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