You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.

2186 lines
71KB

  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. int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
  71. int dither_state;
  72. int err_recognition;
  73. AVCodecContext* avctx;
  74. MPADSPContext mpadsp;
  75. } MPADecodeContext;
  76. #if CONFIG_FLOAT
  77. # define SHR(a,b) ((a)*(1.0f/(1<<(b))))
  78. # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
  79. # define FIXR(x) ((float)(x))
  80. # define FIXHR(x) ((float)(x))
  81. # define MULH3(x, y, s) ((s)*(y)*(x))
  82. # define MULLx(x, y, s) ((y)*(x))
  83. # define RENAME(a) a ## _float
  84. # define OUT_FMT AV_SAMPLE_FMT_FLT
  85. #else
  86. # define SHR(a,b) ((a)>>(b))
  87. /* WARNING: only correct for positive numbers */
  88. # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
  89. # define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
  90. # define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
  91. # define MULH3(x, y, s) MULH((s)*(x), y)
  92. # define MULLx(x, y, s) MULL(x,y,s)
  93. # define RENAME(a) a ## _fixed
  94. # define OUT_FMT AV_SAMPLE_FMT_S16
  95. #endif
  96. /****************/
  97. #define HEADER_SIZE 4
  98. #include "mpegaudiodata.h"
  99. #include "mpegaudiodectab.h"
  100. /* vlc structure for decoding layer 3 huffman tables */
  101. static VLC huff_vlc[16];
  102. static VLC_TYPE huff_vlc_tables[
  103. 0 + 128 + 128 + 128 + 130 + 128 + 154 + 166 +
  104. 142 + 204 + 190 + 170 + 542 + 460 + 662 + 414
  105. ][2];
  106. static const int huff_vlc_tables_sizes[16] = {
  107. 0, 128, 128, 128, 130, 128, 154, 166,
  108. 142, 204, 190, 170, 542, 460, 662, 414
  109. };
  110. static VLC huff_quad_vlc[2];
  111. static VLC_TYPE huff_quad_vlc_tables[128+16][2];
  112. static const int huff_quad_vlc_tables_sizes[2] = { 128, 16 };
  113. /* computed from band_size_long */
  114. static uint16_t band_index_long[9][23];
  115. #include "mpegaudio_tablegen.h"
  116. /* intensity stereo coef table */
  117. static INTFLOAT is_table[2][16];
  118. static INTFLOAT is_table_lsf[2][2][16];
  119. static INTFLOAT csa_table[8][4];
  120. static INTFLOAT mdct_win[8][36];
  121. static int16_t division_tab3[1<<6 ];
  122. static int16_t division_tab5[1<<8 ];
  123. static int16_t division_tab9[1<<11];
  124. static int16_t * const division_tabs[4] = {
  125. division_tab3, division_tab5, NULL, division_tab9
  126. };
  127. /* lower 2 bits: modulo 3, higher bits: shift */
  128. static uint16_t scale_factor_modshift[64];
  129. /* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
  130. static int32_t scale_factor_mult[15][3];
  131. /* mult table for layer 2 group quantization */
  132. #define SCALE_GEN(v) \
  133. { FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
  134. static const int32_t scale_factor_mult2[3][3] = {
  135. SCALE_GEN(4.0 / 3.0), /* 3 steps */
  136. SCALE_GEN(4.0 / 5.0), /* 5 steps */
  137. SCALE_GEN(4.0 / 9.0), /* 9 steps */
  138. };
  139. /**
  140. * Convert region offsets to region sizes and truncate
  141. * size to big_values.
  142. */
  143. static void ff_region_offset2size(GranuleDef *g)
  144. {
  145. int i, k, j = 0;
  146. g->region_size[2] = 576 / 2;
  147. for (i = 0; i < 3; i++) {
  148. k = FFMIN(g->region_size[i], g->big_values);
  149. g->region_size[i] = k - j;
  150. j = k;
  151. }
  152. }
  153. static void ff_init_short_region(MPADecodeContext *s, GranuleDef *g)
  154. {
  155. if (g->block_type == 2)
  156. g->region_size[0] = (36 / 2);
  157. else {
  158. if (s->sample_rate_index <= 2)
  159. g->region_size[0] = (36 / 2);
  160. else if (s->sample_rate_index != 8)
  161. g->region_size[0] = (54 / 2);
  162. else
  163. g->region_size[0] = (108 / 2);
  164. }
  165. g->region_size[1] = (576 / 2);
  166. }
  167. static void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2)
  168. {
  169. int l;
  170. g->region_size[0] = band_index_long[s->sample_rate_index][ra1 + 1] >> 1;
  171. /* should not overflow */
  172. l = FFMIN(ra1 + ra2 + 2, 22);
  173. g->region_size[1] = band_index_long[s->sample_rate_index][ l] >> 1;
  174. }
  175. static void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g)
  176. {
  177. if (g->block_type == 2) {
  178. if (g->switch_point) {
  179. /* if switched mode, we handle the 36 first samples as
  180. long blocks. For 8000Hz, we handle the 48 first
  181. exponents as long blocks (XXX: check this!) */
  182. if (s->sample_rate_index <= 2)
  183. g->long_end = 8;
  184. else if (s->sample_rate_index != 8)
  185. g->long_end = 6;
  186. else
  187. g->long_end = 4; /* 8000 Hz */
  188. g->short_start = 2 + (s->sample_rate_index != 8);
  189. } else {
  190. g->long_end = 0;
  191. g->short_start = 0;
  192. }
  193. } else {
  194. g->short_start = 13;
  195. g->long_end = 22;
  196. }
  197. }
  198. /* layer 1 unscaling */
  199. /* n = number of bits of the mantissa minus 1 */
  200. static inline int l1_unscale(int n, int mant, int scale_factor)
  201. {
  202. int shift, mod;
  203. int64_t val;
  204. shift = scale_factor_modshift[scale_factor];
  205. mod = shift & 3;
  206. shift >>= 2;
  207. val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]);
  208. shift += n;
  209. /* NOTE: at this point, 1 <= shift >= 21 + 15 */
  210. return (int)((val + (1LL << (shift - 1))) >> shift);
  211. }
  212. static inline int l2_unscale_group(int steps, int mant, int scale_factor)
  213. {
  214. int shift, mod, val;
  215. shift = scale_factor_modshift[scale_factor];
  216. mod = shift & 3;
  217. shift >>= 2;
  218. val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
  219. /* NOTE: at this point, 0 <= shift <= 21 */
  220. if (shift > 0)
  221. val = (val + (1 << (shift - 1))) >> shift;
  222. return val;
  223. }
  224. /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
  225. static inline int l3_unscale(int value, int exponent)
  226. {
  227. unsigned int m;
  228. int e;
  229. e = table_4_3_exp [4 * value + (exponent & 3)];
  230. m = table_4_3_value[4 * value + (exponent & 3)];
  231. e -= exponent >> 2;
  232. assert(e >= 1);
  233. if (e > 31)
  234. return 0;
  235. m = (m + (1 << (e - 1))) >> e;
  236. return m;
  237. }
  238. static void decode_init_static(AVCodec *codec)
  239. {
  240. int i, j, k;
  241. int offset;
  242. /* scale factors table for layer 1/2 */
  243. for (i = 0; i < 64; i++) {
  244. int shift, mod;
  245. /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
  246. shift = i / 3;
  247. mod = i % 3;
  248. scale_factor_modshift[i] = mod | (shift << 2);
  249. }
  250. /* scale factor multiply for layer 1 */
  251. for (i = 0; i < 15; i++) {
  252. int n, norm;
  253. n = i + 2;
  254. norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
  255. scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
  256. scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
  257. scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
  258. av_dlog(NULL, "%d: norm=%x s=%x %x %x\n", i, norm,
  259. scale_factor_mult[i][0],
  260. scale_factor_mult[i][1],
  261. scale_factor_mult[i][2]);
  262. }
  263. RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));
  264. /* huffman decode tables */
  265. offset = 0;
  266. for (i = 1; i < 16; i++) {
  267. const HuffTable *h = &mpa_huff_tables[i];
  268. int xsize, x, y;
  269. uint8_t tmp_bits [512];
  270. uint16_t tmp_codes[512];
  271. memset(tmp_bits , 0, sizeof(tmp_bits ));
  272. memset(tmp_codes, 0, sizeof(tmp_codes));
  273. xsize = h->xsize;
  274. j = 0;
  275. for (x = 0; x < xsize; x++) {
  276. for (y = 0; y < xsize; y++) {
  277. tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ];
  278. tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];
  279. }
  280. }
  281. /* XXX: fail test */
  282. huff_vlc[i].table = huff_vlc_tables+offset;
  283. huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
  284. init_vlc(&huff_vlc[i], 7, 512,
  285. tmp_bits, 1, 1, tmp_codes, 2, 2,
  286. INIT_VLC_USE_NEW_STATIC);
  287. offset += huff_vlc_tables_sizes[i];
  288. }
  289. assert(offset == FF_ARRAY_ELEMS(huff_vlc_tables));
  290. offset = 0;
  291. for (i = 0; i < 2; i++) {
  292. huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
  293. huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
  294. init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
  295. mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
  296. INIT_VLC_USE_NEW_STATIC);
  297. offset += huff_quad_vlc_tables_sizes[i];
  298. }
  299. assert(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));
  300. for (i = 0; i < 9; i++) {
  301. k = 0;
  302. for (j = 0; j < 22; j++) {
  303. band_index_long[i][j] = k;
  304. k += band_size_long[i][j];
  305. }
  306. band_index_long[i][22] = k;
  307. }
  308. /* compute n ^ (4/3) and store it in mantissa/exp format */
  309. mpegaudio_tableinit();
  310. for (i = 0; i < 4; i++) {
  311. if (ff_mpa_quant_bits[i] < 0) {
  312. for (j = 0; j < (1 << (-ff_mpa_quant_bits[i]+1)); j++) {
  313. int val1, val2, val3, steps;
  314. int val = j;
  315. steps = ff_mpa_quant_steps[i];
  316. val1 = val % steps;
  317. val /= steps;
  318. val2 = val % steps;
  319. val3 = val / steps;
  320. division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);
  321. }
  322. }
  323. }
  324. for (i = 0; i < 7; i++) {
  325. float f;
  326. INTFLOAT v;
  327. if (i != 6) {
  328. f = tan((double)i * M_PI / 12.0);
  329. v = FIXR(f / (1.0 + f));
  330. } else {
  331. v = FIXR(1.0);
  332. }
  333. is_table[0][ i] = v;
  334. is_table[1][6 - i] = v;
  335. }
  336. /* invalid values */
  337. for (i = 7; i < 16; i++)
  338. is_table[0][i] = is_table[1][i] = 0.0;
  339. for (i = 0; i < 16; i++) {
  340. double f;
  341. int e, k;
  342. for (j = 0; j < 2; j++) {
  343. e = -(j + 1) * ((i + 1) >> 1);
  344. f = pow(2.0, e / 4.0);
  345. k = i & 1;
  346. is_table_lsf[j][k ^ 1][i] = FIXR(f);
  347. is_table_lsf[j][k ][i] = FIXR(1.0);
  348. av_dlog(NULL, "is_table_lsf %d %d: %f %f\n",
  349. i, j, (float) is_table_lsf[j][0][i],
  350. (float) is_table_lsf[j][1][i]);
  351. }
  352. }
  353. for (i = 0; i < 8; i++) {
  354. float ci, cs, ca;
  355. ci = ci_table[i];
  356. cs = 1.0 / sqrt(1.0 + ci * ci);
  357. ca = cs * ci;
  358. #if !CONFIG_FLOAT
  359. csa_table[i][0] = FIXHR(cs/4);
  360. csa_table[i][1] = FIXHR(ca/4);
  361. csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);
  362. csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);
  363. #else
  364. csa_table[i][0] = cs;
  365. csa_table[i][1] = ca;
  366. csa_table[i][2] = ca + cs;
  367. csa_table[i][3] = ca - cs;
  368. #endif
  369. }
  370. /* compute mdct windows */
  371. for (i = 0; i < 36; i++) {
  372. for (j = 0; j < 4; j++) {
  373. double d;
  374. if (j == 2 && i % 3 != 1)
  375. continue;
  376. d = sin(M_PI * (i + 0.5) / 36.0);
  377. if (j == 1) {
  378. if (i >= 30) d = 0;
  379. else if (i >= 24) d = sin(M_PI * (i - 18 + 0.5) / 12.0);
  380. else if (i >= 18) d = 1;
  381. } else if (j == 3) {
  382. if (i < 6) d = 0;
  383. else if (i < 12) d = sin(M_PI * (i - 6 + 0.5) / 12.0);
  384. else if (i < 18) d = 1;
  385. }
  386. //merge last stage of imdct into the window coefficients
  387. d *= 0.5 / cos(M_PI * (2 * i + 19) / 72);
  388. if (j == 2)
  389. mdct_win[j][i/3] = FIXHR((d / (1<<5)));
  390. else
  391. mdct_win[j][i ] = FIXHR((d / (1<<5)));
  392. }
  393. }
  394. /* NOTE: we do frequency inversion adter the MDCT by changing
  395. the sign of the right window coefs */
  396. for (j = 0; j < 4; j++) {
  397. for (i = 0; i < 36; i += 2) {
  398. mdct_win[j + 4][i ] = mdct_win[j][i ];
  399. mdct_win[j + 4][i + 1] = -mdct_win[j][i + 1];
  400. }
  401. }
  402. }
  403. static av_cold int decode_init(AVCodecContext * avctx)
  404. {
  405. MPADecodeContext *s = avctx->priv_data;
  406. s->avctx = avctx;
  407. ff_mpadsp_init(&s->mpadsp);
  408. avctx->sample_fmt= OUT_FMT;
  409. s->err_recognition = avctx->err_recognition;
  410. if (avctx->codec_id == CODEC_ID_MP3ADU)
  411. s->adu_mode = 1;
  412. return 0;
  413. }
  414. #define C3 FIXHR(0.86602540378443864676/2)
  415. /* 0.5 / cos(pi*(2*i+1)/36) */
  416. static const INTFLOAT icos36[9] = {
  417. FIXR(0.50190991877167369479),
  418. FIXR(0.51763809020504152469), //0
  419. FIXR(0.55168895948124587824),
  420. FIXR(0.61038729438072803416),
  421. FIXR(0.70710678118654752439), //1
  422. FIXR(0.87172339781054900991),
  423. FIXR(1.18310079157624925896),
  424. FIXR(1.93185165257813657349), //2
  425. FIXR(5.73685662283492756461),
  426. };
  427. /* 0.5 / cos(pi*(2*i+1)/36) */
  428. static const INTFLOAT icos36h[9] = {
  429. FIXHR(0.50190991877167369479/2),
  430. FIXHR(0.51763809020504152469/2), //0
  431. FIXHR(0.55168895948124587824/2),
  432. FIXHR(0.61038729438072803416/2),
  433. FIXHR(0.70710678118654752439/2), //1
  434. FIXHR(0.87172339781054900991/2),
  435. FIXHR(1.18310079157624925896/4),
  436. FIXHR(1.93185165257813657349/4), //2
  437. // FIXHR(5.73685662283492756461),
  438. };
  439. /* 12 points IMDCT. We compute it "by hand" by factorizing obvious
  440. cases. */
  441. static void imdct12(INTFLOAT *out, INTFLOAT *in)
  442. {
  443. INTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
  444. in0 = in[0*3];
  445. in1 = in[1*3] + in[0*3];
  446. in2 = in[2*3] + in[1*3];
  447. in3 = in[3*3] + in[2*3];
  448. in4 = in[4*3] + in[3*3];
  449. in5 = in[5*3] + in[4*3];
  450. in5 += in3;
  451. in3 += in1;
  452. in2 = MULH3(in2, C3, 2);
  453. in3 = MULH3(in3, C3, 4);
  454. t1 = in0 - in4;
  455. t2 = MULH3(in1 - in5, icos36h[4], 2);
  456. out[ 7] =
  457. out[10] = t1 + t2;
  458. out[ 1] =
  459. out[ 4] = t1 - t2;
  460. in0 += SHR(in4, 1);
  461. in4 = in0 + in2;
  462. in5 += 2*in1;
  463. in1 = MULH3(in5 + in3, icos36h[1], 1);
  464. out[ 8] =
  465. out[ 9] = in4 + in1;
  466. out[ 2] =
  467. out[ 3] = in4 - in1;
  468. in0 -= in2;
  469. in5 = MULH3(in5 - in3, icos36h[7], 2);
  470. out[ 0] =
  471. out[ 5] = in0 - in5;
  472. out[ 6] =
  473. out[11] = in0 + in5;
  474. }
  475. /* cos(pi*i/18) */
  476. #define C1 FIXHR(0.98480775301220805936/2)
  477. #define C2 FIXHR(0.93969262078590838405/2)
  478. #define C3 FIXHR(0.86602540378443864676/2)
  479. #define C4 FIXHR(0.76604444311897803520/2)
  480. #define C5 FIXHR(0.64278760968653932632/2)
  481. #define C6 FIXHR(0.5/2)
  482. #define C7 FIXHR(0.34202014332566873304/2)
  483. #define C8 FIXHR(0.17364817766693034885/2)
  484. /* using Lee like decomposition followed by hand coded 9 points DCT */
  485. static void imdct36(INTFLOAT *out, INTFLOAT *buf, INTFLOAT *in, INTFLOAT *win)
  486. {
  487. int i, j;
  488. INTFLOAT t0, t1, t2, t3, s0, s1, s2, s3;
  489. INTFLOAT tmp[18], *tmp1, *in1;
  490. for (i = 17; i >= 1; i--)
  491. in[i] += in[i-1];
  492. for (i = 17; i >= 3; i -= 2)
  493. in[i] += in[i-2];
  494. for (j = 0; j < 2; j++) {
  495. tmp1 = tmp + j;
  496. in1 = in + j;
  497. t2 = in1[2*4] + in1[2*8] - in1[2*2];
  498. t3 = in1[2*0] + SHR(in1[2*6],1);
  499. t1 = in1[2*0] - in1[2*6];
  500. tmp1[ 6] = t1 - SHR(t2,1);
  501. tmp1[16] = t1 + t2;
  502. t0 = MULH3(in1[2*2] + in1[2*4] , C2, 2);
  503. t1 = MULH3(in1[2*4] - in1[2*8] , -2*C8, 1);
  504. t2 = MULH3(in1[2*2] + in1[2*8] , -C4, 2);
  505. tmp1[10] = t3 - t0 - t2;
  506. tmp1[ 2] = t3 + t0 + t1;
  507. tmp1[14] = t3 + t2 - t1;
  508. tmp1[ 4] = MULH3(in1[2*5] + in1[2*7] - in1[2*1], -C3, 2);
  509. t2 = MULH3(in1[2*1] + in1[2*5], C1, 2);
  510. t3 = MULH3(in1[2*5] - in1[2*7], -2*C7, 1);
  511. t0 = MULH3(in1[2*3], C3, 2);
  512. t1 = MULH3(in1[2*1] + in1[2*7], -C5, 2);
  513. tmp1[ 0] = t2 + t3 + t0;
  514. tmp1[12] = t2 + t1 - t0;
  515. tmp1[ 8] = t3 - t1 - t0;
  516. }
  517. i = 0;
  518. for (j = 0; j < 4; j++) {
  519. t0 = tmp[i];
  520. t1 = tmp[i + 2];
  521. s0 = t1 + t0;
  522. s2 = t1 - t0;
  523. t2 = tmp[i + 1];
  524. t3 = tmp[i + 3];
  525. s1 = MULH3(t3 + t2, icos36h[ j], 2);
  526. s3 = MULLx(t3 - t2, icos36 [8 - j], FRAC_BITS);
  527. t0 = s0 + s1;
  528. t1 = s0 - s1;
  529. out[(9 + j) * SBLIMIT] = MULH3(t1, win[ 9 + j], 1) + buf[9 + j];
  530. out[(8 - j) * SBLIMIT] = MULH3(t1, win[ 8 - j], 1) + buf[8 - j];
  531. buf[ 9 + j ] = MULH3(t0, win[18 + 9 + j], 1);
  532. buf[ 8 - j ] = MULH3(t0, win[18 + 8 - j], 1);
  533. t0 = s2 + s3;
  534. t1 = s2 - s3;
  535. out[(9 + 8 - j) * SBLIMIT] = MULH3(t1, win[ 9 + 8 - j], 1) + buf[9 + 8 - j];
  536. out[ j * SBLIMIT] = MULH3(t1, win[ j], 1) + buf[ j];
  537. buf[ 9 + 8 - j ] = MULH3(t0, win[18 + 9 + 8 - j], 1);
  538. buf[ j ] = MULH3(t0, win[18 + j], 1);
  539. i += 4;
  540. }
  541. s0 = tmp[16];
  542. s1 = MULH3(tmp[17], icos36h[4], 2);
  543. t0 = s0 + s1;
  544. t1 = s0 - s1;
  545. out[(9 + 4) * SBLIMIT] = MULH3(t1, win[ 9 + 4], 1) + buf[9 + 4];
  546. out[(8 - 4) * SBLIMIT] = MULH3(t1, win[ 8 - 4], 1) + buf[8 - 4];
  547. buf[ 9 + 4 ] = MULH3(t0, win[18 + 9 + 4], 1);
  548. buf[ 8 - 4 ] = MULH3(t0, win[18 + 8 - 4], 1);
  549. }
  550. /* return the number of decoded frames */
  551. static int mp_decode_layer1(MPADecodeContext *s)
  552. {
  553. int bound, i, v, n, ch, j, mant;
  554. uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
  555. uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
  556. if (s->mode == MPA_JSTEREO)
  557. bound = (s->mode_ext + 1) * 4;
  558. else
  559. bound = SBLIMIT;
  560. /* allocation bits */
  561. for (i = 0; i < bound; i++) {
  562. for (ch = 0; ch < s->nb_channels; ch++) {
  563. allocation[ch][i] = get_bits(&s->gb, 4);
  564. }
  565. }
  566. for (i = bound; i < SBLIMIT; i++)
  567. allocation[0][i] = get_bits(&s->gb, 4);
  568. /* scale factors */
  569. for (i = 0; i < bound; i++) {
  570. for (ch = 0; ch < s->nb_channels; ch++) {
  571. if (allocation[ch][i])
  572. scale_factors[ch][i] = get_bits(&s->gb, 6);
  573. }
  574. }
  575. for (i = bound; i < SBLIMIT; i++) {
  576. if (allocation[0][i]) {
  577. scale_factors[0][i] = get_bits(&s->gb, 6);
  578. scale_factors[1][i] = get_bits(&s->gb, 6);
  579. }
  580. }
  581. /* compute samples */
  582. for (j = 0; j < 12; j++) {
  583. for (i = 0; i < bound; i++) {
  584. for (ch = 0; ch < s->nb_channels; ch++) {
  585. n = allocation[ch][i];
  586. if (n) {
  587. mant = get_bits(&s->gb, n + 1);
  588. v = l1_unscale(n, mant, scale_factors[ch][i]);
  589. } else {
  590. v = 0;
  591. }
  592. s->sb_samples[ch][j][i] = v;
  593. }
  594. }
  595. for (i = bound; i < SBLIMIT; i++) {
  596. n = allocation[0][i];
  597. if (n) {
  598. mant = get_bits(&s->gb, n + 1);
  599. v = l1_unscale(n, mant, scale_factors[0][i]);
  600. s->sb_samples[0][j][i] = v;
  601. v = l1_unscale(n, mant, scale_factors[1][i]);
  602. s->sb_samples[1][j][i] = v;
  603. } else {
  604. s->sb_samples[0][j][i] = 0;
  605. s->sb_samples[1][j][i] = 0;
  606. }
  607. }
  608. }
  609. return 12;
  610. }
  611. static int mp_decode_layer2(MPADecodeContext *s)
  612. {
  613. int sblimit; /* number of used subbands */
  614. const unsigned char *alloc_table;
  615. int table, bit_alloc_bits, i, j, ch, bound, v;
  616. unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
  617. unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
  618. unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
  619. int scale, qindex, bits, steps, k, l, m, b;
  620. /* select decoding table */
  621. table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
  622. s->sample_rate, s->lsf);
  623. sblimit = ff_mpa_sblimit_table[table];
  624. alloc_table = ff_mpa_alloc_tables[table];
  625. if (s->mode == MPA_JSTEREO)
  626. bound = (s->mode_ext + 1) * 4;
  627. else
  628. bound = sblimit;
  629. av_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
  630. /* sanity check */
  631. if (bound > sblimit)
  632. bound = sblimit;
  633. /* parse bit allocation */
  634. j = 0;
  635. for (i = 0; i < bound; i++) {
  636. bit_alloc_bits = alloc_table[j];
  637. for (ch = 0; ch < s->nb_channels; ch++)
  638. bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
  639. j += 1 << bit_alloc_bits;
  640. }
  641. for (i = bound; i < sblimit; i++) {
  642. bit_alloc_bits = alloc_table[j];
  643. v = get_bits(&s->gb, bit_alloc_bits);
  644. bit_alloc[0][i] = v;
  645. bit_alloc[1][i] = v;
  646. j += 1 << bit_alloc_bits;
  647. }
  648. /* scale codes */
  649. for (i = 0; i < sblimit; i++) {
  650. for (ch = 0; ch < s->nb_channels; ch++) {
  651. if (bit_alloc[ch][i])
  652. scale_code[ch][i] = get_bits(&s->gb, 2);
  653. }
  654. }
  655. /* scale factors */
  656. for (i = 0; i < sblimit; i++) {
  657. for (ch = 0; ch < s->nb_channels; ch++) {
  658. if (bit_alloc[ch][i]) {
  659. sf = scale_factors[ch][i];
  660. switch (scale_code[ch][i]) {
  661. default:
  662. case 0:
  663. sf[0] = get_bits(&s->gb, 6);
  664. sf[1] = get_bits(&s->gb, 6);
  665. sf[2] = get_bits(&s->gb, 6);
  666. break;
  667. case 2:
  668. sf[0] = get_bits(&s->gb, 6);
  669. sf[1] = sf[0];
  670. sf[2] = sf[0];
  671. break;
  672. case 1:
  673. sf[0] = get_bits(&s->gb, 6);
  674. sf[2] = get_bits(&s->gb, 6);
  675. sf[1] = sf[0];
  676. break;
  677. case 3:
  678. sf[0] = get_bits(&s->gb, 6);
  679. sf[2] = get_bits(&s->gb, 6);
  680. sf[1] = sf[2];
  681. break;
  682. }
  683. }
  684. }
  685. }
  686. /* samples */
  687. for (k = 0; k < 3; k++) {
  688. for (l = 0; l < 12; l += 3) {
  689. j = 0;
  690. for (i = 0; i < bound; i++) {
  691. bit_alloc_bits = alloc_table[j];
  692. for (ch = 0; ch < s->nb_channels; ch++) {
  693. b = bit_alloc[ch][i];
  694. if (b) {
  695. scale = scale_factors[ch][i][k];
  696. qindex = alloc_table[j+b];
  697. bits = ff_mpa_quant_bits[qindex];
  698. if (bits < 0) {
  699. int v2;
  700. /* 3 values at the same time */
  701. v = get_bits(&s->gb, -bits);
  702. v2 = division_tabs[qindex][v];
  703. steps = ff_mpa_quant_steps[qindex];
  704. s->sb_samples[ch][k * 12 + l + 0][i] =
  705. l2_unscale_group(steps, v2 & 15, scale);
  706. s->sb_samples[ch][k * 12 + l + 1][i] =
  707. l2_unscale_group(steps, (v2 >> 4) & 15, scale);
  708. s->sb_samples[ch][k * 12 + l + 2][i] =
  709. l2_unscale_group(steps, v2 >> 8 , scale);
  710. } else {
  711. for (m = 0; m < 3; m++) {
  712. v = get_bits(&s->gb, bits);
  713. v = l1_unscale(bits - 1, v, scale);
  714. s->sb_samples[ch][k * 12 + l + m][i] = v;
  715. }
  716. }
  717. } else {
  718. s->sb_samples[ch][k * 12 + l + 0][i] = 0;
  719. s->sb_samples[ch][k * 12 + l + 1][i] = 0;
  720. s->sb_samples[ch][k * 12 + l + 2][i] = 0;
  721. }
  722. }
  723. /* next subband in alloc table */
  724. j += 1 << bit_alloc_bits;
  725. }
  726. /* XXX: find a way to avoid this duplication of code */
  727. for (i = bound; i < sblimit; i++) {
  728. bit_alloc_bits = alloc_table[j];
  729. b = bit_alloc[0][i];
  730. if (b) {
  731. int mant, scale0, scale1;
  732. scale0 = scale_factors[0][i][k];
  733. scale1 = scale_factors[1][i][k];
  734. qindex = alloc_table[j+b];
  735. bits = ff_mpa_quant_bits[qindex];
  736. if (bits < 0) {
  737. /* 3 values at the same time */
  738. v = get_bits(&s->gb, -bits);
  739. steps = ff_mpa_quant_steps[qindex];
  740. mant = v % steps;
  741. v = v / steps;
  742. s->sb_samples[0][k * 12 + l + 0][i] =
  743. l2_unscale_group(steps, mant, scale0);
  744. s->sb_samples[1][k * 12 + l + 0][i] =
  745. l2_unscale_group(steps, mant, scale1);
  746. mant = v % steps;
  747. v = v / steps;
  748. s->sb_samples[0][k * 12 + l + 1][i] =
  749. l2_unscale_group(steps, mant, scale0);
  750. s->sb_samples[1][k * 12 + l + 1][i] =
  751. l2_unscale_group(steps, mant, scale1);
  752. s->sb_samples[0][k * 12 + l + 2][i] =
  753. l2_unscale_group(steps, v, scale0);
  754. s->sb_samples[1][k * 12 + l + 2][i] =
  755. l2_unscale_group(steps, v, scale1);
  756. } else {
  757. for (m = 0; m < 3; m++) {
  758. mant = get_bits(&s->gb, bits);
  759. s->sb_samples[0][k * 12 + l + m][i] =
  760. l1_unscale(bits - 1, mant, scale0);
  761. s->sb_samples[1][k * 12 + l + m][i] =
  762. l1_unscale(bits - 1, mant, scale1);
  763. }
  764. }
  765. } else {
  766. s->sb_samples[0][k * 12 + l + 0][i] = 0;
  767. s->sb_samples[0][k * 12 + l + 1][i] = 0;
  768. s->sb_samples[0][k * 12 + l + 2][i] = 0;
  769. s->sb_samples[1][k * 12 + l + 0][i] = 0;
  770. s->sb_samples[1][k * 12 + l + 1][i] = 0;
  771. s->sb_samples[1][k * 12 + l + 2][i] = 0;
  772. }
  773. /* next subband in alloc table */
  774. j += 1 << bit_alloc_bits;
  775. }
  776. /* fill remaining samples to zero */
  777. for (i = sblimit; i < SBLIMIT; i++) {
  778. for (ch = 0; ch < s->nb_channels; ch++) {
  779. s->sb_samples[ch][k * 12 + l + 0][i] = 0;
  780. s->sb_samples[ch][k * 12 + l + 1][i] = 0;
  781. s->sb_samples[ch][k * 12 + l + 2][i] = 0;
  782. }
  783. }
  784. }
  785. }
  786. return 3 * 12;
  787. }
  788. #define SPLIT(dst,sf,n) \
  789. if (n == 3) { \
  790. int m = (sf * 171) >> 9; \
  791. dst = sf - 3 * m; \
  792. sf = m; \
  793. } else if (n == 4) { \
  794. dst = sf & 3; \
  795. sf >>= 2; \
  796. } else if (n == 5) { \
  797. int m = (sf * 205) >> 10; \
  798. dst = sf - 5 * m; \
  799. sf = m; \
  800. } else if (n == 6) { \
  801. int m = (sf * 171) >> 10; \
  802. dst = sf - 6 * m; \
  803. sf = m; \
  804. } else { \
  805. dst = 0; \
  806. }
  807. static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2,
  808. int n3)
  809. {
  810. SPLIT(slen[3], sf, n3)
  811. SPLIT(slen[2], sf, n2)
  812. SPLIT(slen[1], sf, n1)
  813. slen[0] = sf;
  814. }
  815. static void exponents_from_scale_factors(MPADecodeContext *s, GranuleDef *g,
  816. int16_t *exponents)
  817. {
  818. const uint8_t *bstab, *pretab;
  819. int len, i, j, k, l, v0, shift, gain, gains[3];
  820. int16_t *exp_ptr;
  821. exp_ptr = exponents;
  822. gain = g->global_gain - 210;
  823. shift = g->scalefac_scale + 1;
  824. bstab = band_size_long[s->sample_rate_index];
  825. pretab = mpa_pretab[g->preflag];
  826. for (i = 0; i < g->long_end; i++) {
  827. v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
  828. len = bstab[i];
  829. for (j = len; j > 0; j--)
  830. *exp_ptr++ = v0;
  831. }
  832. if (g->short_start < 13) {
  833. bstab = band_size_short[s->sample_rate_index];
  834. gains[0] = gain - (g->subblock_gain[0] << 3);
  835. gains[1] = gain - (g->subblock_gain[1] << 3);
  836. gains[2] = gain - (g->subblock_gain[2] << 3);
  837. k = g->long_end;
  838. for (i = g->short_start; i < 13; i++) {
  839. len = bstab[i];
  840. for (l = 0; l < 3; l++) {
  841. v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
  842. for (j = len; j > 0; j--)
  843. *exp_ptr++ = v0;
  844. }
  845. }
  846. }
  847. }
  848. /* handle n = 0 too */
  849. static inline int get_bitsz(GetBitContext *s, int n)
  850. {
  851. return n ? get_bits(s, n) : 0;
  852. }
  853. static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos,
  854. int *end_pos2)
  855. {
  856. if (s->in_gb.buffer && *pos >= s->gb.size_in_bits) {
  857. s->gb = s->in_gb;
  858. s->in_gb.buffer = NULL;
  859. assert((get_bits_count(&s->gb) & 7) == 0);
  860. skip_bits_long(&s->gb, *pos - *end_pos);
  861. *end_pos2 =
  862. *end_pos = *end_pos2 + get_bits_count(&s->gb) - *pos;
  863. *pos = get_bits_count(&s->gb);
  864. }
  865. }
  866. /* Following is a optimized code for
  867. INTFLOAT v = *src
  868. if(get_bits1(&s->gb))
  869. v = -v;
  870. *dst = v;
  871. */
  872. #if CONFIG_FLOAT
  873. #define READ_FLIP_SIGN(dst,src) \
  874. v = AV_RN32A(src) ^ (get_bits1(&s->gb) << 31); \
  875. AV_WN32A(dst, v);
  876. #else
  877. #define READ_FLIP_SIGN(dst,src) \
  878. v = -get_bits1(&s->gb); \
  879. *(dst) = (*(src) ^ v) - v;
  880. #endif
  881. static int huffman_decode(MPADecodeContext *s, GranuleDef *g,
  882. int16_t *exponents, int end_pos2)
  883. {
  884. int s_index;
  885. int i;
  886. int last_pos, bits_left;
  887. VLC *vlc;
  888. int end_pos = FFMIN(end_pos2, s->gb.size_in_bits);
  889. /* low frequencies (called big values) */
  890. s_index = 0;
  891. for (i = 0; i < 3; i++) {
  892. int j, k, l, linbits;
  893. j = g->region_size[i];
  894. if (j == 0)
  895. continue;
  896. /* select vlc table */
  897. k = g->table_select[i];
  898. l = mpa_huff_data[k][0];
  899. linbits = mpa_huff_data[k][1];
  900. vlc = &huff_vlc[l];
  901. if (!l) {
  902. memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * 2 * j);
  903. s_index += 2 * j;
  904. continue;
  905. }
  906. /* read huffcode and compute each couple */
  907. for (; j > 0; j--) {
  908. int exponent, x, y;
  909. int v;
  910. int pos = get_bits_count(&s->gb);
  911. if (pos >= end_pos){
  912. // av_log(NULL, AV_LOG_ERROR, "pos: %d %d %d %d\n", pos, end_pos, end_pos2, s_index);
  913. switch_buffer(s, &pos, &end_pos, &end_pos2);
  914. // av_log(NULL, AV_LOG_ERROR, "new pos: %d %d\n", pos, end_pos);
  915. if (pos >= end_pos)
  916. break;
  917. }
  918. y = get_vlc2(&s->gb, vlc->table, 7, 3);
  919. if (!y) {
  920. g->sb_hybrid[s_index ] =
  921. g->sb_hybrid[s_index+1] = 0;
  922. s_index += 2;
  923. continue;
  924. }
  925. exponent= exponents[s_index];
  926. av_dlog(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n",
  927. i, g->region_size[i] - j, x, y, exponent);
  928. if (y & 16) {
  929. x = y >> 5;
  930. y = y & 0x0f;
  931. if (x < 15) {
  932. READ_FLIP_SIGN(g->sb_hybrid + s_index, RENAME(expval_table)[exponent] + x)
  933. } else {
  934. x += get_bitsz(&s->gb, linbits);
  935. v = l3_unscale(x, exponent);
  936. if (get_bits1(&s->gb))
  937. v = -v;
  938. g->sb_hybrid[s_index] = v;
  939. }
  940. if (y < 15) {
  941. READ_FLIP_SIGN(g->sb_hybrid + s_index + 1, RENAME(expval_table)[exponent] + y)
  942. } else {
  943. y += get_bitsz(&s->gb, linbits);
  944. v = l3_unscale(y, exponent);
  945. if (get_bits1(&s->gb))
  946. v = -v;
  947. g->sb_hybrid[s_index+1] = v;
  948. }
  949. } else {
  950. x = y >> 5;
  951. y = y & 0x0f;
  952. x += y;
  953. if (x < 15) {
  954. READ_FLIP_SIGN(g->sb_hybrid + s_index + !!y, RENAME(expval_table)[exponent] + x)
  955. } else {
  956. x += get_bitsz(&s->gb, linbits);
  957. v = l3_unscale(x, exponent);
  958. if (get_bits1(&s->gb))
  959. v = -v;
  960. g->sb_hybrid[s_index+!!y] = v;
  961. }
  962. g->sb_hybrid[s_index + !y] = 0;
  963. }
  964. s_index += 2;
  965. }
  966. }
  967. /* high frequencies */
  968. vlc = &huff_quad_vlc[g->count1table_select];
  969. last_pos = 0;
  970. while (s_index <= 572) {
  971. int pos, code;
  972. pos = get_bits_count(&s->gb);
  973. if (pos >= end_pos) {
  974. if (pos > end_pos2 && last_pos) {
  975. /* some encoders generate an incorrect size for this
  976. part. We must go back into the data */
  977. s_index -= 4;
  978. skip_bits_long(&s->gb, last_pos - pos);
  979. av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
  980. if(s->err_recognition & AV_EF_BITSTREAM)
  981. s_index=0;
  982. break;
  983. }
  984. // av_log(NULL, AV_LOG_ERROR, "pos2: %d %d %d %d\n", pos, end_pos, end_pos2, s_index);
  985. switch_buffer(s, &pos, &end_pos, &end_pos2);
  986. // av_log(NULL, AV_LOG_ERROR, "new pos2: %d %d %d\n", pos, end_pos, s_index);
  987. if (pos >= end_pos)
  988. break;
  989. }
  990. last_pos = pos;
  991. code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
  992. av_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
  993. g->sb_hybrid[s_index+0] =
  994. g->sb_hybrid[s_index+1] =
  995. g->sb_hybrid[s_index+2] =
  996. g->sb_hybrid[s_index+3] = 0;
  997. while (code) {
  998. static const int idxtab[16] = { 3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0 };
  999. int v;
  1000. int pos = s_index + idxtab[code];
  1001. code ^= 8 >> idxtab[code];
  1002. READ_FLIP_SIGN(g->sb_hybrid + pos, RENAME(exp_table)+exponents[pos])
  1003. }
  1004. s_index += 4;
  1005. }
  1006. /* skip extension bits */
  1007. bits_left = end_pos2 - get_bits_count(&s->gb);
  1008. //av_log(NULL, AV_LOG_ERROR, "left:%d buf:%p\n", bits_left, s->in_gb.buffer);
  1009. if (bits_left < 0 && (s->err_recognition & AV_EF_BITSTREAM)) {
  1010. av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
  1011. s_index=0;
  1012. } else if (bits_left > 0 && (s->err_recognition & AV_EF_BUFFER)) {
  1013. av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
  1014. s_index = 0;
  1015. }
  1016. memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * (576 - s_index));
  1017. skip_bits_long(&s->gb, bits_left);
  1018. i = get_bits_count(&s->gb);
  1019. switch_buffer(s, &i, &end_pos, &end_pos2);
  1020. return 0;
  1021. }
  1022. /* Reorder short blocks from bitstream order to interleaved order. It
  1023. would be faster to do it in parsing, but the code would be far more
  1024. complicated */
  1025. static void reorder_block(MPADecodeContext *s, GranuleDef *g)
  1026. {
  1027. int i, j, len;
  1028. INTFLOAT *ptr, *dst, *ptr1;
  1029. INTFLOAT tmp[576];
  1030. if (g->block_type != 2)
  1031. return;
  1032. if (g->switch_point) {
  1033. if (s->sample_rate_index != 8)
  1034. ptr = g->sb_hybrid + 36;
  1035. else
  1036. ptr = g->sb_hybrid + 48;
  1037. } else {
  1038. ptr = g->sb_hybrid;
  1039. }
  1040. for (i = g->short_start; i < 13; i++) {
  1041. len = band_size_short[s->sample_rate_index][i];
  1042. ptr1 = ptr;
  1043. dst = tmp;
  1044. for (j = len; j > 0; j--) {
  1045. *dst++ = ptr[0*len];
  1046. *dst++ = ptr[1*len];
  1047. *dst++ = ptr[2*len];
  1048. ptr++;
  1049. }
  1050. ptr += 2 * len;
  1051. memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
  1052. }
  1053. }
  1054. #define ISQRT2 FIXR(0.70710678118654752440)
  1055. static void compute_stereo(MPADecodeContext *s, GranuleDef *g0, GranuleDef *g1)
  1056. {
  1057. int i, j, k, l;
  1058. int sf_max, sf, len, non_zero_found;
  1059. INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;
  1060. int non_zero_found_short[3];
  1061. /* intensity stereo */
  1062. if (s->mode_ext & MODE_EXT_I_STEREO) {
  1063. if (!s->lsf) {
  1064. is_tab = is_table;
  1065. sf_max = 7;
  1066. } else {
  1067. is_tab = is_table_lsf[g1->scalefac_compress & 1];
  1068. sf_max = 16;
  1069. }
  1070. tab0 = g0->sb_hybrid + 576;
  1071. tab1 = g1->sb_hybrid + 576;
  1072. non_zero_found_short[0] = 0;
  1073. non_zero_found_short[1] = 0;
  1074. non_zero_found_short[2] = 0;
  1075. k = (13 - g1->short_start) * 3 + g1->long_end - 3;
  1076. for (i = 12; i >= g1->short_start; i--) {
  1077. /* for last band, use previous scale factor */
  1078. if (i != 11)
  1079. k -= 3;
  1080. len = band_size_short[s->sample_rate_index][i];
  1081. for (l = 2; l >= 0; l--) {
  1082. tab0 -= len;
  1083. tab1 -= len;
  1084. if (!non_zero_found_short[l]) {
  1085. /* test if non zero band. if so, stop doing i-stereo */
  1086. for (j = 0; j < len; j++) {
  1087. if (tab1[j] != 0) {
  1088. non_zero_found_short[l] = 1;
  1089. goto found1;
  1090. }
  1091. }
  1092. sf = g1->scale_factors[k + l];
  1093. if (sf >= sf_max)
  1094. goto found1;
  1095. v1 = is_tab[0][sf];
  1096. v2 = is_tab[1][sf];
  1097. for (j = 0; j < len; j++) {
  1098. tmp0 = tab0[j];
  1099. tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
  1100. tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
  1101. }
  1102. } else {
  1103. found1:
  1104. if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1105. /* lower part of the spectrum : do ms stereo
  1106. if enabled */
  1107. for (j = 0; j < len; j++) {
  1108. tmp0 = tab0[j];
  1109. tmp1 = tab1[j];
  1110. tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
  1111. tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
  1112. }
  1113. }
  1114. }
  1115. }
  1116. }
  1117. non_zero_found = non_zero_found_short[0] |
  1118. non_zero_found_short[1] |
  1119. non_zero_found_short[2];
  1120. for (i = g1->long_end - 1;i >= 0;i--) {
  1121. len = band_size_long[s->sample_rate_index][i];
  1122. tab0 -= len;
  1123. tab1 -= len;
  1124. /* test if non zero band. if so, stop doing i-stereo */
  1125. if (!non_zero_found) {
  1126. for (j = 0; j < len; j++) {
  1127. if (tab1[j] != 0) {
  1128. non_zero_found = 1;
  1129. goto found2;
  1130. }
  1131. }
  1132. /* for last band, use previous scale factor */
  1133. k = (i == 21) ? 20 : i;
  1134. sf = g1->scale_factors[k];
  1135. if (sf >= sf_max)
  1136. goto found2;
  1137. v1 = is_tab[0][sf];
  1138. v2 = is_tab[1][sf];
  1139. for (j = 0; j < len; j++) {
  1140. tmp0 = tab0[j];
  1141. tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
  1142. tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
  1143. }
  1144. } else {
  1145. found2:
  1146. if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1147. /* lower part of the spectrum : do ms stereo
  1148. if enabled */
  1149. for (j = 0; j < len; j++) {
  1150. tmp0 = tab0[j];
  1151. tmp1 = tab1[j];
  1152. tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
  1153. tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
  1154. }
  1155. }
  1156. }
  1157. }
  1158. } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1159. /* ms stereo ONLY */
  1160. /* NOTE: the 1/sqrt(2) normalization factor is included in the
  1161. global gain */
  1162. tab0 = g0->sb_hybrid;
  1163. tab1 = g1->sb_hybrid;
  1164. for (i = 0; i < 576; i++) {
  1165. tmp0 = tab0[i];
  1166. tmp1 = tab1[i];
  1167. tab0[i] = tmp0 + tmp1;
  1168. tab1[i] = tmp0 - tmp1;
  1169. }
  1170. }
  1171. }
  1172. #if CONFIG_FLOAT
  1173. #define AA(j) do { \
  1174. float tmp0 = ptr[-1-j]; \
  1175. float tmp1 = ptr[ j]; \
  1176. ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1]; \
  1177. ptr[ j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0]; \
  1178. } while (0)
  1179. #else
  1180. #define AA(j) do { \
  1181. int tmp0 = ptr[-1-j]; \
  1182. int tmp1 = ptr[ j]; \
  1183. int tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]); \
  1184. ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2])); \
  1185. ptr[ j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3])); \
  1186. } while (0)
  1187. #endif
  1188. static void compute_antialias(MPADecodeContext *s, GranuleDef *g)
  1189. {
  1190. INTFLOAT *ptr;
  1191. int n, i;
  1192. /* we antialias only "long" bands */
  1193. if (g->block_type == 2) {
  1194. if (!g->switch_point)
  1195. return;
  1196. /* XXX: check this for 8000Hz case */
  1197. n = 1;
  1198. } else {
  1199. n = SBLIMIT - 1;
  1200. }
  1201. ptr = g->sb_hybrid + 18;
  1202. for (i = n; i > 0; i--) {
  1203. AA(0);
  1204. AA(1);
  1205. AA(2);
  1206. AA(3);
  1207. AA(4);
  1208. AA(5);
  1209. AA(6);
  1210. AA(7);
  1211. ptr += 18;
  1212. }
  1213. }
  1214. static void compute_imdct(MPADecodeContext *s, GranuleDef *g,
  1215. INTFLOAT *sb_samples, INTFLOAT *mdct_buf)
  1216. {
  1217. INTFLOAT *win, *win1, *out_ptr, *ptr, *buf, *ptr1;
  1218. INTFLOAT out2[12];
  1219. int i, j, mdct_long_end, sblimit;
  1220. /* find last non zero block */
  1221. ptr = g->sb_hybrid + 576;
  1222. ptr1 = g->sb_hybrid + 2 * 18;
  1223. while (ptr >= ptr1) {
  1224. int32_t *p;
  1225. ptr -= 6;
  1226. p = (int32_t*)ptr;
  1227. if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
  1228. break;
  1229. }
  1230. sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
  1231. if (g->block_type == 2) {
  1232. /* XXX: check for 8000 Hz */
  1233. if (g->switch_point)
  1234. mdct_long_end = 2;
  1235. else
  1236. mdct_long_end = 0;
  1237. } else {
  1238. mdct_long_end = sblimit;
  1239. }
  1240. buf = mdct_buf;
  1241. ptr = g->sb_hybrid;
  1242. for (j = 0; j < mdct_long_end; j++) {
  1243. /* apply window & overlap with previous buffer */
  1244. out_ptr = sb_samples + j;
  1245. /* select window */
  1246. if (g->switch_point && j < 2)
  1247. win1 = mdct_win[0];
  1248. else
  1249. win1 = mdct_win[g->block_type];
  1250. /* select frequency inversion */
  1251. win = win1 + ((4 * 36) & -(j & 1));
  1252. imdct36(out_ptr, buf, ptr, win);
  1253. out_ptr += 18 * SBLIMIT;
  1254. ptr += 18;
  1255. buf += 18;
  1256. }
  1257. for (j = mdct_long_end; j < sblimit; j++) {
  1258. /* select frequency inversion */
  1259. win = mdct_win[2] + ((4 * 36) & -(j & 1));
  1260. out_ptr = sb_samples + j;
  1261. for (i = 0; i < 6; i++) {
  1262. *out_ptr = buf[i];
  1263. out_ptr += SBLIMIT;
  1264. }
  1265. imdct12(out2, ptr + 0);
  1266. for (i = 0; i < 6; i++) {
  1267. *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[i + 6*1];
  1268. buf[i + 6*2] = MULH3(out2[i + 6], win[i + 6], 1);
  1269. out_ptr += SBLIMIT;
  1270. }
  1271. imdct12(out2, ptr + 1);
  1272. for (i = 0; i < 6; i++) {
  1273. *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[i + 6*2];
  1274. buf[i + 6*0] = MULH3(out2[i + 6], win[i + 6], 1);
  1275. out_ptr += SBLIMIT;
  1276. }
  1277. imdct12(out2, ptr + 2);
  1278. for (i = 0; i < 6; i++) {
  1279. buf[i + 6*0] = MULH3(out2[i ], win[i ], 1) + buf[i + 6*0];
  1280. buf[i + 6*1] = MULH3(out2[i + 6], win[i + 6], 1);
  1281. buf[i + 6*2] = 0;
  1282. }
  1283. ptr += 18;
  1284. buf += 18;
  1285. }
  1286. /* zero bands */
  1287. for (j = sblimit; j < SBLIMIT; j++) {
  1288. /* overlap */
  1289. out_ptr = sb_samples + j;
  1290. for (i = 0; i < 18; i++) {
  1291. *out_ptr = buf[i];
  1292. buf[i] = 0;
  1293. out_ptr += SBLIMIT;
  1294. }
  1295. buf += 18;
  1296. }
  1297. }
  1298. /* main layer3 decoding function */
  1299. static int mp_decode_layer3(MPADecodeContext *s)
  1300. {
  1301. int nb_granules, main_data_begin;
  1302. int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
  1303. GranuleDef *g;
  1304. int16_t exponents[576]; //FIXME try INTFLOAT
  1305. /* read side info */
  1306. if (s->lsf) {
  1307. main_data_begin = get_bits(&s->gb, 8);
  1308. skip_bits(&s->gb, s->nb_channels);
  1309. nb_granules = 1;
  1310. } else {
  1311. main_data_begin = get_bits(&s->gb, 9);
  1312. if (s->nb_channels == 2)
  1313. skip_bits(&s->gb, 3);
  1314. else
  1315. skip_bits(&s->gb, 5);
  1316. nb_granules = 2;
  1317. for (ch = 0; ch < s->nb_channels; ch++) {
  1318. s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
  1319. s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
  1320. }
  1321. }
  1322. for (gr = 0; gr < nb_granules; gr++) {
  1323. for (ch = 0; ch < s->nb_channels; ch++) {
  1324. av_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
  1325. g = &s->granules[ch][gr];
  1326. g->part2_3_length = get_bits(&s->gb, 12);
  1327. g->big_values = get_bits(&s->gb, 9);
  1328. if (g->big_values > 288) {
  1329. av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
  1330. return AVERROR_INVALIDDATA;
  1331. }
  1332. g->global_gain = get_bits(&s->gb, 8);
  1333. /* if MS stereo only is selected, we precompute the
  1334. 1/sqrt(2) renormalization factor */
  1335. if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
  1336. MODE_EXT_MS_STEREO)
  1337. g->global_gain -= 2;
  1338. if (s->lsf)
  1339. g->scalefac_compress = get_bits(&s->gb, 9);
  1340. else
  1341. g->scalefac_compress = get_bits(&s->gb, 4);
  1342. blocksplit_flag = get_bits1(&s->gb);
  1343. if (blocksplit_flag) {
  1344. g->block_type = get_bits(&s->gb, 2);
  1345. if (g->block_type == 0) {
  1346. av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
  1347. return AVERROR_INVALIDDATA;
  1348. }
  1349. g->switch_point = get_bits1(&s->gb);
  1350. for (i = 0; i < 2; i++)
  1351. g->table_select[i] = get_bits(&s->gb, 5);
  1352. for (i = 0; i < 3; i++)
  1353. g->subblock_gain[i] = get_bits(&s->gb, 3);
  1354. ff_init_short_region(s, g);
  1355. } else {
  1356. int region_address1, region_address2;
  1357. g->block_type = 0;
  1358. g->switch_point = 0;
  1359. for (i = 0; i < 3; i++)
  1360. g->table_select[i] = get_bits(&s->gb, 5);
  1361. /* compute huffman coded region sizes */
  1362. region_address1 = get_bits(&s->gb, 4);
  1363. region_address2 = get_bits(&s->gb, 3);
  1364. av_dlog(s->avctx, "region1=%d region2=%d\n",
  1365. region_address1, region_address2);
  1366. ff_init_long_region(s, g, region_address1, region_address2);
  1367. }
  1368. ff_region_offset2size(g);
  1369. ff_compute_band_indexes(s, g);
  1370. g->preflag = 0;
  1371. if (!s->lsf)
  1372. g->preflag = get_bits1(&s->gb);
  1373. g->scalefac_scale = get_bits1(&s->gb);
  1374. g->count1table_select = get_bits1(&s->gb);
  1375. av_dlog(s->avctx, "block_type=%d switch_point=%d\n",
  1376. g->block_type, g->switch_point);
  1377. }
  1378. }
  1379. if (!s->adu_mode) {
  1380. const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
  1381. assert((get_bits_count(&s->gb) & 7) == 0);
  1382. /* now we get bits from the main_data_begin offset */
  1383. av_dlog(s->avctx, "seekback: %d\n", main_data_begin);
  1384. //av_log(NULL, AV_LOG_ERROR, "backstep:%d, lastbuf:%d\n", main_data_begin, s->last_buf_size);
  1385. memcpy(s->last_buf + s->last_buf_size, ptr, EXTRABYTES);
  1386. s->in_gb = s->gb;
  1387. init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);
  1388. skip_bits_long(&s->gb, 8*(s->last_buf_size - main_data_begin));
  1389. }
  1390. for (gr = 0; gr < nb_granules; gr++) {
  1391. for (ch = 0; ch < s->nb_channels; ch++) {
  1392. g = &s->granules[ch][gr];
  1393. if (get_bits_count(&s->gb) < 0) {
  1394. av_log(s->avctx, AV_LOG_DEBUG, "mdb:%d, lastbuf:%d skipping granule %d\n",
  1395. main_data_begin, s->last_buf_size, gr);
  1396. skip_bits_long(&s->gb, g->part2_3_length);
  1397. memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
  1398. if (get_bits_count(&s->gb) >= s->gb.size_in_bits && s->in_gb.buffer) {
  1399. skip_bits_long(&s->in_gb, get_bits_count(&s->gb) - s->gb.size_in_bits);
  1400. s->gb = s->in_gb;
  1401. s->in_gb.buffer = NULL;
  1402. }
  1403. continue;
  1404. }
  1405. bits_pos = get_bits_count(&s->gb);
  1406. if (!s->lsf) {
  1407. uint8_t *sc;
  1408. int slen, slen1, slen2;
  1409. /* MPEG1 scale factors */
  1410. slen1 = slen_table[0][g->scalefac_compress];
  1411. slen2 = slen_table[1][g->scalefac_compress];
  1412. av_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
  1413. if (g->block_type == 2) {
  1414. n = g->switch_point ? 17 : 18;
  1415. j = 0;
  1416. if (slen1) {
  1417. for (i = 0; i < n; i++)
  1418. g->scale_factors[j++] = get_bits(&s->gb, slen1);
  1419. } else {
  1420. for (i = 0; i < n; i++)
  1421. g->scale_factors[j++] = 0;
  1422. }
  1423. if (slen2) {
  1424. for (i = 0; i < 18; i++)
  1425. g->scale_factors[j++] = get_bits(&s->gb, slen2);
  1426. for (i = 0; i < 3; i++)
  1427. g->scale_factors[j++] = 0;
  1428. } else {
  1429. for (i = 0; i < 21; i++)
  1430. g->scale_factors[j++] = 0;
  1431. }
  1432. } else {
  1433. sc = s->granules[ch][0].scale_factors;
  1434. j = 0;
  1435. for (k = 0; k < 4; k++) {
  1436. n = k == 0 ? 6 : 5;
  1437. if ((g->scfsi & (0x8 >> k)) == 0) {
  1438. slen = (k < 2) ? slen1 : slen2;
  1439. if (slen) {
  1440. for (i = 0; i < n; i++)
  1441. g->scale_factors[j++] = get_bits(&s->gb, slen);
  1442. } else {
  1443. for (i = 0; i < n; i++)
  1444. g->scale_factors[j++] = 0;
  1445. }
  1446. } else {
  1447. /* simply copy from last granule */
  1448. for (i = 0; i < n; i++) {
  1449. g->scale_factors[j] = sc[j];
  1450. j++;
  1451. }
  1452. }
  1453. }
  1454. g->scale_factors[j++] = 0;
  1455. }
  1456. } else {
  1457. int tindex, tindex2, slen[4], sl, sf;
  1458. /* LSF scale factors */
  1459. if (g->block_type == 2)
  1460. tindex = g->switch_point ? 2 : 1;
  1461. else
  1462. tindex = 0;
  1463. sf = g->scalefac_compress;
  1464. if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
  1465. /* intensity stereo case */
  1466. sf >>= 1;
  1467. if (sf < 180) {
  1468. lsf_sf_expand(slen, sf, 6, 6, 0);
  1469. tindex2 = 3;
  1470. } else if (sf < 244) {
  1471. lsf_sf_expand(slen, sf - 180, 4, 4, 0);
  1472. tindex2 = 4;
  1473. } else {
  1474. lsf_sf_expand(slen, sf - 244, 3, 0, 0);
  1475. tindex2 = 5;
  1476. }
  1477. } else {
  1478. /* normal case */
  1479. if (sf < 400) {
  1480. lsf_sf_expand(slen, sf, 5, 4, 4);
  1481. tindex2 = 0;
  1482. } else if (sf < 500) {
  1483. lsf_sf_expand(slen, sf - 400, 5, 4, 0);
  1484. tindex2 = 1;
  1485. } else {
  1486. lsf_sf_expand(slen, sf - 500, 3, 0, 0);
  1487. tindex2 = 2;
  1488. g->preflag = 1;
  1489. }
  1490. }
  1491. j = 0;
  1492. for (k = 0; k < 4; k++) {
  1493. n = lsf_nsf_table[tindex2][tindex][k];
  1494. sl = slen[k];
  1495. if (sl) {
  1496. for (i = 0; i < n; i++)
  1497. g->scale_factors[j++] = get_bits(&s->gb, sl);
  1498. } else {
  1499. for (i = 0; i < n; i++)
  1500. g->scale_factors[j++] = 0;
  1501. }
  1502. }
  1503. /* XXX: should compute exact size */
  1504. for (; j < 40; j++)
  1505. g->scale_factors[j] = 0;
  1506. }
  1507. exponents_from_scale_factors(s, g, exponents);
  1508. /* read Huffman coded residue */
  1509. huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
  1510. } /* ch */
  1511. if (s->nb_channels == 2)
  1512. compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
  1513. for (ch = 0; ch < s->nb_channels; ch++) {
  1514. g = &s->granules[ch][gr];
  1515. reorder_block(s, g);
  1516. compute_antialias(s, g);
  1517. compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
  1518. }
  1519. } /* gr */
  1520. if (get_bits_count(&s->gb) < 0)
  1521. skip_bits_long(&s->gb, -get_bits_count(&s->gb));
  1522. return nb_granules * 18;
  1523. }
  1524. static int mp_decode_frame(MPADecodeContext *s, OUT_INT *samples,
  1525. const uint8_t *buf, int buf_size)
  1526. {
  1527. int i, nb_frames, ch;
  1528. OUT_INT *samples_ptr;
  1529. init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);
  1530. /* skip error protection field */
  1531. if (s->error_protection)
  1532. skip_bits(&s->gb, 16);
  1533. switch(s->layer) {
  1534. case 1:
  1535. s->avctx->frame_size = 384;
  1536. nb_frames = mp_decode_layer1(s);
  1537. break;
  1538. case 2:
  1539. s->avctx->frame_size = 1152;
  1540. nb_frames = mp_decode_layer2(s);
  1541. break;
  1542. case 3:
  1543. s->avctx->frame_size = s->lsf ? 576 : 1152;
  1544. default:
  1545. nb_frames = mp_decode_layer3(s);
  1546. s->last_buf_size=0;
  1547. if (s->in_gb.buffer) {
  1548. align_get_bits(&s->gb);
  1549. i = get_bits_left(&s->gb)>>3;
  1550. if (i >= 0 && i <= BACKSTEP_SIZE) {
  1551. memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);
  1552. s->last_buf_size=i;
  1553. } else
  1554. av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
  1555. s->gb = s->in_gb;
  1556. s->in_gb.buffer = NULL;
  1557. }
  1558. align_get_bits(&s->gb);
  1559. assert((get_bits_count(&s->gb) & 7) == 0);
  1560. i = get_bits_left(&s->gb) >> 3;
  1561. if (i < 0 || i > BACKSTEP_SIZE || nb_frames < 0) {
  1562. if (i < 0)
  1563. av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
  1564. i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
  1565. }
  1566. assert(i <= buf_size - HEADER_SIZE && i >= 0);
  1567. memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
  1568. s->last_buf_size += i;
  1569. break;
  1570. }
  1571. /* apply the synthesis filter */
  1572. for (ch = 0; ch < s->nb_channels; ch++) {
  1573. samples_ptr = samples + ch;
  1574. for (i = 0; i < nb_frames; i++) {
  1575. RENAME(ff_mpa_synth_filter)(
  1576. &s->mpadsp,
  1577. s->synth_buf[ch], &(s->synth_buf_offset[ch]),
  1578. RENAME(ff_mpa_synth_window), &s->dither_state,
  1579. samples_ptr, s->nb_channels,
  1580. s->sb_samples[ch][i]);
  1581. samples_ptr += 32 * s->nb_channels;
  1582. }
  1583. }
  1584. return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
  1585. }
  1586. static int decode_frame(AVCodecContext * avctx, void *data, int *data_size,
  1587. AVPacket *avpkt)
  1588. {
  1589. const uint8_t *buf = avpkt->data;
  1590. int buf_size = avpkt->size;
  1591. MPADecodeContext *s = avctx->priv_data;
  1592. uint32_t header;
  1593. int out_size;
  1594. OUT_INT *out_samples = data;
  1595. if (buf_size < HEADER_SIZE)
  1596. return AVERROR_INVALIDDATA;
  1597. header = AV_RB32(buf);
  1598. if (ff_mpa_check_header(header) < 0) {
  1599. av_log(avctx, AV_LOG_ERROR, "Header missing\n");
  1600. return AVERROR_INVALIDDATA;
  1601. }
  1602. if (avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
  1603. /* free format: prepare to compute frame size */
  1604. s->frame_size = -1;
  1605. return AVERROR_INVALIDDATA;
  1606. }
  1607. /* update codec info */
  1608. avctx->channels = s->nb_channels;
  1609. avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
  1610. if (!avctx->bit_rate)
  1611. avctx->bit_rate = s->bit_rate;
  1612. avctx->sub_id = s->layer;
  1613. if (*data_size < avctx->frame_size * avctx->channels * sizeof(OUT_INT))
  1614. return AVERROR(EINVAL);
  1615. *data_size = 0;
  1616. if (s->frame_size <= 0 || s->frame_size > buf_size) {
  1617. av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
  1618. return AVERROR_INVALIDDATA;
  1619. } else if (s->frame_size < buf_size) {
  1620. av_log(avctx, AV_LOG_ERROR, "incorrect frame size\n");
  1621. buf_size= s->frame_size;
  1622. }
  1623. out_size = mp_decode_frame(s, out_samples, buf, buf_size);
  1624. if (out_size >= 0) {
  1625. *data_size = out_size;
  1626. avctx->sample_rate = s->sample_rate;
  1627. //FIXME maybe move the other codec info stuff from above here too
  1628. } else {
  1629. av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
  1630. /* Only return an error if the bad frame makes up the whole packet.
  1631. If there is more data in the packet, just consume the bad frame
  1632. instead of returning an error, which would discard the whole
  1633. packet. */
  1634. if (buf_size == avpkt->size)
  1635. return out_size;
  1636. }
  1637. s->frame_size = 0;
  1638. return buf_size;
  1639. }
  1640. static void flush(AVCodecContext *avctx)
  1641. {
  1642. MPADecodeContext *s = avctx->priv_data;
  1643. memset(s->synth_buf, 0, sizeof(s->synth_buf));
  1644. s->last_buf_size = 0;
  1645. }
  1646. #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
  1647. static int decode_frame_adu(AVCodecContext *avctx, void *data, int *data_size,
  1648. AVPacket *avpkt)
  1649. {
  1650. const uint8_t *buf = avpkt->data;
  1651. int buf_size = avpkt->size;
  1652. MPADecodeContext *s = avctx->priv_data;
  1653. uint32_t header;
  1654. int len, out_size;
  1655. OUT_INT *out_samples = data;
  1656. len = buf_size;
  1657. // Discard too short frames
  1658. if (buf_size < HEADER_SIZE) {
  1659. av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
  1660. return AVERROR_INVALIDDATA;
  1661. }
  1662. if (len > MPA_MAX_CODED_FRAME_SIZE)
  1663. len = MPA_MAX_CODED_FRAME_SIZE;
  1664. // Get header and restore sync word
  1665. header = AV_RB32(buf) | 0xffe00000;
  1666. if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame
  1667. av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n");
  1668. return AVERROR_INVALIDDATA;
  1669. }
  1670. avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);
  1671. /* update codec info */
  1672. avctx->sample_rate = s->sample_rate;
  1673. avctx->channels = s->nb_channels;
  1674. if (!avctx->bit_rate)
  1675. avctx->bit_rate = s->bit_rate;
  1676. avctx->sub_id = s->layer;
  1677. if (*data_size < avctx->frame_size * avctx->channels * sizeof(OUT_INT))
  1678. return AVERROR(EINVAL);
  1679. s->frame_size = len;
  1680. #if FF_API_PARSE_FRAME
  1681. if (avctx->parse_only)
  1682. out_size = buf_size;
  1683. else
  1684. #endif
  1685. out_size = mp_decode_frame(s, out_samples, buf, buf_size);
  1686. *data_size = out_size;
  1687. return buf_size;
  1688. }
  1689. #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
  1690. #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
  1691. /**
  1692. * Context for MP3On4 decoder
  1693. */
  1694. typedef struct MP3On4DecodeContext {
  1695. int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
  1696. int syncword; ///< syncword patch
  1697. const uint8_t *coff; ///< channel offsets in output buffer
  1698. MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
  1699. OUT_INT *decoded_buf; ///< output buffer for decoded samples
  1700. } MP3On4DecodeContext;
  1701. #include "mpeg4audio.h"
  1702. /* Next 3 arrays are indexed by channel config number (passed via codecdata) */
  1703. /* number of mp3 decoder instances */
  1704. static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 };
  1705. /* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */
  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. { 2, 0, 3 }, // C FLR BLRS
  1713. { 2, 0, 4, 3 }, // C FLR BLRS LFE
  1714. { 2, 0, 6, 4, 3 }, // C FLR BLRS BLR LFE
  1715. };
  1716. /* mp3on4 channel layouts */
  1717. static const int16_t chan_layout[8] = {
  1718. 0,
  1719. AV_CH_LAYOUT_MONO,
  1720. AV_CH_LAYOUT_STEREO,
  1721. AV_CH_LAYOUT_SURROUND,
  1722. AV_CH_LAYOUT_4POINT0,
  1723. AV_CH_LAYOUT_5POINT0,
  1724. AV_CH_LAYOUT_5POINT1,
  1725. AV_CH_LAYOUT_7POINT1
  1726. };
  1727. static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
  1728. {
  1729. MP3On4DecodeContext *s = avctx->priv_data;
  1730. int i;
  1731. for (i = 0; i < s->frames; i++)
  1732. av_free(s->mp3decctx[i]);
  1733. av_freep(&s->decoded_buf);
  1734. return 0;
  1735. }
  1736. static int decode_init_mp3on4(AVCodecContext * avctx)
  1737. {
  1738. MP3On4DecodeContext *s = avctx->priv_data;
  1739. MPEG4AudioConfig cfg;
  1740. int i;
  1741. if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) {
  1742. av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
  1743. return AVERROR_INVALIDDATA;
  1744. }
  1745. avpriv_mpeg4audio_get_config(&cfg, avctx->extradata, avctx->extradata_size);
  1746. if (!cfg.chan_config || cfg.chan_config > 7) {
  1747. av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
  1748. return AVERROR_INVALIDDATA;
  1749. }
  1750. s->frames = mp3Frames[cfg.chan_config];
  1751. s->coff = chan_offset[cfg.chan_config];
  1752. avctx->channels = ff_mpeg4audio_channels[cfg.chan_config];
  1753. avctx->channel_layout = chan_layout[cfg.chan_config];
  1754. if (cfg.sample_rate < 16000)
  1755. s->syncword = 0xffe00000;
  1756. else
  1757. s->syncword = 0xfff00000;
  1758. /* Init the first mp3 decoder in standard way, so that all tables get builded
  1759. * We replace avctx->priv_data with the context of the first decoder so that
  1760. * decode_init() does not have to be changed.
  1761. * Other decoders will be initialized here copying data from the first context
  1762. */
  1763. // Allocate zeroed memory for the first decoder context
  1764. s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
  1765. if (!s->mp3decctx[0])
  1766. goto alloc_fail;
  1767. // Put decoder context in place to make init_decode() happy
  1768. avctx->priv_data = s->mp3decctx[0];
  1769. decode_init(avctx);
  1770. // Restore mp3on4 context pointer
  1771. avctx->priv_data = s;
  1772. s->mp3decctx[0]->adu_mode = 1; // Set adu mode
  1773. /* Create a separate codec/context for each frame (first is already ok).
  1774. * Each frame is 1 or 2 channels - up to 5 frames allowed
  1775. */
  1776. for (i = 1; i < s->frames; i++) {
  1777. s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
  1778. if (!s->mp3decctx[i])
  1779. goto alloc_fail;
  1780. s->mp3decctx[i]->adu_mode = 1;
  1781. s->mp3decctx[i]->avctx = avctx;
  1782. s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;
  1783. }
  1784. /* Allocate buffer for multi-channel output if needed */
  1785. if (s->frames > 1) {
  1786. s->decoded_buf = av_malloc(MPA_FRAME_SIZE * MPA_MAX_CHANNELS *
  1787. sizeof(*s->decoded_buf));
  1788. if (!s->decoded_buf)
  1789. goto alloc_fail;
  1790. }
  1791. return 0;
  1792. alloc_fail:
  1793. decode_close_mp3on4(avctx);
  1794. return AVERROR(ENOMEM);
  1795. }
  1796. static void flush_mp3on4(AVCodecContext *avctx)
  1797. {
  1798. int i;
  1799. MP3On4DecodeContext *s = avctx->priv_data;
  1800. for (i = 0; i < s->frames; i++) {
  1801. MPADecodeContext *m = s->mp3decctx[i];
  1802. memset(m->synth_buf, 0, sizeof(m->synth_buf));
  1803. m->last_buf_size = 0;
  1804. }
  1805. }
  1806. static int decode_frame_mp3on4(AVCodecContext * avctx,
  1807. void *data, int *data_size,
  1808. AVPacket *avpkt)
  1809. {
  1810. const uint8_t *buf = avpkt->data;
  1811. int buf_size = avpkt->size;
  1812. MP3On4DecodeContext *s = avctx->priv_data;
  1813. MPADecodeContext *m;
  1814. int fsize, len = buf_size, out_size = 0;
  1815. uint32_t header;
  1816. OUT_INT *out_samples = data;
  1817. OUT_INT *outptr, *bp;
  1818. int fr, j, n, ch;
  1819. if (*data_size < MPA_FRAME_SIZE * avctx->channels * sizeof(OUT_INT)) {
  1820. av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
  1821. return AVERROR(EINVAL);
  1822. }
  1823. // Discard too short frames
  1824. if (buf_size < HEADER_SIZE)
  1825. return AVERROR_INVALIDDATA;
  1826. // If only one decoder interleave is not needed
  1827. outptr = s->frames == 1 ? out_samples : s->decoded_buf;
  1828. avctx->bit_rate = 0;
  1829. ch = 0;
  1830. for (fr = 0; fr < s->frames; fr++) {
  1831. fsize = AV_RB16(buf) >> 4;
  1832. fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
  1833. m = s->mp3decctx[fr];
  1834. assert(m != NULL);
  1835. header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
  1836. if (ff_mpa_check_header(header) < 0) // Bad header, discard block
  1837. break;
  1838. avpriv_mpegaudio_decode_header((MPADecodeHeader *)m, header);
  1839. if (ch + m->nb_channels > avctx->channels) {
  1840. av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "
  1841. "channel count\n");
  1842. return AVERROR_INVALIDDATA;
  1843. }
  1844. ch += m->nb_channels;
  1845. out_size += mp_decode_frame(m, outptr, buf, fsize);
  1846. buf += fsize;
  1847. len -= fsize;
  1848. if (s->frames > 1) {
  1849. n = m->avctx->frame_size*m->nb_channels;
  1850. /* interleave output data */
  1851. bp = out_samples + s->coff[fr];
  1852. if (m->nb_channels == 1) {
  1853. for (j = 0; j < n; j++) {
  1854. *bp = s->decoded_buf[j];
  1855. bp += avctx->channels;
  1856. }
  1857. } else {
  1858. for (j = 0; j < n; j++) {
  1859. bp[0] = s->decoded_buf[j++];
  1860. bp[1] = s->decoded_buf[j];
  1861. bp += avctx->channels;
  1862. }
  1863. }
  1864. }
  1865. avctx->bit_rate += m->bit_rate;
  1866. }
  1867. /* update codec info */
  1868. avctx->sample_rate = s->mp3decctx[0]->sample_rate;
  1869. *data_size = out_size;
  1870. return buf_size;
  1871. }
  1872. #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
  1873. #if !CONFIG_FLOAT
  1874. #if CONFIG_MP1_DECODER
  1875. AVCodec ff_mp1_decoder = {
  1876. .name = "mp1",
  1877. .type = AVMEDIA_TYPE_AUDIO,
  1878. .id = CODEC_ID_MP1,
  1879. .priv_data_size = sizeof(MPADecodeContext),
  1880. .init_static_data = decode_init_static,
  1881. .init = decode_init,
  1882. .decode = decode_frame,
  1883. #if FF_API_PARSE_FRAME
  1884. .capabilities = CODEC_CAP_PARSE_ONLY,
  1885. #endif
  1886. .flush = flush,
  1887. .long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
  1888. };
  1889. #endif
  1890. #if CONFIG_MP2_DECODER
  1891. AVCodec ff_mp2_decoder = {
  1892. .name = "mp2",
  1893. .type = AVMEDIA_TYPE_AUDIO,
  1894. .id = CODEC_ID_MP2,
  1895. .priv_data_size = sizeof(MPADecodeContext),
  1896. .init_static_data = decode_init_static,
  1897. .init = decode_init,
  1898. .decode = decode_frame,
  1899. #if FF_API_PARSE_FRAME
  1900. .capabilities = CODEC_CAP_PARSE_ONLY,
  1901. #endif
  1902. .flush = flush,
  1903. .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
  1904. };
  1905. #endif
  1906. #if CONFIG_MP3_DECODER
  1907. AVCodec ff_mp3_decoder = {
  1908. .name = "mp3",
  1909. .type = AVMEDIA_TYPE_AUDIO,
  1910. .id = CODEC_ID_MP3,
  1911. .priv_data_size = sizeof(MPADecodeContext),
  1912. .init_static_data = decode_init_static,
  1913. .init = decode_init,
  1914. .decode = decode_frame,
  1915. #if FF_API_PARSE_FRAME
  1916. .capabilities = CODEC_CAP_PARSE_ONLY,
  1917. #endif
  1918. .flush = flush,
  1919. .long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
  1920. };
  1921. #endif
  1922. #if CONFIG_MP3ADU_DECODER
  1923. AVCodec ff_mp3adu_decoder = {
  1924. .name = "mp3adu",
  1925. .type = AVMEDIA_TYPE_AUDIO,
  1926. .id = CODEC_ID_MP3ADU,
  1927. .priv_data_size = sizeof(MPADecodeContext),
  1928. .init_static_data = decode_init_static,
  1929. .init = decode_init,
  1930. .decode = decode_frame_adu,
  1931. #if FF_API_PARSE_FRAME
  1932. .capabilities = CODEC_CAP_PARSE_ONLY,
  1933. #endif
  1934. .flush = flush,
  1935. .long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
  1936. };
  1937. #endif
  1938. #if CONFIG_MP3ON4_DECODER
  1939. AVCodec ff_mp3on4_decoder = {
  1940. .name = "mp3on4",
  1941. .type = AVMEDIA_TYPE_AUDIO,
  1942. .id = CODEC_ID_MP3ON4,
  1943. .priv_data_size = sizeof(MP3On4DecodeContext),
  1944. .init_static_data = decode_init_static,
  1945. .init = decode_init_mp3on4,
  1946. .close = decode_close_mp3on4,
  1947. .decode = decode_frame_mp3on4,
  1948. .flush = flush_mp3on4,
  1949. .long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
  1950. };
  1951. #endif
  1952. #endif