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  1. /*
  2. * MPEG Audio decoder
  3. * Copyright (c) 2001, 2002 Fabrice Bellard
  4. *
  5. * This file is part of FFmpeg.
  6. *
  7. * FFmpeg is free software; you can redistribute it and/or
  8. * modify it under the terms of the GNU Lesser General Public
  9. * License as published by the Free Software Foundation; either
  10. * version 2.1 of the License, or (at your option) any later version.
  11. *
  12. * FFmpeg is distributed in the hope that it will be useful,
  13. * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14. * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
  15. * Lesser General Public License for more details.
  16. *
  17. * You should have received a copy of the GNU Lesser General Public
  18. * License along with FFmpeg; if not, write to the Free Software
  19. * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20. */
  21. /**
  22. * @file
  23. * MPEG Audio decoder.
  24. */
  25. #include "libavutil/audioconvert.h"
  26. #include "avcodec.h"
  27. #include "get_bits.h"
  28. #include "mathops.h"
  29. #include "mpegaudiodsp.h"
  30. /*
  31. * TODO:
  32. * - test lsf / mpeg25 extensively.
  33. */
  34. #include "mpegaudio.h"
  35. #include "mpegaudiodecheader.h"
  36. #define BACKSTEP_SIZE 512
  37. #define EXTRABYTES 24
  38. /* layer 3 "granule" */
  39. typedef struct GranuleDef {
  40. uint8_t scfsi;
  41. int part2_3_length;
  42. int big_values;
  43. int global_gain;
  44. int scalefac_compress;
  45. uint8_t block_type;
  46. uint8_t switch_point;
  47. int table_select[3];
  48. int subblock_gain[3];
  49. uint8_t scalefac_scale;
  50. uint8_t count1table_select;
  51. int region_size[3]; /* number of huffman codes in each region */
  52. int preflag;
  53. int short_start, long_end; /* long/short band indexes */
  54. uint8_t scale_factors[40];
  55. INTFLOAT sb_hybrid[SBLIMIT * 18]; /* 576 samples */
  56. } GranuleDef;
  57. typedef struct MPADecodeContext {
  58. MPA_DECODE_HEADER
  59. uint8_t last_buf[2*BACKSTEP_SIZE + EXTRABYTES];
  60. int last_buf_size;
  61. /* next header (used in free format parsing) */
  62. uint32_t free_format_next_header;
  63. GetBitContext gb;
  64. GetBitContext in_gb;
  65. DECLARE_ALIGNED(32, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512 * 2];
  66. int synth_buf_offset[MPA_MAX_CHANNELS];
  67. DECLARE_ALIGNED(32, INTFLOAT, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT];
  68. INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
  69. GranuleDef granules[2][2]; /* Used in Layer 3 */
  70. #ifdef DEBUG
  71. int frame_count;
  72. #endif
  73. int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
  74. int dither_state;
  75. int error_recognition;
  76. AVCodecContext* avctx;
  77. MPADSPContext mpadsp;
  78. } MPADecodeContext;
  79. #if CONFIG_FLOAT
  80. # define SHR(a,b) ((a)*(1.0f/(1<<(b))))
  81. # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
  82. # define FIXR(x) ((float)(x))
  83. # define FIXHR(x) ((float)(x))
  84. # define MULH3(x, y, s) ((s)*(y)*(x))
  85. # define MULLx(x, y, s) ((y)*(x))
  86. # define RENAME(a) a ## _float
  87. # define OUT_FMT AV_SAMPLE_FMT_FLT
  88. #else
  89. # define SHR(a,b) ((a)>>(b))
  90. /* WARNING: only correct for posititive numbers */
  91. # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
  92. # define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
  93. # define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
  94. # define MULH3(x, y, s) MULH((s)*(x), y)
  95. # define MULLx(x, y, s) MULL(x,y,s)
  96. # define RENAME(a) a ## _fixed
  97. # define OUT_FMT AV_SAMPLE_FMT_S16
  98. #endif
  99. /****************/
  100. #define HEADER_SIZE 4
  101. #include "mpegaudiodata.h"
  102. #include "mpegaudiodectab.h"
  103. static void RENAME(compute_antialias)(MPADecodeContext *s, GranuleDef *g);
  104. /* vlc structure for decoding layer 3 huffman tables */
  105. static VLC huff_vlc[16];
  106. static VLC_TYPE huff_vlc_tables[
  107. 0+128+128+128+130+128+154+166+
  108. 142+204+190+170+542+460+662+414
  109. ][2];
  110. static const int huff_vlc_tables_sizes[16] = {
  111. 0, 128, 128, 128, 130, 128, 154, 166,
  112. 142, 204, 190, 170, 542, 460, 662, 414
  113. };
  114. static VLC huff_quad_vlc[2];
  115. static VLC_TYPE huff_quad_vlc_tables[128+16][2];
  116. static const int huff_quad_vlc_tables_sizes[2] = {
  117. 128, 16
  118. };
  119. /* computed from band_size_long */
  120. static uint16_t band_index_long[9][23];
  121. #include "mpegaudio_tablegen.h"
  122. /* intensity stereo coef table */
  123. static INTFLOAT is_table[2][16];
  124. static INTFLOAT is_table_lsf[2][2][16];
  125. static int32_t csa_table[8][4];
  126. static float csa_table_float[8][4];
  127. static INTFLOAT mdct_win[8][36];
  128. static int16_t division_tab3[1<<6 ];
  129. static int16_t division_tab5[1<<8 ];
  130. static int16_t division_tab9[1<<11];
  131. static int16_t * const division_tabs[4] = {
  132. division_tab3, division_tab5, NULL, division_tab9
  133. };
  134. /* lower 2 bits: modulo 3, higher bits: shift */
  135. static uint16_t scale_factor_modshift[64];
  136. /* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
  137. static int32_t scale_factor_mult[15][3];
  138. /* mult table for layer 2 group quantization */
  139. #define SCALE_GEN(v) \
  140. { FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
  141. static const int32_t scale_factor_mult2[3][3] = {
  142. SCALE_GEN(4.0 / 3.0), /* 3 steps */
  143. SCALE_GEN(4.0 / 5.0), /* 5 steps */
  144. SCALE_GEN(4.0 / 9.0), /* 9 steps */
  145. };
  146. /**
  147. * Convert region offsets to region sizes and truncate
  148. * size to big_values.
  149. */
  150. static void ff_region_offset2size(GranuleDef *g){
  151. int i, k, j=0;
  152. g->region_size[2] = (576 / 2);
  153. for(i=0;i<3;i++) {
  154. k = FFMIN(g->region_size[i], g->big_values);
  155. g->region_size[i] = k - j;
  156. j = k;
  157. }
  158. }
  159. static void ff_init_short_region(MPADecodeContext *s, GranuleDef *g){
  160. if (g->block_type == 2)
  161. g->region_size[0] = (36 / 2);
  162. else {
  163. if (s->sample_rate_index <= 2)
  164. g->region_size[0] = (36 / 2);
  165. else if (s->sample_rate_index != 8)
  166. g->region_size[0] = (54 / 2);
  167. else
  168. g->region_size[0] = (108 / 2);
  169. }
  170. g->region_size[1] = (576 / 2);
  171. }
  172. static void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2){
  173. int l;
  174. g->region_size[0] =
  175. band_index_long[s->sample_rate_index][ra1 + 1] >> 1;
  176. /* should not overflow */
  177. l = FFMIN(ra1 + ra2 + 2, 22);
  178. g->region_size[1] =
  179. band_index_long[s->sample_rate_index][l] >> 1;
  180. }
  181. static void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g){
  182. if (g->block_type == 2) {
  183. if (g->switch_point) {
  184. /* if switched mode, we handle the 36 first samples as
  185. long blocks. For 8000Hz, we handle the 48 first
  186. exponents as long blocks (XXX: check this!) */
  187. if (s->sample_rate_index <= 2)
  188. g->long_end = 8;
  189. else if (s->sample_rate_index != 8)
  190. g->long_end = 6;
  191. else
  192. g->long_end = 4; /* 8000 Hz */
  193. g->short_start = 2 + (s->sample_rate_index != 8);
  194. } else {
  195. g->long_end = 0;
  196. g->short_start = 0;
  197. }
  198. } else {
  199. g->short_start = 13;
  200. g->long_end = 22;
  201. }
  202. }
  203. /* layer 1 unscaling */
  204. /* n = number of bits of the mantissa minus 1 */
  205. static inline int l1_unscale(int n, int mant, int scale_factor)
  206. {
  207. int shift, mod;
  208. int64_t val;
  209. shift = scale_factor_modshift[scale_factor];
  210. mod = shift & 3;
  211. shift >>= 2;
  212. val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]);
  213. shift += n;
  214. /* NOTE: at this point, 1 <= shift >= 21 + 15 */
  215. return (int)((val + (1LL << (shift - 1))) >> shift);
  216. }
  217. static inline int l2_unscale_group(int steps, int mant, int scale_factor)
  218. {
  219. int shift, mod, val;
  220. shift = scale_factor_modshift[scale_factor];
  221. mod = shift & 3;
  222. shift >>= 2;
  223. val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
  224. /* NOTE: at this point, 0 <= shift <= 21 */
  225. if (shift > 0)
  226. val = (val + (1 << (shift - 1))) >> shift;
  227. return val;
  228. }
  229. /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
  230. static inline int l3_unscale(int value, int exponent)
  231. {
  232. unsigned int m;
  233. int e;
  234. e = table_4_3_exp [4*value + (exponent&3)];
  235. m = table_4_3_value[4*value + (exponent&3)];
  236. e -= (exponent >> 2);
  237. assert(e>=1);
  238. if (e > 31)
  239. return 0;
  240. m = (m + (1 << (e-1))) >> e;
  241. return m;
  242. }
  243. /* all integer n^(4/3) computation code */
  244. #define DEV_ORDER 13
  245. #define POW_FRAC_BITS 24
  246. #define POW_FRAC_ONE (1 << POW_FRAC_BITS)
  247. #define POW_FIX(a) ((int)((a) * POW_FRAC_ONE))
  248. #define POW_MULL(a,b) (((int64_t)(a) * (int64_t)(b)) >> POW_FRAC_BITS)
  249. static int dev_4_3_coefs[DEV_ORDER];
  250. static av_cold void int_pow_init(void)
  251. {
  252. int i, a;
  253. a = POW_FIX(1.0);
  254. for(i=0;i<DEV_ORDER;i++) {
  255. a = POW_MULL(a, POW_FIX(4.0 / 3.0) - i * POW_FIX(1.0)) / (i + 1);
  256. dev_4_3_coefs[i] = a;
  257. }
  258. }
  259. static av_cold int decode_init(AVCodecContext * avctx)
  260. {
  261. MPADecodeContext *s = avctx->priv_data;
  262. static int init=0;
  263. int i, j, k;
  264. s->avctx = avctx;
  265. ff_mpadsp_init(&s->mpadsp);
  266. avctx->sample_fmt= OUT_FMT;
  267. s->error_recognition= avctx->error_recognition;
  268. if (!init && !avctx->parse_only) {
  269. int offset;
  270. /* scale factors table for layer 1/2 */
  271. for(i=0;i<64;i++) {
  272. int shift, mod;
  273. /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
  274. shift = (i / 3);
  275. mod = i % 3;
  276. scale_factor_modshift[i] = mod | (shift << 2);
  277. }
  278. /* scale factor multiply for layer 1 */
  279. for(i=0;i<15;i++) {
  280. int n, norm;
  281. n = i + 2;
  282. norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
  283. scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
  284. scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
  285. scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
  286. av_dlog(avctx, "%d: norm=%x s=%x %x %x\n",
  287. i, norm,
  288. scale_factor_mult[i][0],
  289. scale_factor_mult[i][1],
  290. scale_factor_mult[i][2]);
  291. }
  292. RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));
  293. /* huffman decode tables */
  294. offset = 0;
  295. for(i=1;i<16;i++) {
  296. const HuffTable *h = &mpa_huff_tables[i];
  297. int xsize, x, y;
  298. uint8_t tmp_bits [512];
  299. uint16_t tmp_codes[512];
  300. memset(tmp_bits , 0, sizeof(tmp_bits ));
  301. memset(tmp_codes, 0, sizeof(tmp_codes));
  302. xsize = h->xsize;
  303. j = 0;
  304. for(x=0;x<xsize;x++) {
  305. for(y=0;y<xsize;y++){
  306. tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ];
  307. tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];
  308. }
  309. }
  310. /* XXX: fail test */
  311. huff_vlc[i].table = huff_vlc_tables+offset;
  312. huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
  313. init_vlc(&huff_vlc[i], 7, 512,
  314. tmp_bits, 1, 1, tmp_codes, 2, 2,
  315. INIT_VLC_USE_NEW_STATIC);
  316. offset += huff_vlc_tables_sizes[i];
  317. }
  318. assert(offset == FF_ARRAY_ELEMS(huff_vlc_tables));
  319. offset = 0;
  320. for(i=0;i<2;i++) {
  321. huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
  322. huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
  323. init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
  324. mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
  325. INIT_VLC_USE_NEW_STATIC);
  326. offset += huff_quad_vlc_tables_sizes[i];
  327. }
  328. assert(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));
  329. for(i=0;i<9;i++) {
  330. k = 0;
  331. for(j=0;j<22;j++) {
  332. band_index_long[i][j] = k;
  333. k += band_size_long[i][j];
  334. }
  335. band_index_long[i][22] = k;
  336. }
  337. /* compute n ^ (4/3) and store it in mantissa/exp format */
  338. int_pow_init();
  339. mpegaudio_tableinit();
  340. for (i = 0; i < 4; i++)
  341. if (ff_mpa_quant_bits[i] < 0)
  342. for (j = 0; j < (1<<(-ff_mpa_quant_bits[i]+1)); j++) {
  343. int val1, val2, val3, steps;
  344. int val = j;
  345. steps = ff_mpa_quant_steps[i];
  346. val1 = val % steps;
  347. val /= steps;
  348. val2 = val % steps;
  349. val3 = val / steps;
  350. division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);
  351. }
  352. for(i=0;i<7;i++) {
  353. float f;
  354. INTFLOAT v;
  355. if (i != 6) {
  356. f = tan((double)i * M_PI / 12.0);
  357. v = FIXR(f / (1.0 + f));
  358. } else {
  359. v = FIXR(1.0);
  360. }
  361. is_table[0][i] = v;
  362. is_table[1][6 - i] = v;
  363. }
  364. /* invalid values */
  365. for(i=7;i<16;i++)
  366. is_table[0][i] = is_table[1][i] = 0.0;
  367. for(i=0;i<16;i++) {
  368. double f;
  369. int e, k;
  370. for(j=0;j<2;j++) {
  371. e = -(j + 1) * ((i + 1) >> 1);
  372. f = pow(2.0, e / 4.0);
  373. k = i & 1;
  374. is_table_lsf[j][k ^ 1][i] = FIXR(f);
  375. is_table_lsf[j][k][i] = FIXR(1.0);
  376. av_dlog(avctx, "is_table_lsf %d %d: %x %x\n",
  377. i, j, is_table_lsf[j][0][i], is_table_lsf[j][1][i]);
  378. }
  379. }
  380. for(i=0;i<8;i++) {
  381. float ci, cs, ca;
  382. ci = ci_table[i];
  383. cs = 1.0 / sqrt(1.0 + ci * ci);
  384. ca = cs * ci;
  385. csa_table[i][0] = FIXHR(cs/4);
  386. csa_table[i][1] = FIXHR(ca/4);
  387. csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);
  388. csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);
  389. csa_table_float[i][0] = cs;
  390. csa_table_float[i][1] = ca;
  391. csa_table_float[i][2] = ca + cs;
  392. csa_table_float[i][3] = ca - cs;
  393. }
  394. /* compute mdct windows */
  395. for(i=0;i<36;i++) {
  396. for(j=0; j<4; j++){
  397. double d;
  398. if(j==2 && i%3 != 1)
  399. continue;
  400. d= sin(M_PI * (i + 0.5) / 36.0);
  401. if(j==1){
  402. if (i>=30) d= 0;
  403. else if(i>=24) d= sin(M_PI * (i - 18 + 0.5) / 12.0);
  404. else if(i>=18) d= 1;
  405. }else if(j==3){
  406. if (i< 6) d= 0;
  407. else if(i< 12) d= sin(M_PI * (i - 6 + 0.5) / 12.0);
  408. else if(i< 18) d= 1;
  409. }
  410. //merge last stage of imdct into the window coefficients
  411. d*= 0.5 / cos(M_PI*(2*i + 19)/72);
  412. if(j==2)
  413. mdct_win[j][i/3] = FIXHR((d / (1<<5)));
  414. else
  415. mdct_win[j][i ] = FIXHR((d / (1<<5)));
  416. }
  417. }
  418. /* NOTE: we do frequency inversion adter the MDCT by changing
  419. the sign of the right window coefs */
  420. for(j=0;j<4;j++) {
  421. for(i=0;i<36;i+=2) {
  422. mdct_win[j + 4][i] = mdct_win[j][i];
  423. mdct_win[j + 4][i + 1] = -mdct_win[j][i + 1];
  424. }
  425. }
  426. init = 1;
  427. }
  428. if (avctx->codec_id == CODEC_ID_MP3ADU)
  429. s->adu_mode = 1;
  430. return 0;
  431. }
  432. #define C3 FIXHR(0.86602540378443864676/2)
  433. /* 0.5 / cos(pi*(2*i+1)/36) */
  434. static const INTFLOAT icos36[9] = {
  435. FIXR(0.50190991877167369479),
  436. FIXR(0.51763809020504152469), //0
  437. FIXR(0.55168895948124587824),
  438. FIXR(0.61038729438072803416),
  439. FIXR(0.70710678118654752439), //1
  440. FIXR(0.87172339781054900991),
  441. FIXR(1.18310079157624925896),
  442. FIXR(1.93185165257813657349), //2
  443. FIXR(5.73685662283492756461),
  444. };
  445. /* 0.5 / cos(pi*(2*i+1)/36) */
  446. static const INTFLOAT icos36h[9] = {
  447. FIXHR(0.50190991877167369479/2),
  448. FIXHR(0.51763809020504152469/2), //0
  449. FIXHR(0.55168895948124587824/2),
  450. FIXHR(0.61038729438072803416/2),
  451. FIXHR(0.70710678118654752439/2), //1
  452. FIXHR(0.87172339781054900991/2),
  453. FIXHR(1.18310079157624925896/4),
  454. FIXHR(1.93185165257813657349/4), //2
  455. // FIXHR(5.73685662283492756461),
  456. };
  457. /* 12 points IMDCT. We compute it "by hand" by factorizing obvious
  458. cases. */
  459. static void imdct12(INTFLOAT *out, INTFLOAT *in)
  460. {
  461. INTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
  462. in0= in[0*3];
  463. in1= in[1*3] + in[0*3];
  464. in2= in[2*3] + in[1*3];
  465. in3= in[3*3] + in[2*3];
  466. in4= in[4*3] + in[3*3];
  467. in5= in[5*3] + in[4*3];
  468. in5 += in3;
  469. in3 += in1;
  470. in2= MULH3(in2, C3, 2);
  471. in3= MULH3(in3, C3, 4);
  472. t1 = in0 - in4;
  473. t2 = MULH3(in1 - in5, icos36h[4], 2);
  474. out[ 7]=
  475. out[10]= t1 + t2;
  476. out[ 1]=
  477. out[ 4]= t1 - t2;
  478. in0 += SHR(in4, 1);
  479. in4 = in0 + in2;
  480. in5 += 2*in1;
  481. in1 = MULH3(in5 + in3, icos36h[1], 1);
  482. out[ 8]=
  483. out[ 9]= in4 + in1;
  484. out[ 2]=
  485. out[ 3]= in4 - in1;
  486. in0 -= in2;
  487. in5 = MULH3(in5 - in3, icos36h[7], 2);
  488. out[ 0]=
  489. out[ 5]= in0 - in5;
  490. out[ 6]=
  491. out[11]= in0 + in5;
  492. }
  493. /* cos(pi*i/18) */
  494. #define C1 FIXHR(0.98480775301220805936/2)
  495. #define C2 FIXHR(0.93969262078590838405/2)
  496. #define C3 FIXHR(0.86602540378443864676/2)
  497. #define C4 FIXHR(0.76604444311897803520/2)
  498. #define C5 FIXHR(0.64278760968653932632/2)
  499. #define C6 FIXHR(0.5/2)
  500. #define C7 FIXHR(0.34202014332566873304/2)
  501. #define C8 FIXHR(0.17364817766693034885/2)
  502. /* using Lee like decomposition followed by hand coded 9 points DCT */
  503. static void imdct36(INTFLOAT *out, INTFLOAT *buf, INTFLOAT *in, INTFLOAT *win)
  504. {
  505. int i, j;
  506. INTFLOAT t0, t1, t2, t3, s0, s1, s2, s3;
  507. INTFLOAT tmp[18], *tmp1, *in1;
  508. for(i=17;i>=1;i--)
  509. in[i] += in[i-1];
  510. for(i=17;i>=3;i-=2)
  511. in[i] += in[i-2];
  512. for(j=0;j<2;j++) {
  513. tmp1 = tmp + j;
  514. in1 = in + j;
  515. t2 = in1[2*4] + in1[2*8] - in1[2*2];
  516. t3 = in1[2*0] + SHR(in1[2*6],1);
  517. t1 = in1[2*0] - in1[2*6];
  518. tmp1[ 6] = t1 - SHR(t2,1);
  519. tmp1[16] = t1 + t2;
  520. t0 = MULH3(in1[2*2] + in1[2*4] , C2, 2);
  521. t1 = MULH3(in1[2*4] - in1[2*8] , -2*C8, 1);
  522. t2 = MULH3(in1[2*2] + in1[2*8] , -C4, 2);
  523. tmp1[10] = t3 - t0 - t2;
  524. tmp1[ 2] = t3 + t0 + t1;
  525. tmp1[14] = t3 + t2 - t1;
  526. tmp1[ 4] = MULH3(in1[2*5] + in1[2*7] - in1[2*1], -C3, 2);
  527. t2 = MULH3(in1[2*1] + in1[2*5], C1, 2);
  528. t3 = MULH3(in1[2*5] - in1[2*7], -2*C7, 1);
  529. t0 = MULH3(in1[2*3], C3, 2);
  530. t1 = MULH3(in1[2*1] + in1[2*7], -C5, 2);
  531. tmp1[ 0] = t2 + t3 + t0;
  532. tmp1[12] = t2 + t1 - t0;
  533. tmp1[ 8] = t3 - t1 - t0;
  534. }
  535. i = 0;
  536. for(j=0;j<4;j++) {
  537. t0 = tmp[i];
  538. t1 = tmp[i + 2];
  539. s0 = t1 + t0;
  540. s2 = t1 - t0;
  541. t2 = tmp[i + 1];
  542. t3 = tmp[i + 3];
  543. s1 = MULH3(t3 + t2, icos36h[j], 2);
  544. s3 = MULLx(t3 - t2, icos36[8 - j], FRAC_BITS);
  545. t0 = s0 + s1;
  546. t1 = s0 - s1;
  547. out[(9 + j)*SBLIMIT] = MULH3(t1, win[9 + j], 1) + buf[9 + j];
  548. out[(8 - j)*SBLIMIT] = MULH3(t1, win[8 - j], 1) + buf[8 - j];
  549. buf[9 + j] = MULH3(t0, win[18 + 9 + j], 1);
  550. buf[8 - j] = MULH3(t0, win[18 + 8 - j], 1);
  551. t0 = s2 + s3;
  552. t1 = s2 - s3;
  553. out[(9 + 8 - j)*SBLIMIT] = MULH3(t1, win[9 + 8 - j], 1) + buf[9 + 8 - j];
  554. out[( j)*SBLIMIT] = MULH3(t1, win[ j], 1) + buf[ j];
  555. buf[9 + 8 - j] = MULH3(t0, win[18 + 9 + 8 - j], 1);
  556. buf[ + j] = MULH3(t0, win[18 + j], 1);
  557. i += 4;
  558. }
  559. s0 = tmp[16];
  560. s1 = MULH3(tmp[17], icos36h[4], 2);
  561. t0 = s0 + s1;
  562. t1 = s0 - s1;
  563. out[(9 + 4)*SBLIMIT] = MULH3(t1, win[9 + 4], 1) + buf[9 + 4];
  564. out[(8 - 4)*SBLIMIT] = MULH3(t1, win[8 - 4], 1) + buf[8 - 4];
  565. buf[9 + 4] = MULH3(t0, win[18 + 9 + 4], 1);
  566. buf[8 - 4] = MULH3(t0, win[18 + 8 - 4], 1);
  567. }
  568. /* return the number of decoded frames */
  569. static int mp_decode_layer1(MPADecodeContext *s)
  570. {
  571. int bound, i, v, n, ch, j, mant;
  572. uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
  573. uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
  574. if (s->mode == MPA_JSTEREO)
  575. bound = (s->mode_ext + 1) * 4;
  576. else
  577. bound = SBLIMIT;
  578. /* allocation bits */
  579. for(i=0;i<bound;i++) {
  580. for(ch=0;ch<s->nb_channels;ch++) {
  581. allocation[ch][i] = get_bits(&s->gb, 4);
  582. }
  583. }
  584. for(i=bound;i<SBLIMIT;i++) {
  585. allocation[0][i] = get_bits(&s->gb, 4);
  586. }
  587. /* scale factors */
  588. for(i=0;i<bound;i++) {
  589. for(ch=0;ch<s->nb_channels;ch++) {
  590. if (allocation[ch][i])
  591. scale_factors[ch][i] = get_bits(&s->gb, 6);
  592. }
  593. }
  594. for(i=bound;i<SBLIMIT;i++) {
  595. if (allocation[0][i]) {
  596. scale_factors[0][i] = get_bits(&s->gb, 6);
  597. scale_factors[1][i] = get_bits(&s->gb, 6);
  598. }
  599. }
  600. /* compute samples */
  601. for(j=0;j<12;j++) {
  602. for(i=0;i<bound;i++) {
  603. for(ch=0;ch<s->nb_channels;ch++) {
  604. n = allocation[ch][i];
  605. if (n) {
  606. mant = get_bits(&s->gb, n + 1);
  607. v = l1_unscale(n, mant, scale_factors[ch][i]);
  608. } else {
  609. v = 0;
  610. }
  611. s->sb_samples[ch][j][i] = v;
  612. }
  613. }
  614. for(i=bound;i<SBLIMIT;i++) {
  615. n = allocation[0][i];
  616. if (n) {
  617. mant = get_bits(&s->gb, n + 1);
  618. v = l1_unscale(n, mant, scale_factors[0][i]);
  619. s->sb_samples[0][j][i] = v;
  620. v = l1_unscale(n, mant, scale_factors[1][i]);
  621. s->sb_samples[1][j][i] = v;
  622. } else {
  623. s->sb_samples[0][j][i] = 0;
  624. s->sb_samples[1][j][i] = 0;
  625. }
  626. }
  627. }
  628. return 12;
  629. }
  630. static int mp_decode_layer2(MPADecodeContext *s)
  631. {
  632. int sblimit; /* number of used subbands */
  633. const unsigned char *alloc_table;
  634. int table, bit_alloc_bits, i, j, ch, bound, v;
  635. unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
  636. unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
  637. unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
  638. int scale, qindex, bits, steps, k, l, m, b;
  639. /* select decoding table */
  640. table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
  641. s->sample_rate, s->lsf);
  642. sblimit = ff_mpa_sblimit_table[table];
  643. alloc_table = ff_mpa_alloc_tables[table];
  644. if (s->mode == MPA_JSTEREO)
  645. bound = (s->mode_ext + 1) * 4;
  646. else
  647. bound = sblimit;
  648. av_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
  649. /* sanity check */
  650. if( bound > sblimit ) bound = sblimit;
  651. /* parse bit allocation */
  652. j = 0;
  653. for(i=0;i<bound;i++) {
  654. bit_alloc_bits = alloc_table[j];
  655. for(ch=0;ch<s->nb_channels;ch++) {
  656. bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
  657. }
  658. j += 1 << bit_alloc_bits;
  659. }
  660. for(i=bound;i<sblimit;i++) {
  661. bit_alloc_bits = alloc_table[j];
  662. v = get_bits(&s->gb, bit_alloc_bits);
  663. bit_alloc[0][i] = v;
  664. bit_alloc[1][i] = v;
  665. j += 1 << bit_alloc_bits;
  666. }
  667. /* scale codes */
  668. for(i=0;i<sblimit;i++) {
  669. for(ch=0;ch<s->nb_channels;ch++) {
  670. if (bit_alloc[ch][i])
  671. scale_code[ch][i] = get_bits(&s->gb, 2);
  672. }
  673. }
  674. /* scale factors */
  675. for(i=0;i<sblimit;i++) {
  676. for(ch=0;ch<s->nb_channels;ch++) {
  677. if (bit_alloc[ch][i]) {
  678. sf = scale_factors[ch][i];
  679. switch(scale_code[ch][i]) {
  680. default:
  681. case 0:
  682. sf[0] = get_bits(&s->gb, 6);
  683. sf[1] = get_bits(&s->gb, 6);
  684. sf[2] = get_bits(&s->gb, 6);
  685. break;
  686. case 2:
  687. sf[0] = get_bits(&s->gb, 6);
  688. sf[1] = sf[0];
  689. sf[2] = sf[0];
  690. break;
  691. case 1:
  692. sf[0] = get_bits(&s->gb, 6);
  693. sf[2] = get_bits(&s->gb, 6);
  694. sf[1] = sf[0];
  695. break;
  696. case 3:
  697. sf[0] = get_bits(&s->gb, 6);
  698. sf[2] = get_bits(&s->gb, 6);
  699. sf[1] = sf[2];
  700. break;
  701. }
  702. }
  703. }
  704. }
  705. /* samples */
  706. for(k=0;k<3;k++) {
  707. for(l=0;l<12;l+=3) {
  708. j = 0;
  709. for(i=0;i<bound;i++) {
  710. bit_alloc_bits = alloc_table[j];
  711. for(ch=0;ch<s->nb_channels;ch++) {
  712. b = bit_alloc[ch][i];
  713. if (b) {
  714. scale = scale_factors[ch][i][k];
  715. qindex = alloc_table[j+b];
  716. bits = ff_mpa_quant_bits[qindex];
  717. if (bits < 0) {
  718. int v2;
  719. /* 3 values at the same time */
  720. v = get_bits(&s->gb, -bits);
  721. v2 = division_tabs[qindex][v];
  722. steps = ff_mpa_quant_steps[qindex];
  723. s->sb_samples[ch][k * 12 + l + 0][i] =
  724. l2_unscale_group(steps, v2 & 15, scale);
  725. s->sb_samples[ch][k * 12 + l + 1][i] =
  726. l2_unscale_group(steps, (v2 >> 4) & 15, scale);
  727. s->sb_samples[ch][k * 12 + l + 2][i] =
  728. l2_unscale_group(steps, v2 >> 8 , scale);
  729. } else {
  730. for(m=0;m<3;m++) {
  731. v = get_bits(&s->gb, bits);
  732. v = l1_unscale(bits - 1, v, scale);
  733. s->sb_samples[ch][k * 12 + l + m][i] = v;
  734. }
  735. }
  736. } else {
  737. s->sb_samples[ch][k * 12 + l + 0][i] = 0;
  738. s->sb_samples[ch][k * 12 + l + 1][i] = 0;
  739. s->sb_samples[ch][k * 12 + l + 2][i] = 0;
  740. }
  741. }
  742. /* next subband in alloc table */
  743. j += 1 << bit_alloc_bits;
  744. }
  745. /* XXX: find a way to avoid this duplication of code */
  746. for(i=bound;i<sblimit;i++) {
  747. bit_alloc_bits = alloc_table[j];
  748. b = bit_alloc[0][i];
  749. if (b) {
  750. int mant, scale0, scale1;
  751. scale0 = scale_factors[0][i][k];
  752. scale1 = scale_factors[1][i][k];
  753. qindex = alloc_table[j+b];
  754. bits = ff_mpa_quant_bits[qindex];
  755. if (bits < 0) {
  756. /* 3 values at the same time */
  757. v = get_bits(&s->gb, -bits);
  758. steps = ff_mpa_quant_steps[qindex];
  759. mant = v % steps;
  760. v = v / steps;
  761. s->sb_samples[0][k * 12 + l + 0][i] =
  762. l2_unscale_group(steps, mant, scale0);
  763. s->sb_samples[1][k * 12 + l + 0][i] =
  764. l2_unscale_group(steps, mant, scale1);
  765. mant = v % steps;
  766. v = v / steps;
  767. s->sb_samples[0][k * 12 + l + 1][i] =
  768. l2_unscale_group(steps, mant, scale0);
  769. s->sb_samples[1][k * 12 + l + 1][i] =
  770. l2_unscale_group(steps, mant, scale1);
  771. s->sb_samples[0][k * 12 + l + 2][i] =
  772. l2_unscale_group(steps, v, scale0);
  773. s->sb_samples[1][k * 12 + l + 2][i] =
  774. l2_unscale_group(steps, v, scale1);
  775. } else {
  776. for(m=0;m<3;m++) {
  777. mant = get_bits(&s->gb, bits);
  778. s->sb_samples[0][k * 12 + l + m][i] =
  779. l1_unscale(bits - 1, mant, scale0);
  780. s->sb_samples[1][k * 12 + l + m][i] =
  781. l1_unscale(bits - 1, mant, scale1);
  782. }
  783. }
  784. } else {
  785. s->sb_samples[0][k * 12 + l + 0][i] = 0;
  786. s->sb_samples[0][k * 12 + l + 1][i] = 0;
  787. s->sb_samples[0][k * 12 + l + 2][i] = 0;
  788. s->sb_samples[1][k * 12 + l + 0][i] = 0;
  789. s->sb_samples[1][k * 12 + l + 1][i] = 0;
  790. s->sb_samples[1][k * 12 + l + 2][i] = 0;
  791. }
  792. /* next subband in alloc table */
  793. j += 1 << bit_alloc_bits;
  794. }
  795. /* fill remaining samples to zero */
  796. for(i=sblimit;i<SBLIMIT;i++) {
  797. for(ch=0;ch<s->nb_channels;ch++) {
  798. s->sb_samples[ch][k * 12 + l + 0][i] = 0;
  799. s->sb_samples[ch][k * 12 + l + 1][i] = 0;
  800. s->sb_samples[ch][k * 12 + l + 2][i] = 0;
  801. }
  802. }
  803. }
  804. }
  805. return 3 * 12;
  806. }
  807. #define SPLIT(dst,sf,n)\
  808. if(n==3){\
  809. int m= (sf*171)>>9;\
  810. dst= sf - 3*m;\
  811. sf=m;\
  812. }else if(n==4){\
  813. dst= sf&3;\
  814. sf>>=2;\
  815. }else if(n==5){\
  816. int m= (sf*205)>>10;\
  817. dst= sf - 5*m;\
  818. sf=m;\
  819. }else if(n==6){\
  820. int m= (sf*171)>>10;\
  821. dst= sf - 6*m;\
  822. sf=m;\
  823. }else{\
  824. dst=0;\
  825. }
  826. static av_always_inline void lsf_sf_expand(int *slen,
  827. int sf, int n1, int n2, int n3)
  828. {
  829. SPLIT(slen[3], sf, n3)
  830. SPLIT(slen[2], sf, n2)
  831. SPLIT(slen[1], sf, n1)
  832. slen[0] = sf;
  833. }
  834. static void exponents_from_scale_factors(MPADecodeContext *s,
  835. GranuleDef *g,
  836. int16_t *exponents)
  837. {
  838. const uint8_t *bstab, *pretab;
  839. int len, i, j, k, l, v0, shift, gain, gains[3];
  840. int16_t *exp_ptr;
  841. exp_ptr = exponents;
  842. gain = g->global_gain - 210;
  843. shift = g->scalefac_scale + 1;
  844. bstab = band_size_long[s->sample_rate_index];
  845. pretab = mpa_pretab[g->preflag];
  846. for(i=0;i<g->long_end;i++) {
  847. v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
  848. len = bstab[i];
  849. for(j=len;j>0;j--)
  850. *exp_ptr++ = v0;
  851. }
  852. if (g->short_start < 13) {
  853. bstab = band_size_short[s->sample_rate_index];
  854. gains[0] = gain - (g->subblock_gain[0] << 3);
  855. gains[1] = gain - (g->subblock_gain[1] << 3);
  856. gains[2] = gain - (g->subblock_gain[2] << 3);
  857. k = g->long_end;
  858. for(i=g->short_start;i<13;i++) {
  859. len = bstab[i];
  860. for(l=0;l<3;l++) {
  861. v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
  862. for(j=len;j>0;j--)
  863. *exp_ptr++ = v0;
  864. }
  865. }
  866. }
  867. }
  868. /* handle n = 0 too */
  869. static inline int get_bitsz(GetBitContext *s, int n)
  870. {
  871. if (n == 0)
  872. return 0;
  873. else
  874. return get_bits(s, n);
  875. }
  876. static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos, int *end_pos2){
  877. if(s->in_gb.buffer && *pos >= s->gb.size_in_bits){
  878. s->gb= s->in_gb;
  879. s->in_gb.buffer=NULL;
  880. assert((get_bits_count(&s->gb) & 7) == 0);
  881. skip_bits_long(&s->gb, *pos - *end_pos);
  882. *end_pos2=
  883. *end_pos= *end_pos2 + get_bits_count(&s->gb) - *pos;
  884. *pos= get_bits_count(&s->gb);
  885. }
  886. }
  887. /* Following is a optimized code for
  888. INTFLOAT v = *src
  889. if(get_bits1(&s->gb))
  890. v = -v;
  891. *dst = v;
  892. */
  893. #if CONFIG_FLOAT
  894. #define READ_FLIP_SIGN(dst,src)\
  895. v = AV_RN32A(src) ^ (get_bits1(&s->gb)<<31);\
  896. AV_WN32A(dst, v);
  897. #else
  898. #define READ_FLIP_SIGN(dst,src)\
  899. v= -get_bits1(&s->gb);\
  900. *(dst) = (*(src) ^ v) - v;
  901. #endif
  902. static int huffman_decode(MPADecodeContext *s, GranuleDef *g,
  903. int16_t *exponents, int end_pos2)
  904. {
  905. int s_index;
  906. int i;
  907. int last_pos, bits_left;
  908. VLC *vlc;
  909. int end_pos= FFMIN(end_pos2, s->gb.size_in_bits);
  910. /* low frequencies (called big values) */
  911. s_index = 0;
  912. for(i=0;i<3;i++) {
  913. int j, k, l, linbits;
  914. j = g->region_size[i];
  915. if (j == 0)
  916. continue;
  917. /* select vlc table */
  918. k = g->table_select[i];
  919. l = mpa_huff_data[k][0];
  920. linbits = mpa_huff_data[k][1];
  921. vlc = &huff_vlc[l];
  922. if(!l){
  923. memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid)*2*j);
  924. s_index += 2*j;
  925. continue;
  926. }
  927. /* read huffcode and compute each couple */
  928. for(;j>0;j--) {
  929. int exponent, x, y;
  930. int v;
  931. int pos= get_bits_count(&s->gb);
  932. if (pos >= end_pos){
  933. // av_log(NULL, AV_LOG_ERROR, "pos: %d %d %d %d\n", pos, end_pos, end_pos2, s_index);
  934. switch_buffer(s, &pos, &end_pos, &end_pos2);
  935. // av_log(NULL, AV_LOG_ERROR, "new pos: %d %d\n", pos, end_pos);
  936. if(pos >= end_pos)
  937. break;
  938. }
  939. y = get_vlc2(&s->gb, vlc->table, 7, 3);
  940. if(!y){
  941. g->sb_hybrid[s_index ] =
  942. g->sb_hybrid[s_index+1] = 0;
  943. s_index += 2;
  944. continue;
  945. }
  946. exponent= exponents[s_index];
  947. av_dlog(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n",
  948. i, g->region_size[i] - j, x, y, exponent);
  949. if(y&16){
  950. x = y >> 5;
  951. y = y & 0x0f;
  952. if (x < 15){
  953. READ_FLIP_SIGN(g->sb_hybrid+s_index, RENAME(expval_table)[ exponent ]+x)
  954. }else{
  955. x += get_bitsz(&s->gb, linbits);
  956. v = l3_unscale(x, exponent);
  957. if (get_bits1(&s->gb))
  958. v = -v;
  959. g->sb_hybrid[s_index] = v;
  960. }
  961. if (y < 15){
  962. READ_FLIP_SIGN(g->sb_hybrid+s_index+1, RENAME(expval_table)[ exponent ]+y)
  963. }else{
  964. y += get_bitsz(&s->gb, linbits);
  965. v = l3_unscale(y, exponent);
  966. if (get_bits1(&s->gb))
  967. v = -v;
  968. g->sb_hybrid[s_index+1] = v;
  969. }
  970. }else{
  971. x = y >> 5;
  972. y = y & 0x0f;
  973. x += y;
  974. if (x < 15){
  975. READ_FLIP_SIGN(g->sb_hybrid+s_index+!!y, RENAME(expval_table)[ exponent ]+x)
  976. }else{
  977. x += get_bitsz(&s->gb, linbits);
  978. v = l3_unscale(x, exponent);
  979. if (get_bits1(&s->gb))
  980. v = -v;
  981. g->sb_hybrid[s_index+!!y] = v;
  982. }
  983. g->sb_hybrid[s_index+ !y] = 0;
  984. }
  985. s_index+=2;
  986. }
  987. }
  988. /* high frequencies */
  989. vlc = &huff_quad_vlc[g->count1table_select];
  990. last_pos=0;
  991. while (s_index <= 572) {
  992. int pos, code;
  993. pos = get_bits_count(&s->gb);
  994. if (pos >= end_pos) {
  995. if (pos > end_pos2 && last_pos){
  996. /* some encoders generate an incorrect size for this
  997. part. We must go back into the data */
  998. s_index -= 4;
  999. skip_bits_long(&s->gb, last_pos - pos);
  1000. av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
  1001. if(s->error_recognition >= FF_ER_COMPLIANT)
  1002. s_index=0;
  1003. break;
  1004. }
  1005. // av_log(NULL, AV_LOG_ERROR, "pos2: %d %d %d %d\n", pos, end_pos, end_pos2, s_index);
  1006. switch_buffer(s, &pos, &end_pos, &end_pos2);
  1007. // av_log(NULL, AV_LOG_ERROR, "new pos2: %d %d %d\n", pos, end_pos, s_index);
  1008. if(pos >= end_pos)
  1009. break;
  1010. }
  1011. last_pos= pos;
  1012. code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
  1013. av_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
  1014. g->sb_hybrid[s_index+0]=
  1015. g->sb_hybrid[s_index+1]=
  1016. g->sb_hybrid[s_index+2]=
  1017. g->sb_hybrid[s_index+3]= 0;
  1018. while(code){
  1019. static const int idxtab[16]={3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0};
  1020. int v;
  1021. int pos= s_index+idxtab[code];
  1022. code ^= 8>>idxtab[code];
  1023. READ_FLIP_SIGN(g->sb_hybrid+pos, RENAME(exp_table)+exponents[pos])
  1024. }
  1025. s_index+=4;
  1026. }
  1027. /* skip extension bits */
  1028. bits_left = end_pos2 - get_bits_count(&s->gb);
  1029. //av_log(NULL, AV_LOG_ERROR, "left:%d buf:%p\n", bits_left, s->in_gb.buffer);
  1030. if (bits_left < 0 && s->error_recognition >= FF_ER_COMPLIANT) {
  1031. av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
  1032. s_index=0;
  1033. }else if(bits_left > 0 && s->error_recognition >= FF_ER_AGGRESSIVE){
  1034. av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
  1035. s_index=0;
  1036. }
  1037. memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid)*(576 - s_index));
  1038. skip_bits_long(&s->gb, bits_left);
  1039. i= get_bits_count(&s->gb);
  1040. switch_buffer(s, &i, &end_pos, &end_pos2);
  1041. return 0;
  1042. }
  1043. /* Reorder short blocks from bitstream order to interleaved order. It
  1044. would be faster to do it in parsing, but the code would be far more
  1045. complicated */
  1046. static void reorder_block(MPADecodeContext *s, GranuleDef *g)
  1047. {
  1048. int i, j, len;
  1049. INTFLOAT *ptr, *dst, *ptr1;
  1050. INTFLOAT tmp[576];
  1051. if (g->block_type != 2)
  1052. return;
  1053. if (g->switch_point) {
  1054. if (s->sample_rate_index != 8) {
  1055. ptr = g->sb_hybrid + 36;
  1056. } else {
  1057. ptr = g->sb_hybrid + 48;
  1058. }
  1059. } else {
  1060. ptr = g->sb_hybrid;
  1061. }
  1062. for(i=g->short_start;i<13;i++) {
  1063. len = band_size_short[s->sample_rate_index][i];
  1064. ptr1 = ptr;
  1065. dst = tmp;
  1066. for(j=len;j>0;j--) {
  1067. *dst++ = ptr[0*len];
  1068. *dst++ = ptr[1*len];
  1069. *dst++ = ptr[2*len];
  1070. ptr++;
  1071. }
  1072. ptr+=2*len;
  1073. memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
  1074. }
  1075. }
  1076. #define ISQRT2 FIXR(0.70710678118654752440)
  1077. static void compute_stereo(MPADecodeContext *s,
  1078. GranuleDef *g0, GranuleDef *g1)
  1079. {
  1080. int i, j, k, l;
  1081. int sf_max, sf, len, non_zero_found;
  1082. INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;
  1083. int non_zero_found_short[3];
  1084. /* intensity stereo */
  1085. if (s->mode_ext & MODE_EXT_I_STEREO) {
  1086. if (!s->lsf) {
  1087. is_tab = is_table;
  1088. sf_max = 7;
  1089. } else {
  1090. is_tab = is_table_lsf[g1->scalefac_compress & 1];
  1091. sf_max = 16;
  1092. }
  1093. tab0 = g0->sb_hybrid + 576;
  1094. tab1 = g1->sb_hybrid + 576;
  1095. non_zero_found_short[0] = 0;
  1096. non_zero_found_short[1] = 0;
  1097. non_zero_found_short[2] = 0;
  1098. k = (13 - g1->short_start) * 3 + g1->long_end - 3;
  1099. for(i = 12;i >= g1->short_start;i--) {
  1100. /* for last band, use previous scale factor */
  1101. if (i != 11)
  1102. k -= 3;
  1103. len = band_size_short[s->sample_rate_index][i];
  1104. for(l=2;l>=0;l--) {
  1105. tab0 -= len;
  1106. tab1 -= len;
  1107. if (!non_zero_found_short[l]) {
  1108. /* test if non zero band. if so, stop doing i-stereo */
  1109. for(j=0;j<len;j++) {
  1110. if (tab1[j] != 0) {
  1111. non_zero_found_short[l] = 1;
  1112. goto found1;
  1113. }
  1114. }
  1115. sf = g1->scale_factors[k + l];
  1116. if (sf >= sf_max)
  1117. goto found1;
  1118. v1 = is_tab[0][sf];
  1119. v2 = is_tab[1][sf];
  1120. for(j=0;j<len;j++) {
  1121. tmp0 = tab0[j];
  1122. tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
  1123. tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
  1124. }
  1125. } else {
  1126. found1:
  1127. if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1128. /* lower part of the spectrum : do ms stereo
  1129. if enabled */
  1130. for(j=0;j<len;j++) {
  1131. tmp0 = tab0[j];
  1132. tmp1 = tab1[j];
  1133. tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
  1134. tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
  1135. }
  1136. }
  1137. }
  1138. }
  1139. }
  1140. non_zero_found = non_zero_found_short[0] |
  1141. non_zero_found_short[1] |
  1142. non_zero_found_short[2];
  1143. for(i = g1->long_end - 1;i >= 0;i--) {
  1144. len = band_size_long[s->sample_rate_index][i];
  1145. tab0 -= len;
  1146. tab1 -= len;
  1147. /* test if non zero band. if so, stop doing i-stereo */
  1148. if (!non_zero_found) {
  1149. for(j=0;j<len;j++) {
  1150. if (tab1[j] != 0) {
  1151. non_zero_found = 1;
  1152. goto found2;
  1153. }
  1154. }
  1155. /* for last band, use previous scale factor */
  1156. k = (i == 21) ? 20 : i;
  1157. sf = g1->scale_factors[k];
  1158. if (sf >= sf_max)
  1159. goto found2;
  1160. v1 = is_tab[0][sf];
  1161. v2 = is_tab[1][sf];
  1162. for(j=0;j<len;j++) {
  1163. tmp0 = tab0[j];
  1164. tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
  1165. tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
  1166. }
  1167. } else {
  1168. found2:
  1169. if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1170. /* lower part of the spectrum : do ms stereo
  1171. if enabled */
  1172. for(j=0;j<len;j++) {
  1173. tmp0 = tab0[j];
  1174. tmp1 = tab1[j];
  1175. tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
  1176. tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
  1177. }
  1178. }
  1179. }
  1180. }
  1181. } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1182. /* ms stereo ONLY */
  1183. /* NOTE: the 1/sqrt(2) normalization factor is included in the
  1184. global gain */
  1185. tab0 = g0->sb_hybrid;
  1186. tab1 = g1->sb_hybrid;
  1187. for(i=0;i<576;i++) {
  1188. tmp0 = tab0[i];
  1189. tmp1 = tab1[i];
  1190. tab0[i] = tmp0 + tmp1;
  1191. tab1[i] = tmp0 - tmp1;
  1192. }
  1193. }
  1194. }
  1195. #if !CONFIG_FLOAT
  1196. static void compute_antialias_fixed(MPADecodeContext *s, GranuleDef *g)
  1197. {
  1198. int32_t *ptr, *csa;
  1199. int n, i;
  1200. /* we antialias only "long" bands */
  1201. if (g->block_type == 2) {
  1202. if (!g->switch_point)
  1203. return;
  1204. /* XXX: check this for 8000Hz case */
  1205. n = 1;
  1206. } else {
  1207. n = SBLIMIT - 1;
  1208. }
  1209. ptr = g->sb_hybrid + 18;
  1210. for(i = n;i > 0;i--) {
  1211. int tmp0, tmp1, tmp2;
  1212. csa = &csa_table[0][0];
  1213. #define INT_AA(j) \
  1214. tmp0 = ptr[-1-j];\
  1215. tmp1 = ptr[ j];\
  1216. tmp2= MULH(tmp0 + tmp1, csa[0+4*j]);\
  1217. ptr[-1-j] = 4*(tmp2 - MULH(tmp1, csa[2+4*j]));\
  1218. ptr[ j] = 4*(tmp2 + MULH(tmp0, csa[3+4*j]));
  1219. INT_AA(0)
  1220. INT_AA(1)
  1221. INT_AA(2)
  1222. INT_AA(3)
  1223. INT_AA(4)
  1224. INT_AA(5)
  1225. INT_AA(6)
  1226. INT_AA(7)
  1227. ptr += 18;
  1228. }
  1229. }
  1230. #endif
  1231. static void compute_imdct(MPADecodeContext *s,
  1232. GranuleDef *g,
  1233. INTFLOAT *sb_samples,
  1234. INTFLOAT *mdct_buf)
  1235. {
  1236. INTFLOAT *win, *win1, *out_ptr, *ptr, *buf, *ptr1;
  1237. INTFLOAT out2[12];
  1238. int i, j, mdct_long_end, sblimit;
  1239. /* find last non zero block */
  1240. ptr = g->sb_hybrid + 576;
  1241. ptr1 = g->sb_hybrid + 2 * 18;
  1242. while (ptr >= ptr1) {
  1243. int32_t *p;
  1244. ptr -= 6;
  1245. p= (int32_t*)ptr;
  1246. if(p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
  1247. break;
  1248. }
  1249. sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
  1250. if (g->block_type == 2) {
  1251. /* XXX: check for 8000 Hz */
  1252. if (g->switch_point)
  1253. mdct_long_end = 2;
  1254. else
  1255. mdct_long_end = 0;
  1256. } else {
  1257. mdct_long_end = sblimit;
  1258. }
  1259. buf = mdct_buf;
  1260. ptr = g->sb_hybrid;
  1261. for(j=0;j<mdct_long_end;j++) {
  1262. /* apply window & overlap with previous buffer */
  1263. out_ptr = sb_samples + j;
  1264. /* select window */
  1265. if (g->switch_point && j < 2)
  1266. win1 = mdct_win[0];
  1267. else
  1268. win1 = mdct_win[g->block_type];
  1269. /* select frequency inversion */
  1270. win = win1 + ((4 * 36) & -(j & 1));
  1271. imdct36(out_ptr, buf, ptr, win);
  1272. out_ptr += 18*SBLIMIT;
  1273. ptr += 18;
  1274. buf += 18;
  1275. }
  1276. for(j=mdct_long_end;j<sblimit;j++) {
  1277. /* select frequency inversion */
  1278. win = mdct_win[2] + ((4 * 36) & -(j & 1));
  1279. out_ptr = sb_samples + j;
  1280. for(i=0; i<6; i++){
  1281. *out_ptr = buf[i];
  1282. out_ptr += SBLIMIT;
  1283. }
  1284. imdct12(out2, ptr + 0);
  1285. for(i=0;i<6;i++) {
  1286. *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[i + 6*1];
  1287. buf[i + 6*2] = MULH3(out2[i + 6], win[i + 6], 1);
  1288. out_ptr += SBLIMIT;
  1289. }
  1290. imdct12(out2, ptr + 1);
  1291. for(i=0;i<6;i++) {
  1292. *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[i + 6*2];
  1293. buf[i + 6*0] = MULH3(out2[i + 6], win[i + 6], 1);
  1294. out_ptr += SBLIMIT;
  1295. }
  1296. imdct12(out2, ptr + 2);
  1297. for(i=0;i<6;i++) {
  1298. buf[i + 6*0] = MULH3(out2[i ], win[i ], 1) + buf[i + 6*0];
  1299. buf[i + 6*1] = MULH3(out2[i + 6], win[i + 6], 1);
  1300. buf[i + 6*2] = 0;
  1301. }
  1302. ptr += 18;
  1303. buf += 18;
  1304. }
  1305. /* zero bands */
  1306. for(j=sblimit;j<SBLIMIT;j++) {
  1307. /* overlap */
  1308. out_ptr = sb_samples + j;
  1309. for(i=0;i<18;i++) {
  1310. *out_ptr = buf[i];
  1311. buf[i] = 0;
  1312. out_ptr += SBLIMIT;
  1313. }
  1314. buf += 18;
  1315. }
  1316. }
  1317. /* main layer3 decoding function */
  1318. static int mp_decode_layer3(MPADecodeContext *s)
  1319. {
  1320. int nb_granules, main_data_begin, private_bits;
  1321. int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
  1322. GranuleDef *g;
  1323. int16_t exponents[576]; //FIXME try INTFLOAT
  1324. /* read side info */
  1325. if (s->lsf) {
  1326. main_data_begin = get_bits(&s->gb, 8);
  1327. private_bits = get_bits(&s->gb, s->nb_channels);
  1328. nb_granules = 1;
  1329. } else {
  1330. main_data_begin = get_bits(&s->gb, 9);
  1331. if (s->nb_channels == 2)
  1332. private_bits = get_bits(&s->gb, 3);
  1333. else
  1334. private_bits = get_bits(&s->gb, 5);
  1335. nb_granules = 2;
  1336. for(ch=0;ch<s->nb_channels;ch++) {
  1337. s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
  1338. s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
  1339. }
  1340. }
  1341. for(gr=0;gr<nb_granules;gr++) {
  1342. for(ch=0;ch<s->nb_channels;ch++) {
  1343. av_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
  1344. g = &s->granules[ch][gr];
  1345. g->part2_3_length = get_bits(&s->gb, 12);
  1346. g->big_values = get_bits(&s->gb, 9);
  1347. if(g->big_values > 288){
  1348. av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
  1349. return -1;
  1350. }
  1351. g->global_gain = get_bits(&s->gb, 8);
  1352. /* if MS stereo only is selected, we precompute the
  1353. 1/sqrt(2) renormalization factor */
  1354. if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
  1355. MODE_EXT_MS_STEREO)
  1356. g->global_gain -= 2;
  1357. if (s->lsf)
  1358. g->scalefac_compress = get_bits(&s->gb, 9);
  1359. else
  1360. g->scalefac_compress = get_bits(&s->gb, 4);
  1361. blocksplit_flag = get_bits1(&s->gb);
  1362. if (blocksplit_flag) {
  1363. g->block_type = get_bits(&s->gb, 2);
  1364. if (g->block_type == 0){
  1365. av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
  1366. return -1;
  1367. }
  1368. g->switch_point = get_bits1(&s->gb);
  1369. for(i=0;i<2;i++)
  1370. g->table_select[i] = get_bits(&s->gb, 5);
  1371. for(i=0;i<3;i++)
  1372. g->subblock_gain[i] = get_bits(&s->gb, 3);
  1373. ff_init_short_region(s, g);
  1374. } else {
  1375. int region_address1, region_address2;
  1376. g->block_type = 0;
  1377. g->switch_point = 0;
  1378. for(i=0;i<3;i++)
  1379. g->table_select[i] = get_bits(&s->gb, 5);
  1380. /* compute huffman coded region sizes */
  1381. region_address1 = get_bits(&s->gb, 4);
  1382. region_address2 = get_bits(&s->gb, 3);
  1383. av_dlog(s->avctx, "region1=%d region2=%d\n",
  1384. region_address1, region_address2);
  1385. ff_init_long_region(s, g, region_address1, region_address2);
  1386. }
  1387. ff_region_offset2size(g);
  1388. ff_compute_band_indexes(s, g);
  1389. g->preflag = 0;
  1390. if (!s->lsf)
  1391. g->preflag = get_bits1(&s->gb);
  1392. g->scalefac_scale = get_bits1(&s->gb);
  1393. g->count1table_select = get_bits1(&s->gb);
  1394. av_dlog(s->avctx, "block_type=%d switch_point=%d\n",
  1395. g->block_type, g->switch_point);
  1396. }
  1397. }
  1398. if (!s->adu_mode) {
  1399. const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
  1400. assert((get_bits_count(&s->gb) & 7) == 0);
  1401. /* now we get bits from the main_data_begin offset */
  1402. av_dlog(s->avctx, "seekback: %d\n", main_data_begin);
  1403. //av_log(NULL, AV_LOG_ERROR, "backstep:%d, lastbuf:%d\n", main_data_begin, s->last_buf_size);
  1404. memcpy(s->last_buf + s->last_buf_size, ptr, EXTRABYTES);
  1405. s->in_gb= s->gb;
  1406. init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);
  1407. skip_bits_long(&s->gb, 8*(s->last_buf_size - main_data_begin));
  1408. }
  1409. for(gr=0;gr<nb_granules;gr++) {
  1410. for(ch=0;ch<s->nb_channels;ch++) {
  1411. g = &s->granules[ch][gr];
  1412. if(get_bits_count(&s->gb)<0){
  1413. av_log(s->avctx, AV_LOG_DEBUG, "mdb:%d, lastbuf:%d skipping granule %d\n",
  1414. main_data_begin, s->last_buf_size, gr);
  1415. skip_bits_long(&s->gb, g->part2_3_length);
  1416. memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
  1417. if(get_bits_count(&s->gb) >= s->gb.size_in_bits && s->in_gb.buffer){
  1418. skip_bits_long(&s->in_gb, get_bits_count(&s->gb) - s->gb.size_in_bits);
  1419. s->gb= s->in_gb;
  1420. s->in_gb.buffer=NULL;
  1421. }
  1422. continue;
  1423. }
  1424. bits_pos = get_bits_count(&s->gb);
  1425. if (!s->lsf) {
  1426. uint8_t *sc;
  1427. int slen, slen1, slen2;
  1428. /* MPEG1 scale factors */
  1429. slen1 = slen_table[0][g->scalefac_compress];
  1430. slen2 = slen_table[1][g->scalefac_compress];
  1431. av_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
  1432. if (g->block_type == 2) {
  1433. n = g->switch_point ? 17 : 18;
  1434. j = 0;
  1435. if(slen1){
  1436. for(i=0;i<n;i++)
  1437. g->scale_factors[j++] = get_bits(&s->gb, slen1);
  1438. }else{
  1439. for(i=0;i<n;i++)
  1440. g->scale_factors[j++] = 0;
  1441. }
  1442. if(slen2){
  1443. for(i=0;i<18;i++)
  1444. g->scale_factors[j++] = get_bits(&s->gb, slen2);
  1445. for(i=0;i<3;i++)
  1446. g->scale_factors[j++] = 0;
  1447. }else{
  1448. for(i=0;i<21;i++)
  1449. g->scale_factors[j++] = 0;
  1450. }
  1451. } else {
  1452. sc = s->granules[ch][0].scale_factors;
  1453. j = 0;
  1454. for(k=0;k<4;k++) {
  1455. n = (k == 0 ? 6 : 5);
  1456. if ((g->scfsi & (0x8 >> k)) == 0) {
  1457. slen = (k < 2) ? slen1 : slen2;
  1458. if(slen){
  1459. for(i=0;i<n;i++)
  1460. g->scale_factors[j++] = get_bits(&s->gb, slen);
  1461. }else{
  1462. for(i=0;i<n;i++)
  1463. g->scale_factors[j++] = 0;
  1464. }
  1465. } else {
  1466. /* simply copy from last granule */
  1467. for(i=0;i<n;i++) {
  1468. g->scale_factors[j] = sc[j];
  1469. j++;
  1470. }
  1471. }
  1472. }
  1473. g->scale_factors[j++] = 0;
  1474. }
  1475. } else {
  1476. int tindex, tindex2, slen[4], sl, sf;
  1477. /* LSF scale factors */
  1478. if (g->block_type == 2) {
  1479. tindex = g->switch_point ? 2 : 1;
  1480. } else {
  1481. tindex = 0;
  1482. }
  1483. sf = g->scalefac_compress;
  1484. if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
  1485. /* intensity stereo case */
  1486. sf >>= 1;
  1487. if (sf < 180) {
  1488. lsf_sf_expand(slen, sf, 6, 6, 0);
  1489. tindex2 = 3;
  1490. } else if (sf < 244) {
  1491. lsf_sf_expand(slen, sf - 180, 4, 4, 0);
  1492. tindex2 = 4;
  1493. } else {
  1494. lsf_sf_expand(slen, sf - 244, 3, 0, 0);
  1495. tindex2 = 5;
  1496. }
  1497. } else {
  1498. /* normal case */
  1499. if (sf < 400) {
  1500. lsf_sf_expand(slen, sf, 5, 4, 4);
  1501. tindex2 = 0;
  1502. } else if (sf < 500) {
  1503. lsf_sf_expand(slen, sf - 400, 5, 4, 0);
  1504. tindex2 = 1;
  1505. } else {
  1506. lsf_sf_expand(slen, sf - 500, 3, 0, 0);
  1507. tindex2 = 2;
  1508. g->preflag = 1;
  1509. }
  1510. }
  1511. j = 0;
  1512. for(k=0;k<4;k++) {
  1513. n = lsf_nsf_table[tindex2][tindex][k];
  1514. sl = slen[k];
  1515. if(sl){
  1516. for(i=0;i<n;i++)
  1517. g->scale_factors[j++] = get_bits(&s->gb, sl);
  1518. }else{
  1519. for(i=0;i<n;i++)
  1520. g->scale_factors[j++] = 0;
  1521. }
  1522. }
  1523. /* XXX: should compute exact size */
  1524. for(;j<40;j++)
  1525. g->scale_factors[j] = 0;
  1526. }
  1527. exponents_from_scale_factors(s, g, exponents);
  1528. /* read Huffman coded residue */
  1529. huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
  1530. } /* ch */
  1531. if (s->nb_channels == 2)
  1532. compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
  1533. for(ch=0;ch<s->nb_channels;ch++) {
  1534. g = &s->granules[ch][gr];
  1535. reorder_block(s, g);
  1536. RENAME(compute_antialias)(s, g);
  1537. compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
  1538. }
  1539. } /* gr */
  1540. if(get_bits_count(&s->gb)<0)
  1541. skip_bits_long(&s->gb, -get_bits_count(&s->gb));
  1542. return nb_granules * 18;
  1543. }
  1544. static int mp_decode_frame(MPADecodeContext *s,
  1545. OUT_INT *samples, const uint8_t *buf, int buf_size)
  1546. {
  1547. int i, nb_frames, ch;
  1548. OUT_INT *samples_ptr;
  1549. init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE)*8);
  1550. /* skip error protection field */
  1551. if (s->error_protection)
  1552. skip_bits(&s->gb, 16);
  1553. switch(s->layer) {
  1554. case 1:
  1555. s->avctx->frame_size = 384;
  1556. nb_frames = mp_decode_layer1(s);
  1557. break;
  1558. case 2:
  1559. s->avctx->frame_size = 1152;
  1560. nb_frames = mp_decode_layer2(s);
  1561. break;
  1562. case 3:
  1563. s->avctx->frame_size = s->lsf ? 576 : 1152;
  1564. default:
  1565. nb_frames = mp_decode_layer3(s);
  1566. s->last_buf_size=0;
  1567. if(s->in_gb.buffer){
  1568. align_get_bits(&s->gb);
  1569. i= get_bits_left(&s->gb)>>3;
  1570. if(i >= 0 && i <= BACKSTEP_SIZE){
  1571. memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);
  1572. s->last_buf_size=i;
  1573. }else
  1574. av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
  1575. s->gb= s->in_gb;
  1576. s->in_gb.buffer= NULL;
  1577. }
  1578. align_get_bits(&s->gb);
  1579. assert((get_bits_count(&s->gb) & 7) == 0);
  1580. i= get_bits_left(&s->gb)>>3;
  1581. if(i<0 || i > BACKSTEP_SIZE || nb_frames<0){
  1582. if(i<0)
  1583. av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
  1584. i= FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
  1585. }
  1586. assert(i <= buf_size - HEADER_SIZE && i>= 0);
  1587. memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
  1588. s->last_buf_size += i;
  1589. break;
  1590. }
  1591. /* apply the synthesis filter */
  1592. for(ch=0;ch<s->nb_channels;ch++) {
  1593. samples_ptr = samples + ch;
  1594. for(i=0;i<nb_frames;i++) {
  1595. RENAME(ff_mpa_synth_filter)(
  1596. &s->mpadsp,
  1597. s->synth_buf[ch], &(s->synth_buf_offset[ch]),
  1598. RENAME(ff_mpa_synth_window), &s->dither_state,
  1599. samples_ptr, s->nb_channels,
  1600. s->sb_samples[ch][i]);
  1601. samples_ptr += 32 * s->nb_channels;
  1602. }
  1603. }
  1604. return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
  1605. }
  1606. static int decode_frame(AVCodecContext * avctx,
  1607. void *data, int *data_size,
  1608. AVPacket *avpkt)
  1609. {
  1610. const uint8_t *buf = avpkt->data;
  1611. int buf_size = avpkt->size;
  1612. MPADecodeContext *s = avctx->priv_data;
  1613. uint32_t header;
  1614. int out_size;
  1615. OUT_INT *out_samples = data;
  1616. if(buf_size < HEADER_SIZE)
  1617. return -1;
  1618. header = AV_RB32(buf);
  1619. if(ff_mpa_check_header(header) < 0){
  1620. av_log(avctx, AV_LOG_ERROR, "Header missing\n");
  1621. return -1;
  1622. }
  1623. if (ff_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
  1624. /* free format: prepare to compute frame size */
  1625. s->frame_size = -1;
  1626. return -1;
  1627. }
  1628. /* update codec info */
  1629. avctx->channels = s->nb_channels;
  1630. avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
  1631. if (!avctx->bit_rate)
  1632. avctx->bit_rate = s->bit_rate;
  1633. avctx->sub_id = s->layer;
  1634. if(*data_size < 1152*avctx->channels*sizeof(OUT_INT))
  1635. return -1;
  1636. *data_size = 0;
  1637. if(s->frame_size<=0 || s->frame_size > buf_size){
  1638. av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
  1639. return -1;
  1640. }else if(s->frame_size < buf_size){
  1641. av_log(avctx, AV_LOG_ERROR, "incorrect frame size\n");
  1642. buf_size= s->frame_size;
  1643. }
  1644. out_size = mp_decode_frame(s, out_samples, buf, buf_size);
  1645. if(out_size>=0){
  1646. *data_size = out_size;
  1647. avctx->sample_rate = s->sample_rate;
  1648. //FIXME maybe move the other codec info stuff from above here too
  1649. }else
  1650. av_log(avctx, AV_LOG_DEBUG, "Error while decoding MPEG audio frame.\n"); //FIXME return -1 / but also return the number of bytes consumed
  1651. s->frame_size = 0;
  1652. return buf_size;
  1653. }
  1654. static void flush(AVCodecContext *avctx){
  1655. MPADecodeContext *s = avctx->priv_data;
  1656. memset(s->synth_buf, 0, sizeof(s->synth_buf));
  1657. s->last_buf_size= 0;
  1658. }
  1659. #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
  1660. static int decode_frame_adu(AVCodecContext * avctx,
  1661. void *data, int *data_size,
  1662. AVPacket *avpkt)
  1663. {
  1664. const uint8_t *buf = avpkt->data;
  1665. int buf_size = avpkt->size;
  1666. MPADecodeContext *s = avctx->priv_data;
  1667. uint32_t header;
  1668. int len, out_size;
  1669. OUT_INT *out_samples = data;
  1670. len = buf_size;
  1671. // Discard too short frames
  1672. if (buf_size < HEADER_SIZE) {
  1673. *data_size = 0;
  1674. return buf_size;
  1675. }
  1676. if (len > MPA_MAX_CODED_FRAME_SIZE)
  1677. len = MPA_MAX_CODED_FRAME_SIZE;
  1678. // Get header and restore sync word
  1679. header = AV_RB32(buf) | 0xffe00000;
  1680. if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame
  1681. *data_size = 0;
  1682. return buf_size;
  1683. }
  1684. ff_mpegaudio_decode_header((MPADecodeHeader *)s, header);
  1685. /* update codec info */
  1686. avctx->sample_rate = s->sample_rate;
  1687. avctx->channels = s->nb_channels;
  1688. if (!avctx->bit_rate)
  1689. avctx->bit_rate = s->bit_rate;
  1690. avctx->sub_id = s->layer;
  1691. s->frame_size = len;
  1692. if (avctx->parse_only) {
  1693. out_size = buf_size;
  1694. } else {
  1695. out_size = mp_decode_frame(s, out_samples, buf, buf_size);
  1696. }
  1697. *data_size = out_size;
  1698. return buf_size;
  1699. }
  1700. #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
  1701. #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
  1702. /**
  1703. * Context for MP3On4 decoder
  1704. */
  1705. typedef struct MP3On4DecodeContext {
  1706. int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
  1707. int syncword; ///< syncword patch
  1708. const uint8_t *coff; ///< channels offsets in output buffer
  1709. MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
  1710. } MP3On4DecodeContext;
  1711. #include "mpeg4audio.h"
  1712. /* Next 3 arrays are indexed by channel config number (passed via codecdata) */
  1713. static const uint8_t mp3Frames[8] = {0,1,1,2,3,3,4,5}; /* number of mp3 decoder instances */
  1714. /* offsets into output buffer, assume output order is FL FR BL BR C LFE */
  1715. static const uint8_t chan_offset[8][5] = {
  1716. {0},
  1717. {0}, // C
  1718. {0}, // FLR
  1719. {2,0}, // C FLR
  1720. {2,0,3}, // C FLR BS
  1721. {4,0,2}, // C FLR BLRS
  1722. {4,0,2,5}, // C FLR BLRS LFE
  1723. {4,0,2,6,5}, // C FLR BLRS BLR LFE
  1724. };
  1725. static int decode_init_mp3on4(AVCodecContext * avctx)
  1726. {
  1727. MP3On4DecodeContext *s = avctx->priv_data;
  1728. MPEG4AudioConfig cfg;
  1729. int i;
  1730. if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) {
  1731. av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
  1732. return -1;
  1733. }
  1734. ff_mpeg4audio_get_config(&cfg, avctx->extradata, avctx->extradata_size);
  1735. if (!cfg.chan_config || cfg.chan_config > 7) {
  1736. av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
  1737. return -1;
  1738. }
  1739. s->frames = mp3Frames[cfg.chan_config];
  1740. s->coff = chan_offset[cfg.chan_config];
  1741. avctx->channels = ff_mpeg4audio_channels[cfg.chan_config];
  1742. if (cfg.sample_rate < 16000)
  1743. s->syncword = 0xffe00000;
  1744. else
  1745. s->syncword = 0xfff00000;
  1746. /* Init the first mp3 decoder in standard way, so that all tables get builded
  1747. * We replace avctx->priv_data with the context of the first decoder so that
  1748. * decode_init() does not have to be changed.
  1749. * Other decoders will be initialized here copying data from the first context
  1750. */
  1751. // Allocate zeroed memory for the first decoder context
  1752. s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
  1753. // Put decoder context in place to make init_decode() happy
  1754. avctx->priv_data = s->mp3decctx[0];
  1755. decode_init(avctx);
  1756. // Restore mp3on4 context pointer
  1757. avctx->priv_data = s;
  1758. s->mp3decctx[0]->adu_mode = 1; // Set adu mode
  1759. /* Create a separate codec/context for each frame (first is already ok).
  1760. * Each frame is 1 or 2 channels - up to 5 frames allowed
  1761. */
  1762. for (i = 1; i < s->frames; i++) {
  1763. s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
  1764. s->mp3decctx[i]->adu_mode = 1;
  1765. s->mp3decctx[i]->avctx = avctx;
  1766. }
  1767. return 0;
  1768. }
  1769. static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
  1770. {
  1771. MP3On4DecodeContext *s = avctx->priv_data;
  1772. int i;
  1773. for (i = 0; i < s->frames; i++)
  1774. av_free(s->mp3decctx[i]);
  1775. return 0;
  1776. }
  1777. static int decode_frame_mp3on4(AVCodecContext * avctx,
  1778. void *data, int *data_size,
  1779. AVPacket *avpkt)
  1780. {
  1781. const uint8_t *buf = avpkt->data;
  1782. int buf_size = avpkt->size;
  1783. MP3On4DecodeContext *s = avctx->priv_data;
  1784. MPADecodeContext *m;
  1785. int fsize, len = buf_size, out_size = 0;
  1786. uint32_t header;
  1787. OUT_INT *out_samples = data;
  1788. OUT_INT decoded_buf[MPA_FRAME_SIZE * MPA_MAX_CHANNELS];
  1789. OUT_INT *outptr, *bp;
  1790. int fr, j, n;
  1791. if(*data_size < MPA_FRAME_SIZE * MPA_MAX_CHANNELS * s->frames * sizeof(OUT_INT))
  1792. return -1;
  1793. *data_size = 0;
  1794. // Discard too short frames
  1795. if (buf_size < HEADER_SIZE)
  1796. return -1;
  1797. // If only one decoder interleave is not needed
  1798. outptr = s->frames == 1 ? out_samples : decoded_buf;
  1799. avctx->bit_rate = 0;
  1800. for (fr = 0; fr < s->frames; fr++) {
  1801. fsize = AV_RB16(buf) >> 4;
  1802. fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
  1803. m = s->mp3decctx[fr];
  1804. assert (m != NULL);
  1805. header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
  1806. if (ff_mpa_check_header(header) < 0) // Bad header, discard block
  1807. break;
  1808. ff_mpegaudio_decode_header((MPADecodeHeader *)m, header);
  1809. out_size += mp_decode_frame(m, outptr, buf, fsize);
  1810. buf += fsize;
  1811. len -= fsize;
  1812. if(s->frames > 1) {
  1813. n = m->avctx->frame_size*m->nb_channels;
  1814. /* interleave output data */
  1815. bp = out_samples + s->coff[fr];
  1816. if(m->nb_channels == 1) {
  1817. for(j = 0; j < n; j++) {
  1818. *bp = decoded_buf[j];
  1819. bp += avctx->channels;
  1820. }
  1821. } else {
  1822. for(j = 0; j < n; j++) {
  1823. bp[0] = decoded_buf[j++];
  1824. bp[1] = decoded_buf[j];
  1825. bp += avctx->channels;
  1826. }
  1827. }
  1828. }
  1829. avctx->bit_rate += m->bit_rate;
  1830. }
  1831. /* update codec info */
  1832. avctx->sample_rate = s->mp3decctx[0]->sample_rate;
  1833. *data_size = out_size;
  1834. return buf_size;
  1835. }
  1836. #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
  1837. #if !CONFIG_FLOAT
  1838. #if CONFIG_MP1_DECODER
  1839. AVCodec ff_mp1_decoder =
  1840. {
  1841. "mp1",
  1842. AVMEDIA_TYPE_AUDIO,
  1843. CODEC_ID_MP1,
  1844. sizeof(MPADecodeContext),
  1845. decode_init,
  1846. NULL,
  1847. NULL,
  1848. decode_frame,
  1849. CODEC_CAP_PARSE_ONLY,
  1850. .flush= flush,
  1851. .long_name= NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
  1852. };
  1853. #endif
  1854. #if CONFIG_MP2_DECODER
  1855. AVCodec ff_mp2_decoder =
  1856. {
  1857. "mp2",
  1858. AVMEDIA_TYPE_AUDIO,
  1859. CODEC_ID_MP2,
  1860. sizeof(MPADecodeContext),
  1861. decode_init,
  1862. NULL,
  1863. NULL,
  1864. decode_frame,
  1865. CODEC_CAP_PARSE_ONLY,
  1866. .flush= flush,
  1867. .long_name= NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
  1868. };
  1869. #endif
  1870. #if CONFIG_MP3_DECODER
  1871. AVCodec ff_mp3_decoder =
  1872. {
  1873. "mp3",
  1874. AVMEDIA_TYPE_AUDIO,
  1875. CODEC_ID_MP3,
  1876. sizeof(MPADecodeContext),
  1877. decode_init,
  1878. NULL,
  1879. NULL,
  1880. decode_frame,
  1881. CODEC_CAP_PARSE_ONLY,
  1882. .flush= flush,
  1883. .long_name= NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
  1884. };
  1885. #endif
  1886. #if CONFIG_MP3ADU_DECODER
  1887. AVCodec ff_mp3adu_decoder =
  1888. {
  1889. "mp3adu",
  1890. AVMEDIA_TYPE_AUDIO,
  1891. CODEC_ID_MP3ADU,
  1892. sizeof(MPADecodeContext),
  1893. decode_init,
  1894. NULL,
  1895. NULL,
  1896. decode_frame_adu,
  1897. CODEC_CAP_PARSE_ONLY,
  1898. .flush= flush,
  1899. .long_name= NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
  1900. };
  1901. #endif
  1902. #if CONFIG_MP3ON4_DECODER
  1903. AVCodec ff_mp3on4_decoder =
  1904. {
  1905. "mp3on4",
  1906. AVMEDIA_TYPE_AUDIO,
  1907. CODEC_ID_MP3ON4,
  1908. sizeof(MP3On4DecodeContext),
  1909. decode_init_mp3on4,
  1910. NULL,
  1911. decode_close_mp3on4,
  1912. decode_frame_mp3on4,
  1913. .flush= flush,
  1914. .long_name= NULL_IF_CONFIG_SMALL("MP3onMP4"),
  1915. };
  1916. #endif
  1917. #endif