1313. if (!non_zero_found_short[l]) {
  1314. /* test if non zero band. if so, stop doing i-stereo */
  1315. for(j=0;j<len;j++) {
  1316. if (tab1[j] != 0) {
  1317. non_zero_found_short[l] = 1;
  1318. goto found1;
  1319. }
  1320. }
  1321. sf = g1->scale_factors[k + l];
  1322. if (sf >= sf_max)
  1323. goto found1;
  1324. v1 = is_tab[0][sf];
  1325. v2 = is_tab[1][sf];
  1326. for(j=0;j<len;j++) {
  1327. tmp0 = tab0[j];
  1328. tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
  1329. tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
  1330. }
  1331. } else {
  1332. found1:
  1333. if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1334. /* lower part of the spectrum : do ms stereo
  1335. if enabled */
  1336. for(j=0;j<len;j++) {
  1337. tmp0 = tab0[j];
  1338. tmp1 = tab1[j];
  1339. tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
  1340. tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
  1341. }
  1342. }
  1343. }
  1344. }
  1345. }
  1346. non_zero_found = non_zero_found_short[0] |
  1347. non_zero_found_short[1] |
  1348. non_zero_found_short[2];
  1349. for(i = g1->long_end - 1;i >= 0;i--) {
  1350. len = band_size_long[s->sample_rate_index][i];
  1351. tab0 -= len;
  1352. tab1 -= len;
  1353. /* test if non zero band. if so, stop doing i-stereo */
  1354. if (!non_zero_found) {
  1355. for(j=0;j<len;j++) {
  1356. if (tab1[j] != 0) {
  1357. non_zero_found = 1;
  1358. goto found2;
  1359. }
  1360. }
  1361. /* for last band, use previous scale factor */
  1362. k = (i == 21) ? 20 : i;
  1363. sf = g1->scale_factors[k];
  1364. if (sf >= sf_max)
  1365. goto found2;
  1366. v1 = is_tab[0][sf];
  1367. v2 = is_tab[1][sf];
  1368. for(j=0;j<len;j++) {
  1369. tmp0 = tab0[j];
  1370. tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
  1371. tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
  1372. }
  1373. } else {
  1374. found2:
  1375. if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1376. /* lower part of the spectrum : do ms stereo
  1377. if enabled */
  1378. for(j=0;j<len;j++) {
  1379. tmp0 = tab0[j];
  1380. tmp1 = tab1[j];
  1381. tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
  1382. tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
  1383. }
  1384. }
  1385. }
  1386. }
  1387. } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1388. /* ms stereo ONLY */
  1389. /* NOTE: the 1/sqrt(2) normalization factor is included in the
  1390. global gain */
  1391. tab0 = g0->sb_hybrid;
  1392. tab1 = g1->sb_hybrid;
  1393. for(i=0;i<576;i++) {
  1394. tmp0 = tab0[i];
  1395. tmp1 = tab1[i];
  1396. tab0[i] = tmp0 + tmp1;
  1397. tab1[i] = tmp0 - tmp1;
  1398. }
  1399. }
  1400. }
  1401. #if !CONFIG_FLOAT
  1402. static void compute_antialias_integer(MPADecodeContext *s,
  1403. GranuleDef *g)
  1404. {
  1405. int32_t *ptr, *csa;
  1406. int n, i;
  1407. /* we antialias only "long" bands */
  1408. if (g->block_type == 2) {
  1409. if (!g->switch_point)
  1410. return;
  1411. /* XXX: check this for 8000Hz case */
  1412. n = 1;
  1413. } else {
  1414. n = SBLIMIT - 1;
  1415. }
  1416. ptr = g->sb_hybrid + 18;
  1417. for(i = n;i > 0;i--) {
  1418. int tmp0, tmp1, tmp2;
  1419. csa = &csa_table[0][0];
  1420. #define INT_AA(j) \
  1421. tmp0 = ptr[-1-j];\
  1422. tmp1 = ptr[ j];\
  1423. tmp2= MULH(tmp0 + tmp1, csa[0+4*j]);\
  1424. ptr[-1-j] = 4*(tmp2 - MULH(tmp1, csa[2+4*j]));\
  1425. ptr[ j] = 4*(tmp2 + MULH(tmp0, csa[3+4*j]));
  1426. INT_AA(0)
  1427. INT_AA(1)
  1428. INT_AA(2)
  1429. INT_AA(3)
  1430. INT_AA(4)
  1431. INT_AA(5)
  1432. INT_AA(6)
  1433. INT_AA(7)
  1434. ptr += 18;
  1435. }
  1436. }
  1437. #endif
  1438. static void compute_imdct(MPADecodeContext *s,
  1439. GranuleDef *g,
  1440. INTFLOAT *sb_samples,
  1441. INTFLOAT *mdct_buf)
  1442. {
  1443. INTFLOAT *win, *win1, *out_ptr, *ptr, *buf, *ptr1;
  1444. INTFLOAT out2[12];
  1445. int i, j, mdct_long_end, sblimit;
  1446. /* find last non zero block */
  1447. ptr = g->sb_hybrid + 576;
  1448. ptr1 = g->sb_hybrid + 2 * 18;
  1449. while (ptr >= ptr1) {
  1450. int32_t *p;
  1451. ptr -= 6;
  1452. p= (int32_t*)ptr;
  1453. if(p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
  1454. break;
  1455. }
  1456. sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
  1457. if (g->block_type == 2) {
  1458. /* XXX: check for 8000 Hz */
  1459. if (g->switch_point)
  1460. mdct_long_end = 2;
  1461. else
  1462. mdct_long_end = 0;
  1463. } else {
  1464. mdct_long_end = sblimit;
  1465. }
  1466. buf = mdct_buf;
  1467. ptr = g->sb_hybrid;
  1468. for(j=0;j<mdct_long_end;j++) {
  1469. /* apply window & overlap with previous buffer */
  1470. out_ptr = sb_samples + j;
  1471. /* select window */
  1472. if (g->switch_point && j < 2)
  1473. win1 = mdct_win[0];
  1474. else
  1475. win1 = mdct_win[g->block_type];
  1476. /* select frequency inversion */
  1477. win = win1 + ((4 * 36) & -(j & 1));
  1478. imdct36(out_ptr, buf, ptr, win);
  1479. out_ptr += 18*SBLIMIT;
  1480. ptr += 18;
  1481. buf += 18;
  1482. }
  1483. for(j=mdct_long_end;j<sblimit;j++) {
  1484. /* select frequency inversion */
  1485. win = mdct_win[2] + ((4 * 36) & -(j & 1));
  1486. out_ptr = sb_samples + j;
  1487. for(i=0; i<6; i++){
  1488. *out_ptr = buf[i];
  1489. out_ptr += SBLIMIT;
  1490. }
  1491. imdct12(out2, ptr + 0);
  1492. for(i=0;i<6;i++) {
  1493. *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[i + 6*1];
  1494. buf[i + 6*2] = MULH3(out2[i + 6], win[i + 6], 1);
  1495. out_ptr += SBLIMIT;
  1496. }
  1497. imdct12(out2, ptr + 1);
  1498. for(i=0;i<6;i++) {
  1499. *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[i + 6*2];
  1500. buf[i + 6*0] = MULH3(out2[i + 6], win[i + 6], 1);
  1501. out_ptr += SBLIMIT;
  1502. }
  1503. imdct12(out2, ptr + 2);
  1504. for(i=0;i<6;i++) {
  1505. buf[i + 6*0] = MULH3(out2[i ], win[i ], 1) + buf[i + 6*0];
  1506. buf[i + 6*1] = MULH3(out2[i + 6], win[i + 6], 1);
  1507. buf[i + 6*2] = 0;
  1508. }
  1509. ptr += 18;
  1510. buf += 18;
  1511. }
  1512. /* zero bands */
  1513. for(j=sblimit;j<SBLIMIT;j++) {
  1514. /* overlap */
  1515. out_ptr = sb_samples + j;
  1516. for(i=0;i<18;i++) {
  1517. *out_ptr = buf[i];
  1518. buf[i] = 0;
  1519. out_ptr += SBLIMIT;
  1520. }
  1521. buf += 18;
  1522. }
  1523. }
  1524. /* main layer3 decoding function */
  1525. static int mp_decode_layer3(MPADecodeContext *s)
  1526. {
  1527. int nb_granules, main_data_begin, private_bits;
  1528. int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
  1529. GranuleDef *g;
  1530. int16_t exponents[576]; //FIXME try INTFLOAT
  1531. /* read side info */
  1532. if (s->lsf) {
  1533. main_data_begin = get_bits(&s->gb, 8);
  1534. private_bits = get_bits(&s->gb, s->nb_channels);
  1535. nb_granules = 1;
  1536. } else {
  1537. main_data_begin = get_bits(&s->gb, 9);
  1538. if (s->nb_channels == 2)
  1539. private_bits = get_bits(&s->gb, 3);
  1540. else
  1541. private_bits = get_bits(&s->gb, 5);
  1542. nb_granules = 2;
  1543. for(ch=0;ch<s->nb_channels;ch++) {
  1544. s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
  1545. s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
  1546. }
  1547. }
  1548. for(gr=0;gr<nb_granules;gr++) {
  1549. for(ch=0;ch<s->nb_channels;ch++) {
  1550. dprintf(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
  1551. g = &s->granules[ch][gr];
  1552. g->part2_3_length = get_bits(&s->gb, 12);
  1553. g->big_values = get_bits(&s->gb, 9);
  1554. if(g->big_values > 288){
  1555. av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
  1556. return -1;
  1557. }
  1558. g->global_gain = get_bits(&s->gb, 8);
  1559. /* if MS stereo only is selected, we precompute the
  1560. 1/sqrt(2) renormalization factor */
  1561. if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
  1562. MODE_EXT_MS_STEREO)
  1563. g->global_gain -= 2;
  1564. if (s->lsf)
  1565. g->scalefac_compress = get_bits(&s->gb, 9);
  1566. else
  1567. g->scalefac_compress = get_bits(&s->gb, 4);
  1568. blocksplit_flag = get_bits1(&s->gb);
  1569. if (blocksplit_flag) {
  1570. g->block_type = get_bits(&s->gb, 2);
  1571. if (g->block_type == 0){
  1572. av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
  1573. return -1;
  1574. }
  1575. g->switch_point = get_bits1(&s->gb);
  1576. for(i=0;i<2;i++)
  1577. g->table_select[i] = get_bits(&s->gb, 5);
  1578. for(i=0;i<3;i++)
  1579. g->subblock_gain[i] = get_bits(&s->gb, 3);
  1580. ff_init_short_region(s, g);
  1581. } else {
  1582. int region_address1, region_address2;
  1583. g->block_type = 0;
  1584. g->switch_point = 0;
  1585. for(i=0;i<3;i++)
  1586. g->table_select[i] = get_bits(&s->gb, 5);
  1587. /* compute huffman coded region sizes */
  1588. region_address1 = get_bits(&s->gb, 4);
  1589. region_address2 = get_bits(&s->gb, 3);
  1590. dprintf(s->avctx, "region1=%d region2=%d\n",
  1591. region_address1, region_address2);
  1592. ff_init_long_region(s, g, region_address1, region_address2);
  1593. }
  1594. ff_region_offset2size(g);
  1595. ff_compute_band_indexes(s, g);
  1596. g->preflag = 0;
  1597. if (!s->lsf)
  1598. g->preflag = get_bits1(&s->gb);
  1599. g->scalefac_scale = get_bits1(&s->gb);
  1600. g->count1table_select = get_bits1(&s->gb);
  1601. dprintf(s->avctx, "block_type=%d switch_point=%d\n",
  1602. g->block_type, g->switch_point);
  1603. }
  1604. }
  1605. if (!s->adu_mode) {
  1606. const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
  1607. assert((get_bits_count(&s->gb) & 7) == 0);
  1608. /* now we get bits from the main_data_begin offset */
  1609. dprintf(s->avctx, "seekback: %d\n", main_data_begin);
  1610. //av_log(NULL, AV_LOG_ERROR, "backstep:%d, lastbuf:%d\n", main_data_begin, s->last_buf_size);
  1611. memcpy(s->last_buf + s->last_buf_size, ptr, EXTRABYTES);
  1612. s->in_gb= s->gb;
  1613. init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);
  1614. skip_bits_long(&s->gb, 8*(s->last_buf_size - main_data_begin));
  1615. }
  1616. for(gr=0;gr<nb_granules;gr++) {
  1617. for(ch=0;ch<s->nb_channels;ch++) {
  1618. g = &s->granules[ch][gr];
  1619. if(get_bits_count(&s->gb)<0){
  1620. av_log(s->avctx, AV_LOG_DEBUG, "mdb:%d, lastbuf:%d skipping granule %d\n",
  1621. main_data_begin, s->last_buf_size, gr);
  1622. skip_bits_long(&s->gb, g->part2_3_length);
  1623. memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
  1624. if(get_bits_count(&s->gb) >= s->gb.size_in_bits && s->in_gb.buffer){
  1625. skip_bits_long(&s->in_gb, get_bits_count(&s->gb) - s->gb.size_in_bits);
  1626. s->gb= s->in_gb;
  1627. s->in_gb.buffer=NULL;
  1628. }
  1629. continue;
  1630. }
  1631. bits_pos = get_bits_count(&s->gb);
  1632. if (!s->lsf) {
  1633. uint8_t *sc;
  1634. int slen, slen1, slen2;
  1635. /* MPEG1 scale factors */
  1636. slen1 = slen_table[0][g->scalefac_compress];
  1637. slen2 = slen_table[1][g->scalefac_compress];
  1638. dprintf(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
  1639. if (g->block_type == 2) {
  1640. n = g->switch_point ? 17 : 18;
  1641. j = 0;
  1642. if(slen1){
  1643. for(i=0;i<n;i++)
  1644. g->scale_factors[j++] = get_bits(&s->gb, slen1);
  1645. }else{
  1646. for(i=0;i<n;i++)
  1647. g->scale_factors[j++] = 0;
  1648. }
  1649. if(slen2){
  1650. for(i=0;i<18;i++)
  1651. g->scale_factors[j++] = get_bits(&s->gb, slen2);
  1652. for(i=0;i<3;i++)
  1653. g->scale_factors[j++] = 0;
  1654. }else{
  1655. for(i=0;i<21;i++)
  1656. g->scale_factors[j++] = 0;
  1657. }
  1658. } else {
  1659. sc = s->granules[ch][0].scale_factors;
  1660. j = 0;
  1661. for(k=0;k<4;k++) {
  1662. n = (k == 0 ? 6 : 5);
  1663. if ((g->scfsi & (0x8 >> k)) == 0) {
  1664. slen = (k < 2) ? slen1 : slen2;
  1665. if(slen){
  1666. for(i=0;i<n;i++)
  1667. g->scale_factors[j++] = get_bits(&s->gb, slen);
  1668. }else{
  1669. for(i=0;i<n;i++)
  1670. g->scale_factors[j++] = 0;
  1671. }
  1672. } else {
  1673. /* simply copy from last granule */
  1674. for(i=0;i<n;i++) {
  1675. g->scale_factors[j] = sc[j];
  1676. j++;
  1677. }
  1678. }
  1679. }
  1680. g->scale_factors[j++] = 0;
  1681. }
  1682. } else {
  1683. int tindex, tindex2, slen[4], sl, sf;
  1684. /* LSF scale factors */
  1685. if (g->block_type == 2) {
  1686. tindex = g->switch_point ? 2 : 1;
  1687. } else {
  1688. tindex = 0;
  1689. }
  1690. sf = g->scalefac_compress;
  1691. if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
  1692. /* intensity stereo case */
  1693. sf >>= 1;
  1694. if (sf < 180) {
  1695. lsf_sf_expand(slen, sf, 6, 6, 0);
  1696. tindex2 = 3;
  1697. } else if (sf < 244) {
  1698. lsf_sf_expand(slen, sf - 180, 4, 4, 0);
  1699. tindex2 = 4;
  1700. } else {
  1701. lsf_sf_expand(slen, sf - 244, 3, 0, 0);
  1702. tindex2 = 5;
  1703. }
  1704. } else {
  1705. /* normal case */
  1706. if (sf < 400) {
  1707. lsf_sf_expand(slen, sf, 5, 4, 4);
  1708. tindex2 = 0;
  1709. } else if (sf < 500) {
  1710. lsf_sf_expand(slen, sf - 400, 5, 4, 0);
  1711. tindex2 = 1;
  1712. } else {
  1713. lsf_sf_expand(slen, sf - 500, 3, 0, 0);
  1714. tindex2 = 2;
  1715. g->preflag = 1;
  1716. }
  1717. }
  1718. j = 0;
  1719. for(k=0;k<4;k++) {
  1720. n = lsf_nsf_table[tindex2][tindex][k];
  1721. sl = slen[k];
  1722. if(sl){
  1723. for(i=0;i<n;i++)
  1724. g->scale_factors[j++] = get_bits(&s->gb, sl);
  1725. }else{
  1726. for(i=0;i<n;i++)
  1727. g->scale_factors[j++] = 0;
  1728. }
  1729. }
  1730. /* XXX: should compute exact size */
  1731. for(;j<40;j++)
  1732. g->scale_factors[j] = 0;
  1733. }
  1734. exponents_from_scale_factors(s, g, exponents);
  1735. /* read Huffman coded residue */
  1736. huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
  1737. } /* ch */
  1738. if (s->nb_channels == 2)
  1739. compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
  1740. for(ch=0;ch<s->nb_channels;ch++) {
  1741. g = &s->granules[ch][gr];
  1742. reorder_block(s, g);
  1743. compute_antialias(s, g);
  1744. compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
  1745. }
  1746. } /* gr */
  1747. if(get_bits_count(&s->gb)<0)
  1748. skip_bits_long(&s->gb, -get_bits_count(&s->gb));
  1749. return nb_granules * 18;
  1750. }
  1751. static int mp_decode_frame(MPADecodeContext *s,
  1752. OUT_INT *samples, const uint8_t *buf, int buf_size)
  1753. {
  1754. int i, nb_frames, ch;
  1755. OUT_INT *samples_ptr;
  1756. init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE)*8);
  1757. /* skip error protection field */
  1758. if (s->error_protection)
  1759. skip_bits(&s->gb, 16);
  1760. dprintf(s->avctx, "frame %d:\n", s->frame_count);
  1761. switch(s->layer) {
  1762. case 1:
  1763. s->avctx->frame_size = 384;
  1764. nb_frames = mp_decode_layer1(s);
  1765. break;
  1766. case 2:
  1767. s->avctx->frame_size = 1152;
  1768. nb_frames = mp_decode_layer2(s);
  1769. break;
  1770. case 3:
  1771. s->avctx->frame_size = s->lsf ? 576 : 1152;
  1772. default:
  1773. nb_frames = mp_decode_layer3(s);
  1774. s->last_buf_size=0;
  1775. if(s->in_gb.buffer){
  1776. align_get_bits(&s->gb);
  1777. i= get_bits_left(&s->gb)>>3;
  1778. if(i >= 0 && i <= BACKSTEP_SIZE){
  1779. memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);
  1780. s->last_buf_size=i;
  1781. }else
  1782. av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
  1783. s->gb= s->in_gb;
  1784. s->in_gb.buffer= NULL;
  1785. }
  1786. align_get_bits(&s->gb);
  1787. assert((get_bits_count(&s->gb) & 7) == 0);
  1788. i= get_bits_left(&s->gb)>>3;
  1789. if(i<0 || i > BACKSTEP_SIZE || nb_frames<0){
  1790. if(i<0)
  1791. av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
  1792. i= FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
  1793. }
  1794. assert(i <= buf_size - HEADER_SIZE && i>= 0);
  1795. memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
  1796. s->last_buf_size += i;
  1797. break;
  1798. }
  1799. /* apply the synthesis filter */
  1800. for(ch=0;ch<s->nb_channels;ch++) {
  1801. samples_ptr = samples + ch;
  1802. for(i=0;i<nb_frames;i++) {
  1803. RENAME(ff_mpa_synth_filter)(
  1804. #if CONFIG_FLOAT
  1805. s,
  1806. #endif
  1807. s->synth_buf[ch], &(s->synth_buf_offset[ch]),
  1808. RENAME(ff_mpa_synth_window), &s->dither_state,
  1809. samples_ptr, s->nb_channels,
  1810. s->sb_samples[ch][i]);
  1811. samples_ptr += 32 * s->nb_channels;
  1812. }
  1813. }
  1814. return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
  1815. }
  1816. static int decode_frame(AVCodecContext * avctx,
  1817. void *data, int *data_size,
  1818. AVPacket *avpkt)
  1819. {
  1820. const uint8_t *buf = avpkt->data;
  1821. int buf_size = avpkt->size;
  1822. MPADecodeContext *s = avctx->priv_data;
  1823. uint32_t header;
  1824. int out_size;
  1825. OUT_INT *out_samples = data;
  1826. if(buf_size < HEADER_SIZE)
  1827. return -1;
  1828. header = AV_RB32(buf);
  1829. if(ff_mpa_check_header(header) < 0){
  1830. if (buf_size == ID3v1_TAG_SIZE
  1831. && buf[0] == 'T' && buf[1] == 'A' && buf[2] == 'G') {
  1832. *data_size = 0;
  1833. return ID3v1_TAG_SIZE;
  1834. }
  1835. av_log(avctx, AV_LOG_ERROR, "Header missing\n");
  1836. return -1;
  1837. }
  1838. if (ff_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
  1839. /* free format: prepare to compute frame size */
  1840. s->frame_size = -1;
  1841. return -1;
  1842. }
  1843. /* update codec info */
  1844. avctx->channels = s->nb_channels;
  1845. if (!avctx->bit_rate)
  1846. avctx->bit_rate = s->bit_rate;
  1847. avctx->sub_id = s->layer;
  1848. if(*data_size < 1152*avctx->channels*sizeof(OUT_INT))
  1849. return -1;
  1850. *data_size = 0;
  1851. if(s->frame_size<=0 || s->frame_size > buf_size){
  1852. av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
  1853. return -1;
  1854. }else if(s->frame_size < buf_size){
  1855. av_log(avctx, AV_LOG_ERROR, "incorrect frame size\n");
  1856. buf_size= s->frame_size;
  1857. }
  1858. out_size = mp_decode_frame(s, out_samples, buf, buf_size);
  1859. if(out_size>=0){
  1860. *data_size = out_size;
  1861. avctx->sample_rate = s->sample_rate;
  1862. //FIXME maybe move the other codec info stuff from above here too
  1863. }else
  1864. av_log(avctx, AV_LOG_DEBUG, "Error while decoding MPEG audio frame.\n"); //FIXME return -1 / but also return the number of bytes consumed
  1865. s->frame_size = 0;
  1866. return buf_size;
  1867. }
  1868. static void flush(AVCodecContext *avctx){
  1869. MPADecodeContext *s = avctx->priv_data;
  1870. memset(s->synth_buf, 0, sizeof(s->synth_buf));
  1871. s->last_buf_size= 0;
  1872. }
  1873. #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
  1874. static int decode_frame_adu(AVCodecContext * avctx,
  1875. void *data, int *data_size,
  1876. AVPacket *avpkt)
  1877. {
  1878. const uint8_t *buf = avpkt->data;
  1879. int buf_size = avpkt->size;
  1880. MPADecodeContext *s = avctx->priv_data;
  1881. uint32_t header;
  1882. int len, out_size;
  1883. OUT_INT *out_samples = data;
  1884. len = buf_size;
  1885. // Discard too short frames
  1886. if (buf_size < HEADER_SIZE) {
  1887. *data_size = 0;
  1888. return buf_size;
  1889. }
  1890. if (len > MPA_MAX_CODED_FRAME_SIZE)
  1891. len = MPA_MAX_CODED_FRAME_SIZE;
  1892. // Get header and restore sync word
  1893. header = AV_RB32(buf) | 0xffe00000;
  1894. if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame
  1895. *data_size = 0;
  1896. return buf_size;
  1897. }
  1898. ff_mpegaudio_decode_header((MPADecodeHeader *)s, header);
  1899. /* update codec info */
  1900. avctx->sample_rate = s->sample_rate;
  1901. avctx->channels = s->nb_channels;
  1902. if (!avctx->bit_rate)
  1903. avctx->bit_rate = s->bit_rate;
  1904. avctx->sub_id = s->layer;
  1905. s->frame_size = len;
  1906. if (avctx->parse_only) {
  1907. out_size = buf_size;
  1908. } else {
  1909. out_size = mp_decode_frame(s, out_samples, buf, buf_size);
  1910. }
  1911. *data_size = out_size;
  1912. return buf_size;
  1913. }
  1914. #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
  1915. #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
  1916. /**
  1917. * Context for MP3On4 decoder
  1918. */
  1919. typedef struct MP3On4DecodeContext {
  1920. int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
  1921. int syncword; ///< syncword patch
  1922. const uint8_t *coff; ///< channels offsets in output buffer
  1923. MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
  1924. } MP3On4DecodeContext;
  1925. #include "mpeg4audio.h"
  1926. /* Next 3 arrays are indexed by channel config number (passed via codecdata) */
  1927. static const uint8_t mp3Frames[8] = {0,1,1,2,3,3,4,5}; /* number of mp3 decoder instances */
  1928. /* offsets into output buffer, assume output order is FL FR BL BR C LFE */
  1929. static const uint8_t chan_offset[8][5] = {
  1930. {0},
  1931. {0}, // C
  1932. {0}, // FLR
  1933. {2,0}, // C FLR
  1934. {2,0,3}, // C FLR BS
  1935. {4,0,2}, // C FLR BLRS
  1936. {4,0,2,5}, // C FLR BLRS LFE
  1937. {4,0,2,6,5}, // C FLR BLRS BLR LFE
  1938. };
  1939. static int decode_init_mp3on4(AVCodecContext * avctx)
  1940. {
  1941. MP3On4DecodeContext *s = avctx->priv_data;
  1942. MPEG4AudioConfig cfg;
  1943. int i;
  1944. if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) {
  1945. av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
  1946. return -1;
  1947. }
  1948. ff_mpeg4audio_get_config(&cfg, avctx->extradata, avctx->extradata_size);
  1949. if (!cfg.chan_config || cfg.chan_config > 7) {
  1950. av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
  1951. return -1;
  1952. }
  1953. s->frames = mp3Frames[cfg.chan_config];
  1954. s->coff = chan_offset[cfg.chan_config];
  1955. avctx->channels = ff_mpeg4audio_channels[cfg.chan_config];
  1956. if (cfg.sample_rate < 16000)
  1957. s->syncword = 0xffe00000;
  1958. else
  1959. s->syncword = 0xfff00000;
  1960. /* Init the first mp3 decoder in standard way, so that all tables get builded
  1961. * We replace avctx->priv_data with the context of the first decoder so that
  1962. * decode_init() does not have to be changed.
  1963. * Other decoders will be initialized here copying data from the first context
  1964. */
  1965. // Allocate zeroed memory for the first decoder context
  1966. s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
  1967. // Put decoder context in place to make init_decode() happy
  1968. avctx->priv_data = s->mp3decctx[0];
  1969. decode_init(avctx);
  1970. // Restore mp3on4 context pointer
  1971. avctx->priv_data = s;
  1972. s->mp3decctx[0]->adu_mode = 1; // Set adu mode
  1973. /* Create a separate codec/context for each frame (first is already ok).
  1974. * Each frame is 1 or 2 channels - up to 5 frames allowed
  1975. */
  1976. for (i = 1; i < s->frames; i++) {
  1977. s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
  1978. s->mp3decctx[i]->adu_mode = 1;
  1979. s->mp3decctx[i]->avctx = avctx;
  1980. }
  1981. return 0;
  1982. }
  1983. static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
  1984. {
  1985. MP3On4DecodeContext *s = avctx->priv_data;
  1986. int i;
  1987. for (i = 0; i < s->frames; i++)
  1988. if (s->mp3decctx[i])
  1989. av_free(s->mp3decctx[i]);
  1990. return 0;
  1991. }
  1992. static int decode_frame_mp3on4(AVCodecContext * avctx,
  1993. void *data, int *data_size,
  1994. AVPacket *avpkt)
  1995. {
  1996. const uint8_t *buf = avpkt->data;
  1997. int buf_size = avpkt->size;
  1998. MP3On4DecodeContext *s = avctx->priv_data;
  1999. MPADecodeContext *m;
  2000. int fsize, len = buf_size, out_size = 0;
  2001. uint32_t header;
  2002. OUT_INT *out_samples = data;
  2003. OUT_INT decoded_buf[MPA_FRAME_SIZE * MPA_MAX_CHANNELS];
  2004. OUT_INT *outptr, *bp;
  2005. int fr, j, n;
  2006. if(*data_size < MPA_FRAME_SIZE * MPA_MAX_CHANNELS * s->frames * sizeof(OUT_INT))
  2007. return -1;
  2008. *data_size = 0;
  2009. // Discard too short frames
  2010. if (buf_size < HEADER_SIZE)
  2011. return -1;
  2012. // If only one decoder interleave is not needed
  2013. outptr = s->frames == 1 ? out_samples : decoded_buf;
  2014. avctx->bit_rate = 0;
  2015. for (fr = 0; fr < s->frames; fr++) {
  2016. fsize = AV_RB16(buf) >> 4;
  2017. fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
  2018. m = s->mp3decctx[fr];
  2019. assert (m != NULL);
  2020. header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
  2021. if (ff_mpa_check_header(header) < 0) // Bad header, discard block
  2022. break;
  2023. ff_mpegaudio_decode_header((MPADecodeHeader *)m, header);
  2024. out_size += mp_decode_frame(m, outptr, buf, fsize);
  2025. buf += fsize;
  2026. len -= fsize;
  2027. if(s->frames > 1) {
  2028. n = m->avctx->frame_size*m->nb_channels;
  2029. /* interleave output data */
  2030. bp = out_samples + s->coff[fr];
  2031. if(m->nb_channels == 1) {
  2032. for(j = 0; j < n; j++) {
  2033. *bp = decoded_buf[j];
  2034. bp += avctx->channels;
  2035. }
  2036. } else {
  2037. for(j = 0; j < n; j++) {
  2038. bp[0] = decoded_buf[j++];
  2039. bp[1] = decoded_buf[j];
  2040. bp += avctx->channels;
  2041. }
  2042. }
  2043. }
  2044. avctx->bit_rate += m->bit_rate;
  2045. }
  2046. /* update codec info */
  2047. avctx->sample_rate = s->mp3decctx[0]->sample_rate;
  2048. *data_size = out_size;
  2049. return buf_size;
  2050. }
  2051. #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
  2052. #if !CONFIG_FLOAT
  2053. #if CONFIG_MP1_DECODER
  2054. AVCodec mp1_decoder =
  2055. {
  2056. "mp1",
  2057. AVMEDIA_TYPE_AUDIO,
  2058. CODEC_ID_MP1,
  2059. sizeof(MPADecodeContext),
  2060. decode_init,
  2061. NULL,
  2062. NULL,
  2063. decode_frame,
  2064. CODEC_CAP_PARSE_ONLY,
  2065. .flush= flush,
  2066. .long_name= NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
  2067. };
  2068. #endif
  2069. #if CONFIG_MP2_DECODER
  2070. AVCodec mp2_decoder =
  2071. {
  2072. "mp2",
  2073. AVMEDIA_TYPE_AUDIO,
  2074. CODEC_ID_MP2,
  2075. sizeof(MPADecodeContext),
  2076. decode_init,
  2077. NULL,
  2078. NULL,
  2079. decode_frame,
  2080. CODEC_CAP_PARSE_ONLY,
  2081. .flush= flush,
  2082. .long_name= NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
  2083. };
  2084. #endif
  2085. #if CONFIG_MP3_DECODER
  2086. AVCodec mp3_decoder =
  2087. {
  2088. "mp3",
  2089. AVMEDIA_TYPE_AUDIO,
  2090. CODEC_ID_MP3,
  2091. sizeof(MPADecodeContext),
  2092. decode_init,
  2093. NULL,
  2094. NULL,
  2095. decode_frame,
  2096. CODEC_CAP_PARSE_ONLY,
  2097. .flush= flush,
  2098. .long_name= NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
  2099. };
  2100. #endif
  2101. #if CONFIG_MP3ADU_DECODER
  2102. AVCodec mp3adu_decoder =
  2103. {
  2104. "mp3adu",
  2105. AVMEDIA_TYPE_AUDIO,
  2106. CODEC_ID_MP3ADU,
  2107. sizeof(MPADecodeContext),
  2108. decode_init,
  2109. NULL,
  2110. NULL,
  2111. decode_frame_adu,
  2112. CODEC_CAP_PARSE_ONLY,
  2113. .flush= flush,
  2114. .long_name= NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
  2115. };
  2116. #endif
  2117. #if CONFIG_MP3ON4_DECODER
  2118. AVCodec mp3on4_decoder =
  2119. {
  2120. "mp3on4",
  2121. AVMEDIA_TYPE_AUDIO,
  2122. CODEC_ID_MP3ON4,
  2123. sizeof(MP3On4DecodeContext),
  2124. decode_init_mp3on4,
  2125. NULL,
  2126. decode_close_mp3on4,
  2127. decode_frame_mp3on4,
  2128. .flush= flush,
  2129. .long_name= NULL_IF_CONFIG_SMALL("MP3onMP4"),
  2130. };
  2131. #endif
  2132. #endif