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