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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/attributes.h"
  26. #include "libavutil/avassert.h"
  27. #include "libavutil/channel_layout.h"
  28. #include "libavutil/float_dsp.h"
  29. #include "avcodec.h"
  30. #include "get_bits.h"
  31. #include "internal.h"
  32. #include "mathops.h"
  33. #include "mpegaudiodsp.h"
  34. /*
  35. * TODO:
  36. * - test lsf / mpeg25 extensively.
  37. */
  38. #include "mpegaudio.h"
  39. #include "mpegaudiodecheader.h"
  40. #define BACKSTEP_SIZE 512
  41. #define EXTRABYTES 24
  42. #define LAST_BUF_SIZE 2 * BACKSTEP_SIZE + EXTRABYTES
  43. /* layer 3 "granule" */
  44. typedef struct GranuleDef {
  45. uint8_t scfsi;
  46. int part2_3_length;
  47. int big_values;
  48. int global_gain;
  49. int scalefac_compress;
  50. uint8_t block_type;
  51. uint8_t switch_point;
  52. int table_select[3];
  53. int subblock_gain[3];
  54. uint8_t scalefac_scale;
  55. uint8_t count1table_select;
  56. int region_size[3]; /* number of huffman codes in each region */
  57. int preflag;
  58. int short_start, long_end; /* long/short band indexes */
  59. uint8_t scale_factors[40];
  60. DECLARE_ALIGNED(16, INTFLOAT, sb_hybrid)[SBLIMIT * 18]; /* 576 samples */
  61. } GranuleDef;
  62. typedef struct MPADecodeContext {
  63. MPA_DECODE_HEADER
  64. uint8_t last_buf[LAST_BUF_SIZE];
  65. int last_buf_size;
  66. int extrasize;
  67. /* next header (used in free format parsing) */
  68. uint32_t free_format_next_header;
  69. GetBitContext gb;
  70. GetBitContext in_gb;
  71. DECLARE_ALIGNED(32, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512 * 2];
  72. int synth_buf_offset[MPA_MAX_CHANNELS];
  73. DECLARE_ALIGNED(32, INTFLOAT, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT];
  74. INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
  75. GranuleDef granules[2][2]; /* Used in Layer 3 */
  76. int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
  77. int dither_state;
  78. int err_recognition;
  79. AVCodecContext* avctx;
  80. MPADSPContext mpadsp;
  81. AVFloatDSPContext fdsp;
  82. AVFrame *frame;
  83. } MPADecodeContext;
  84. #define HEADER_SIZE 4
  85. #include "mpegaudiodata.h"
  86. #include "mpegaudiodectab.h"
  87. /* vlc structure for decoding layer 3 huffman tables */
  88. static VLC huff_vlc[16];
  89. static VLC_TYPE huff_vlc_tables[
  90. 0 + 128 + 128 + 128 + 130 + 128 + 154 + 166 +
  91. 142 + 204 + 190 + 170 + 542 + 460 + 662 + 414
  92. ][2];
  93. static const int huff_vlc_tables_sizes[16] = {
  94. 0, 128, 128, 128, 130, 128, 154, 166,
  95. 142, 204, 190, 170, 542, 460, 662, 414
  96. };
  97. static VLC huff_quad_vlc[2];
  98. static VLC_TYPE huff_quad_vlc_tables[128+16][2];
  99. static const int huff_quad_vlc_tables_sizes[2] = { 128, 16 };
  100. /* computed from band_size_long */
  101. static uint16_t band_index_long[9][23];
  102. #include "mpegaudio_tablegen.h"
  103. /* intensity stereo coef table */
  104. static INTFLOAT is_table[2][16];
  105. static INTFLOAT is_table_lsf[2][2][16];
  106. static INTFLOAT csa_table[8][4];
  107. static int16_t division_tab3[1<<6 ];
  108. static int16_t division_tab5[1<<8 ];
  109. static int16_t division_tab9[1<<11];
  110. static int16_t * const division_tabs[4] = {
  111. division_tab3, division_tab5, NULL, division_tab9
  112. };
  113. /* lower 2 bits: modulo 3, higher bits: shift */
  114. static uint16_t scale_factor_modshift[64];
  115. /* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
  116. static int32_t scale_factor_mult[15][3];
  117. /* mult table for layer 2 group quantization */
  118. #define SCALE_GEN(v) \
  119. { FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
  120. static const int32_t scale_factor_mult2[3][3] = {
  121. SCALE_GEN(4.0 / 3.0), /* 3 steps */
  122. SCALE_GEN(4.0 / 5.0), /* 5 steps */
  123. SCALE_GEN(4.0 / 9.0), /* 9 steps */
  124. };
  125. /**
  126. * Convert region offsets to region sizes and truncate
  127. * size to big_values.
  128. */
  129. static void region_offset2size(GranuleDef *g)
  130. {
  131. int i, k, j = 0;
  132. g->region_size[2] = 576 / 2;
  133. for (i = 0; i < 3; i++) {
  134. k = FFMIN(g->region_size[i], g->big_values);
  135. g->region_size[i] = k - j;
  136. j = k;
  137. }
  138. }
  139. static void init_short_region(MPADecodeContext *s, GranuleDef *g)
  140. {
  141. if (g->block_type == 2) {
  142. if (s->sample_rate_index != 8)
  143. g->region_size[0] = (36 / 2);
  144. else
  145. g->region_size[0] = (72 / 2);
  146. } else {
  147. if (s->sample_rate_index <= 2)
  148. g->region_size[0] = (36 / 2);
  149. else if (s->sample_rate_index != 8)
  150. g->region_size[0] = (54 / 2);
  151. else
  152. g->region_size[0] = (108 / 2);
  153. }
  154. g->region_size[1] = (576 / 2);
  155. }
  156. static void init_long_region(MPADecodeContext *s, GranuleDef *g,
  157. int ra1, int ra2)
  158. {
  159. int l;
  160. g->region_size[0] = band_index_long[s->sample_rate_index][ra1 + 1] >> 1;
  161. /* should not overflow */
  162. l = FFMIN(ra1 + ra2 + 2, 22);
  163. g->region_size[1] = band_index_long[s->sample_rate_index][ l] >> 1;
  164. }
  165. static void compute_band_indexes(MPADecodeContext *s, GranuleDef *g)
  166. {
  167. if (g->block_type == 2) {
  168. if (g->switch_point) {
  169. /* if switched mode, we handle the 36 first samples as
  170. long blocks. For 8000Hz, we handle the 72 first
  171. exponents as long blocks */
  172. if (s->sample_rate_index <= 2)
  173. g->long_end = 8;
  174. else
  175. g->long_end = 6;
  176. g->short_start = 3;
  177. } else {
  178. g->long_end = 0;
  179. g->short_start = 0;
  180. }
  181. } else {
  182. g->short_start = 13;
  183. g->long_end = 22;
  184. }
  185. }
  186. /* layer 1 unscaling */
  187. /* n = number of bits of the mantissa minus 1 */
  188. static inline int l1_unscale(int n, int mant, int scale_factor)
  189. {
  190. int shift, mod;
  191. int64_t val;
  192. shift = scale_factor_modshift[scale_factor];
  193. mod = shift & 3;
  194. shift >>= 2;
  195. val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]);
  196. shift += n;
  197. /* NOTE: at this point, 1 <= shift >= 21 + 15 */
  198. return (int)((val + (1LL << (shift - 1))) >> shift);
  199. }
  200. static inline int l2_unscale_group(int steps, int mant, int scale_factor)
  201. {
  202. int shift, mod, val;
  203. shift = scale_factor_modshift[scale_factor];
  204. mod = shift & 3;
  205. shift >>= 2;
  206. val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
  207. /* NOTE: at this point, 0 <= shift <= 21 */
  208. if (shift > 0)
  209. val = (val + (1 << (shift - 1))) >> shift;
  210. return val;
  211. }
  212. /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
  213. static inline int l3_unscale(int value, int exponent)
  214. {
  215. unsigned int m;
  216. int e;
  217. e = table_4_3_exp [4 * value + (exponent & 3)];
  218. m = table_4_3_value[4 * value + (exponent & 3)];
  219. e -= exponent >> 2;
  220. assert(e >= 1);
  221. if (e > 31)
  222. return 0;
  223. m = (m + (1 << (e - 1))) >> e;
  224. return m;
  225. }
  226. static av_cold void decode_init_static(void)
  227. {
  228. int i, j, k;
  229. int offset;
  230. /* scale factors table for layer 1/2 */
  231. for (i = 0; i < 64; i++) {
  232. int shift, mod;
  233. /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
  234. shift = i / 3;
  235. mod = i % 3;
  236. scale_factor_modshift[i] = mod | (shift << 2);
  237. }
  238. /* scale factor multiply for layer 1 */
  239. for (i = 0; i < 15; i++) {
  240. int n, norm;
  241. n = i + 2;
  242. norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
  243. scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
  244. scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
  245. scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
  246. ff_dlog(NULL, "%d: norm=%x s=%x %x %x\n", i, norm,
  247. scale_factor_mult[i][0],
  248. scale_factor_mult[i][1],
  249. scale_factor_mult[i][2]);
  250. }
  251. RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));
  252. /* huffman decode tables */
  253. offset = 0;
  254. for (i = 1; i < 16; i++) {
  255. const HuffTable *h = &mpa_huff_tables[i];
  256. int xsize, x, y;
  257. uint8_t tmp_bits [512] = { 0 };
  258. uint16_t tmp_codes[512] = { 0 };
  259. xsize = h->xsize;
  260. j = 0;
  261. for (x = 0; x < xsize; x++) {
  262. for (y = 0; y < xsize; y++) {
  263. tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ];
  264. tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];
  265. }
  266. }
  267. /* XXX: fail test */
  268. huff_vlc[i].table = huff_vlc_tables+offset;
  269. huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
  270. init_vlc(&huff_vlc[i], 7, 512,
  271. tmp_bits, 1, 1, tmp_codes, 2, 2,
  272. INIT_VLC_USE_NEW_STATIC);
  273. offset += huff_vlc_tables_sizes[i];
  274. }
  275. assert(offset == FF_ARRAY_ELEMS(huff_vlc_tables));
  276. offset = 0;
  277. for (i = 0; i < 2; i++) {
  278. huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
  279. huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
  280. init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
  281. mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
  282. INIT_VLC_USE_NEW_STATIC);
  283. offset += huff_quad_vlc_tables_sizes[i];
  284. }
  285. assert(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));
  286. for (i = 0; i < 9; i++) {
  287. k = 0;
  288. for (j = 0; j < 22; j++) {
  289. band_index_long[i][j] = k;
  290. k += band_size_long[i][j];
  291. }
  292. band_index_long[i][22] = k;
  293. }
  294. /* compute n ^ (4/3) and store it in mantissa/exp format */
  295. mpegaudio_tableinit();
  296. for (i = 0; i < 4; i++) {
  297. if (ff_mpa_quant_bits[i] < 0) {
  298. for (j = 0; j < (1 << (-ff_mpa_quant_bits[i]+1)); j++) {
  299. int val1, val2, val3, steps;
  300. int val = j;
  301. steps = ff_mpa_quant_steps[i];
  302. val1 = val % steps;
  303. val /= steps;
  304. val2 = val % steps;
  305. val3 = val / steps;
  306. division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);
  307. }
  308. }
  309. }
  310. for (i = 0; i < 7; i++) {
  311. float f;
  312. INTFLOAT v;
  313. if (i != 6) {
  314. f = tan((double)i * M_PI / 12.0);
  315. v = FIXR(f / (1.0 + f));
  316. } else {
  317. v = FIXR(1.0);
  318. }
  319. is_table[0][ i] = v;
  320. is_table[1][6 - i] = v;
  321. }
  322. /* invalid values */
  323. for (i = 7; i < 16; i++)
  324. is_table[0][i] = is_table[1][i] = 0.0;
  325. for (i = 0; i < 16; i++) {
  326. double f;
  327. int e, k;
  328. for (j = 0; j < 2; j++) {
  329. e = -(j + 1) * ((i + 1) >> 1);
  330. f = pow(2.0, e / 4.0);
  331. k = i & 1;
  332. is_table_lsf[j][k ^ 1][i] = FIXR(f);
  333. is_table_lsf[j][k ][i] = FIXR(1.0);
  334. ff_dlog(NULL, "is_table_lsf %d %d: %f %f\n",
  335. i, j, (float) is_table_lsf[j][0][i],
  336. (float) is_table_lsf[j][1][i]);
  337. }
  338. }
  339. for (i = 0; i < 8; i++) {
  340. float ci, cs, ca;
  341. ci = ci_table[i];
  342. cs = 1.0 / sqrt(1.0 + ci * ci);
  343. ca = cs * ci;
  344. #if !CONFIG_FLOAT
  345. csa_table[i][0] = FIXHR(cs/4);
  346. csa_table[i][1] = FIXHR(ca/4);
  347. csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);
  348. csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);
  349. #else
  350. csa_table[i][0] = cs;
  351. csa_table[i][1] = ca;
  352. csa_table[i][2] = ca + cs;
  353. csa_table[i][3] = ca - cs;
  354. #endif
  355. }
  356. }
  357. static av_cold int decode_init(AVCodecContext * avctx)
  358. {
  359. static int initialized_tables = 0;
  360. MPADecodeContext *s = avctx->priv_data;
  361. if (!initialized_tables) {
  362. decode_init_static();
  363. initialized_tables = 1;
  364. }
  365. s->avctx = avctx;
  366. avpriv_float_dsp_init(&s->fdsp, avctx->flags & AV_CODEC_FLAG_BITEXACT);
  367. ff_mpadsp_init(&s->mpadsp);
  368. if (avctx->request_sample_fmt == OUT_FMT &&
  369. avctx->codec_id != AV_CODEC_ID_MP3ON4)
  370. avctx->sample_fmt = OUT_FMT;
  371. else
  372. avctx->sample_fmt = OUT_FMT_P;
  373. s->err_recognition = avctx->err_recognition;
  374. if (avctx->codec_id == AV_CODEC_ID_MP3ADU)
  375. s->adu_mode = 1;
  376. return 0;
  377. }
  378. #define C3 FIXHR(0.86602540378443864676/2)
  379. #define C4 FIXHR(0.70710678118654752439/2) //0.5 / cos(pi*(9)/36)
  380. #define C5 FIXHR(0.51763809020504152469/2) //0.5 / cos(pi*(5)/36)
  381. #define C6 FIXHR(1.93185165257813657349/4) //0.5 / cos(pi*(15)/36)
  382. /* 12 points IMDCT. We compute it "by hand" by factorizing obvious
  383. cases. */
  384. static void imdct12(INTFLOAT *out, INTFLOAT *in)
  385. {
  386. INTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
  387. in0 = in[0*3];
  388. in1 = in[1*3] + in[0*3];
  389. in2 = in[2*3] + in[1*3];
  390. in3 = in[3*3] + in[2*3];
  391. in4 = in[4*3] + in[3*3];
  392. in5 = in[5*3] + in[4*3];
  393. in5 += in3;
  394. in3 += in1;
  395. in2 = MULH3(in2, C3, 2);
  396. in3 = MULH3(in3, C3, 4);
  397. t1 = in0 - in4;
  398. t2 = MULH3(in1 - in5, C4, 2);
  399. out[ 7] =
  400. out[10] = t1 + t2;
  401. out[ 1] =
  402. out[ 4] = t1 - t2;
  403. in0 += SHR(in4, 1);
  404. in4 = in0 + in2;
  405. in5 += 2*in1;
  406. in1 = MULH3(in5 + in3, C5, 1);
  407. out[ 8] =
  408. out[ 9] = in4 + in1;
  409. out[ 2] =
  410. out[ 3] = in4 - in1;
  411. in0 -= in2;
  412. in5 = MULH3(in5 - in3, C6, 2);
  413. out[ 0] =
  414. out[ 5] = in0 - in5;
  415. out[ 6] =
  416. out[11] = in0 + in5;
  417. }
  418. /* return the number of decoded frames */
  419. static int mp_decode_layer1(MPADecodeContext *s)
  420. {
  421. int bound, i, v, n, ch, j, mant;
  422. uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
  423. uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
  424. if (s->mode == MPA_JSTEREO)
  425. bound = (s->mode_ext + 1) * 4;
  426. else
  427. bound = SBLIMIT;
  428. /* allocation bits */
  429. for (i = 0; i < bound; i++) {
  430. for (ch = 0; ch < s->nb_channels; ch++) {
  431. allocation[ch][i] = get_bits(&s->gb, 4);
  432. }
  433. }
  434. for (i = bound; i < SBLIMIT; i++)
  435. allocation[0][i] = get_bits(&s->gb, 4);
  436. /* scale factors */
  437. for (i = 0; i < bound; i++) {
  438. for (ch = 0; ch < s->nb_channels; ch++) {
  439. if (allocation[ch][i])
  440. scale_factors[ch][i] = get_bits(&s->gb, 6);
  441. }
  442. }
  443. for (i = bound; i < SBLIMIT; i++) {
  444. if (allocation[0][i]) {
  445. scale_factors[0][i] = get_bits(&s->gb, 6);
  446. scale_factors[1][i] = get_bits(&s->gb, 6);
  447. }
  448. }
  449. /* compute samples */
  450. for (j = 0; j < 12; j++) {
  451. for (i = 0; i < bound; i++) {
  452. for (ch = 0; ch < s->nb_channels; ch++) {
  453. n = allocation[ch][i];
  454. if (n) {
  455. mant = get_bits(&s->gb, n + 1);
  456. v = l1_unscale(n, mant, scale_factors[ch][i]);
  457. } else {
  458. v = 0;
  459. }
  460. s->sb_samples[ch][j][i] = v;
  461. }
  462. }
  463. for (i = bound; i < SBLIMIT; i++) {
  464. n = allocation[0][i];
  465. if (n) {
  466. mant = get_bits(&s->gb, n + 1);
  467. v = l1_unscale(n, mant, scale_factors[0][i]);
  468. s->sb_samples[0][j][i] = v;
  469. v = l1_unscale(n, mant, scale_factors[1][i]);
  470. s->sb_samples[1][j][i] = v;
  471. } else {
  472. s->sb_samples[0][j][i] = 0;
  473. s->sb_samples[1][j][i] = 0;
  474. }
  475. }
  476. }
  477. return 12;
  478. }
  479. static int mp_decode_layer2(MPADecodeContext *s)
  480. {
  481. int sblimit; /* number of used subbands */
  482. const unsigned char *alloc_table;
  483. int table, bit_alloc_bits, i, j, ch, bound, v;
  484. unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
  485. unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
  486. unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
  487. int scale, qindex, bits, steps, k, l, m, b;
  488. /* select decoding table */
  489. table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
  490. s->sample_rate, s->lsf);
  491. sblimit = ff_mpa_sblimit_table[table];
  492. alloc_table = ff_mpa_alloc_tables[table];
  493. if (s->mode == MPA_JSTEREO)
  494. bound = (s->mode_ext + 1) * 4;
  495. else
  496. bound = sblimit;
  497. ff_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
  498. /* sanity check */
  499. if (bound > sblimit)
  500. bound = sblimit;
  501. /* parse bit allocation */
  502. j = 0;
  503. for (i = 0; i < bound; i++) {
  504. bit_alloc_bits = alloc_table[j];
  505. for (ch = 0; ch < s->nb_channels; ch++)
  506. bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
  507. j += 1 << bit_alloc_bits;
  508. }
  509. for (i = bound; i < sblimit; i++) {
  510. bit_alloc_bits = alloc_table[j];
  511. v = get_bits(&s->gb, bit_alloc_bits);
  512. bit_alloc[0][i] = v;
  513. bit_alloc[1][i] = v;
  514. j += 1 << bit_alloc_bits;
  515. }
  516. /* scale codes */
  517. for (i = 0; i < sblimit; i++) {
  518. for (ch = 0; ch < s->nb_channels; ch++) {
  519. if (bit_alloc[ch][i])
  520. scale_code[ch][i] = get_bits(&s->gb, 2);
  521. }
  522. }
  523. /* scale factors */
  524. for (i = 0; i < sblimit; i++) {
  525. for (ch = 0; ch < s->nb_channels; ch++) {
  526. if (bit_alloc[ch][i]) {
  527. sf = scale_factors[ch][i];
  528. switch (scale_code[ch][i]) {
  529. default:
  530. case 0:
  531. sf[0] = get_bits(&s->gb, 6);
  532. sf[1] = get_bits(&s->gb, 6);
  533. sf[2] = get_bits(&s->gb, 6);
  534. break;
  535. case 2:
  536. sf[0] = get_bits(&s->gb, 6);
  537. sf[1] = sf[0];
  538. sf[2] = sf[0];
  539. break;
  540. case 1:
  541. sf[0] = get_bits(&s->gb, 6);
  542. sf[2] = get_bits(&s->gb, 6);
  543. sf[1] = sf[0];
  544. break;
  545. case 3:
  546. sf[0] = get_bits(&s->gb, 6);
  547. sf[2] = get_bits(&s->gb, 6);
  548. sf[1] = sf[2];
  549. break;
  550. }
  551. }
  552. }
  553. }
  554. /* samples */
  555. for (k = 0; k < 3; k++) {
  556. for (l = 0; l < 12; l += 3) {
  557. j = 0;
  558. for (i = 0; i < bound; i++) {
  559. bit_alloc_bits = alloc_table[j];
  560. for (ch = 0; ch < s->nb_channels; ch++) {
  561. b = bit_alloc[ch][i];
  562. if (b) {
  563. scale = scale_factors[ch][i][k];
  564. qindex = alloc_table[j+b];
  565. bits = ff_mpa_quant_bits[qindex];
  566. if (bits < 0) {
  567. int v2;
  568. /* 3 values at the same time */
  569. v = get_bits(&s->gb, -bits);
  570. v2 = division_tabs[qindex][v];
  571. steps = ff_mpa_quant_steps[qindex];
  572. s->sb_samples[ch][k * 12 + l + 0][i] =
  573. l2_unscale_group(steps, v2 & 15, scale);
  574. s->sb_samples[ch][k * 12 + l + 1][i] =
  575. l2_unscale_group(steps, (v2 >> 4) & 15, scale);
  576. s->sb_samples[ch][k * 12 + l + 2][i] =
  577. l2_unscale_group(steps, v2 >> 8 , scale);
  578. } else {
  579. for (m = 0; m < 3; m++) {
  580. v = get_bits(&s->gb, bits);
  581. v = l1_unscale(bits - 1, v, scale);
  582. s->sb_samples[ch][k * 12 + l + m][i] = v;
  583. }
  584. }
  585. } else {
  586. s->sb_samples[ch][k * 12 + l + 0][i] = 0;
  587. s->sb_samples[ch][k * 12 + l + 1][i] = 0;
  588. s->sb_samples[ch][k * 12 + l + 2][i] = 0;
  589. }
  590. }
  591. /* next subband in alloc table */
  592. j += 1 << bit_alloc_bits;
  593. }
  594. /* XXX: find a way to avoid this duplication of code */
  595. for (i = bound; i < sblimit; i++) {
  596. bit_alloc_bits = alloc_table[j];
  597. b = bit_alloc[0][i];
  598. if (b) {
  599. int mant, scale0, scale1;
  600. scale0 = scale_factors[0][i][k];
  601. scale1 = scale_factors[1][i][k];
  602. qindex = alloc_table[j+b];
  603. bits = ff_mpa_quant_bits[qindex];
  604. if (bits < 0) {
  605. /* 3 values at the same time */
  606. v = get_bits(&s->gb, -bits);
  607. steps = ff_mpa_quant_steps[qindex];
  608. mant = v % steps;
  609. v = v / steps;
  610. s->sb_samples[0][k * 12 + l + 0][i] =
  611. l2_unscale_group(steps, mant, scale0);
  612. s->sb_samples[1][k * 12 + l + 0][i] =
  613. l2_unscale_group(steps, mant, scale1);
  614. mant = v % steps;
  615. v = v / steps;
  616. s->sb_samples[0][k * 12 + l + 1][i] =
  617. l2_unscale_group(steps, mant, scale0);
  618. s->sb_samples[1][k * 12 + l + 1][i] =
  619. l2_unscale_group(steps, mant, scale1);
  620. s->sb_samples[0][k * 12 + l + 2][i] =
  621. l2_unscale_group(steps, v, scale0);
  622. s->sb_samples[1][k * 12 + l + 2][i] =
  623. l2_unscale_group(steps, v, scale1);
  624. } else {
  625. for (m = 0; m < 3; m++) {
  626. mant = get_bits(&s->gb, bits);
  627. s->sb_samples[0][k * 12 + l + m][i] =
  628. l1_unscale(bits - 1, mant, scale0);
  629. s->sb_samples[1][k * 12 + l + m][i] =
  630. l1_unscale(bits - 1, mant, scale1);
  631. }
  632. }
  633. } else {
  634. s->sb_samples[0][k * 12 + l + 0][i] = 0;
  635. s->sb_samples[0][k * 12 + l + 1][i] = 0;
  636. s->sb_samples[0][k * 12 + l + 2][i] = 0;
  637. s->sb_samples[1][k * 12 + l + 0][i] = 0;
  638. s->sb_samples[1][k * 12 + l + 1][i] = 0;
  639. s->sb_samples[1][k * 12 + l + 2][i] = 0;
  640. }
  641. /* next subband in alloc table */
  642. j += 1 << bit_alloc_bits;
  643. }
  644. /* fill remaining samples to zero */
  645. for (i = sblimit; i < SBLIMIT; i++) {
  646. for (ch = 0; ch < s->nb_channels; ch++) {
  647. s->sb_samples[ch][k * 12 + l + 0][i] = 0;
  648. s->sb_samples[ch][k * 12 + l + 1][i] = 0;
  649. s->sb_samples[ch][k * 12 + l + 2][i] = 0;
  650. }
  651. }
  652. }
  653. }
  654. return 3 * 12;
  655. }
  656. #define SPLIT(dst,sf,n) \
  657. if (n == 3) { \
  658. int m = (sf * 171) >> 9; \
  659. dst = sf - 3 * m; \
  660. sf = m; \
  661. } else if (n == 4) { \
  662. dst = sf & 3; \
  663. sf >>= 2; \
  664. } else if (n == 5) { \
  665. int m = (sf * 205) >> 10; \
  666. dst = sf - 5 * m; \
  667. sf = m; \
  668. } else if (n == 6) { \
  669. int m = (sf * 171) >> 10; \
  670. dst = sf - 6 * m; \
  671. sf = m; \
  672. } else { \
  673. dst = 0; \
  674. }
  675. static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2,
  676. int n3)
  677. {
  678. SPLIT(slen[3], sf, n3)
  679. SPLIT(slen[2], sf, n2)
  680. SPLIT(slen[1], sf, n1)
  681. slen[0] = sf;
  682. }
  683. static void exponents_from_scale_factors(MPADecodeContext *s, GranuleDef *g,
  684. int16_t *exponents)
  685. {
  686. const uint8_t *bstab, *pretab;
  687. int len, i, j, k, l, v0, shift, gain, gains[3];
  688. int16_t *exp_ptr;
  689. exp_ptr = exponents;
  690. gain = g->global_gain - 210;
  691. shift = g->scalefac_scale + 1;
  692. bstab = band_size_long[s->sample_rate_index];
  693. pretab = mpa_pretab[g->preflag];
  694. for (i = 0; i < g->long_end; i++) {
  695. v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
  696. len = bstab[i];
  697. for (j = len; j > 0; j--)
  698. *exp_ptr++ = v0;
  699. }
  700. if (g->short_start < 13) {
  701. bstab = band_size_short[s->sample_rate_index];
  702. gains[0] = gain - (g->subblock_gain[0] << 3);
  703. gains[1] = gain - (g->subblock_gain[1] << 3);
  704. gains[2] = gain - (g->subblock_gain[2] << 3);
  705. k = g->long_end;
  706. for (i = g->short_start; i < 13; i++) {
  707. len = bstab[i];
  708. for (l = 0; l < 3; l++) {
  709. v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
  710. for (j = len; j > 0; j--)
  711. *exp_ptr++ = v0;
  712. }
  713. }
  714. }
  715. }
  716. static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos,
  717. int *end_pos2)
  718. {
  719. if (s->in_gb.buffer && *pos >= s->gb.size_in_bits - s->extrasize * 8) {
  720. s->gb = s->in_gb;
  721. s->in_gb.buffer = NULL;
  722. s->extrasize = 0;
  723. assert((get_bits_count(&s->gb) & 7) == 0);
  724. skip_bits_long(&s->gb, *pos - *end_pos);
  725. *end_pos2 =
  726. *end_pos = *end_pos2 + get_bits_count(&s->gb) - *pos;
  727. *pos = get_bits_count(&s->gb);
  728. }
  729. }
  730. /* Following is a optimized code for
  731. INTFLOAT v = *src
  732. if(get_bits1(&s->gb))
  733. v = -v;
  734. *dst = v;
  735. */
  736. #if CONFIG_FLOAT
  737. #define READ_FLIP_SIGN(dst,src) \
  738. v = AV_RN32A(src) ^ (get_bits1(&s->gb) << 31); \
  739. AV_WN32A(dst, v);
  740. #else
  741. #define READ_FLIP_SIGN(dst,src) \
  742. v = -get_bits1(&s->gb); \
  743. *(dst) = (*(src) ^ v) - v;
  744. #endif
  745. static int huffman_decode(MPADecodeContext *s, GranuleDef *g,
  746. int16_t *exponents, int end_pos2)
  747. {
  748. int s_index;
  749. int i;
  750. int last_pos, bits_left;
  751. VLC *vlc;
  752. int end_pos = FFMIN(end_pos2, s->gb.size_in_bits - s->extrasize * 8);
  753. /* low frequencies (called big values) */
  754. s_index = 0;
  755. for (i = 0; i < 3; i++) {
  756. int j, k, l, linbits;
  757. j = g->region_size[i];
  758. if (j == 0)
  759. continue;
  760. /* select vlc table */
  761. k = g->table_select[i];
  762. l = mpa_huff_data[k][0];
  763. linbits = mpa_huff_data[k][1];
  764. vlc = &huff_vlc[l];
  765. if (!l) {
  766. memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * 2 * j);
  767. s_index += 2 * j;
  768. continue;
  769. }
  770. /* read huffcode and compute each couple */
  771. for (; j > 0; j--) {
  772. int exponent, x, y;
  773. int v;
  774. int pos = get_bits_count(&s->gb);
  775. if (pos >= end_pos){
  776. switch_buffer(s, &pos, &end_pos, &end_pos2);
  777. if (pos >= end_pos)
  778. break;
  779. }
  780. y = get_vlc2(&s->gb, vlc->table, 7, 3);
  781. if (!y) {
  782. g->sb_hybrid[s_index ] =
  783. g->sb_hybrid[s_index+1] = 0;
  784. s_index += 2;
  785. continue;
  786. }
  787. exponent= exponents[s_index];
  788. ff_dlog(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n",
  789. i, g->region_size[i] - j, x, y, exponent);
  790. if (y & 16) {
  791. x = y >> 5;
  792. y = y & 0x0f;
  793. if (x < 15) {
  794. READ_FLIP_SIGN(g->sb_hybrid + s_index, RENAME(expval_table)[exponent] + x)
  795. } else {
  796. x += get_bitsz(&s->gb, linbits);
  797. v = l3_unscale(x, exponent);
  798. if (get_bits1(&s->gb))
  799. v = -v;
  800. g->sb_hybrid[s_index] = v;
  801. }
  802. if (y < 15) {
  803. READ_FLIP_SIGN(g->sb_hybrid + s_index + 1, RENAME(expval_table)[exponent] + y)
  804. } else {
  805. y += get_bitsz(&s->gb, linbits);
  806. v = l3_unscale(y, exponent);
  807. if (get_bits1(&s->gb))
  808. v = -v;
  809. g->sb_hybrid[s_index+1] = v;
  810. }
  811. } else {
  812. x = y >> 5;
  813. y = y & 0x0f;
  814. x += y;
  815. if (x < 15) {
  816. READ_FLIP_SIGN(g->sb_hybrid + s_index + !!y, RENAME(expval_table)[exponent] + x)
  817. } else {
  818. x += get_bitsz(&s->gb, linbits);
  819. v = l3_unscale(x, exponent);
  820. if (get_bits1(&s->gb))
  821. v = -v;
  822. g->sb_hybrid[s_index+!!y] = v;
  823. }
  824. g->sb_hybrid[s_index + !y] = 0;
  825. }
  826. s_index += 2;
  827. }
  828. }
  829. /* high frequencies */
  830. vlc = &huff_quad_vlc[g->count1table_select];
  831. last_pos = 0;
  832. while (s_index <= 572) {
  833. int pos, code;
  834. pos = get_bits_count(&s->gb);
  835. if (pos >= end_pos) {
  836. if (pos > end_pos2 && last_pos) {
  837. /* some encoders generate an incorrect size for this
  838. part. We must go back into the data */
  839. s_index -= 4;
  840. skip_bits_long(&s->gb, last_pos - pos);
  841. av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
  842. if(s->err_recognition & AV_EF_BITSTREAM)
  843. s_index=0;
  844. break;
  845. }
  846. switch_buffer(s, &pos, &end_pos, &end_pos2);
  847. if (pos >= end_pos)
  848. break;
  849. }
  850. last_pos = pos;
  851. code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
  852. ff_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
  853. g->sb_hybrid[s_index+0] =
  854. g->sb_hybrid[s_index+1] =
  855. g->sb_hybrid[s_index+2] =
  856. g->sb_hybrid[s_index+3] = 0;
  857. while (code) {
  858. static const int idxtab[16] = { 3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0 };
  859. int v;
  860. int pos = s_index + idxtab[code];
  861. code ^= 8 >> idxtab[code];
  862. READ_FLIP_SIGN(g->sb_hybrid + pos, RENAME(exp_table)+exponents[pos])
  863. }
  864. s_index += 4;
  865. }
  866. /* skip extension bits */
  867. bits_left = end_pos2 - get_bits_count(&s->gb);
  868. if (bits_left < 0 && (s->err_recognition & AV_EF_BUFFER)) {
  869. av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
  870. s_index=0;
  871. } else if (bits_left > 0 && (s->err_recognition & AV_EF_BUFFER)) {
  872. av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
  873. s_index = 0;
  874. }
  875. memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * (576 - s_index));
  876. skip_bits_long(&s->gb, bits_left);
  877. i = get_bits_count(&s->gb);
  878. switch_buffer(s, &i, &end_pos, &end_pos2);
  879. return 0;
  880. }
  881. /* Reorder short blocks from bitstream order to interleaved order. It
  882. would be faster to do it in parsing, but the code would be far more
  883. complicated */
  884. static void reorder_block(MPADecodeContext *s, GranuleDef *g)
  885. {
  886. int i, j, len;
  887. INTFLOAT *ptr, *dst, *ptr1;
  888. INTFLOAT tmp[576];
  889. if (g->block_type != 2)
  890. return;
  891. if (g->switch_point) {
  892. if (s->sample_rate_index != 8)
  893. ptr = g->sb_hybrid + 36;
  894. else
  895. ptr = g->sb_hybrid + 72;
  896. } else {
  897. ptr = g->sb_hybrid;
  898. }
  899. for (i = g->short_start; i < 13; i++) {
  900. len = band_size_short[s->sample_rate_index][i];
  901. ptr1 = ptr;
  902. dst = tmp;
  903. for (j = len; j > 0; j--) {
  904. *dst++ = ptr[0*len];
  905. *dst++ = ptr[1*len];
  906. *dst++ = ptr[2*len];
  907. ptr++;
  908. }
  909. ptr += 2 * len;
  910. memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
  911. }
  912. }
  913. #define ISQRT2 FIXR(0.70710678118654752440)
  914. static void compute_stereo(MPADecodeContext *s, GranuleDef *g0, GranuleDef *g1)
  915. {
  916. int i, j, k, l;
  917. int sf_max, sf, len, non_zero_found;
  918. INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;
  919. int non_zero_found_short[3];
  920. /* intensity stereo */
  921. if (s->mode_ext & MODE_EXT_I_STEREO) {
  922. if (!s->lsf) {
  923. is_tab = is_table;
  924. sf_max = 7;
  925. } else {
  926. is_tab = is_table_lsf[g1->scalefac_compress & 1];
  927. sf_max = 16;
  928. }
  929. tab0 = g0->sb_hybrid + 576;
  930. tab1 = g1->sb_hybrid + 576;
  931. non_zero_found_short[0] = 0;
  932. non_zero_found_short[1] = 0;
  933. non_zero_found_short[2] = 0;
  934. k = (13 - g1->short_start) * 3 + g1->long_end - 3;
  935. for (i = 12; i >= g1->short_start; i--) {
  936. /* for last band, use previous scale factor */
  937. if (i != 11)
  938. k -= 3;
  939. len = band_size_short[s->sample_rate_index][i];
  940. for (l = 2; l >= 0; l--) {
  941. tab0 -= len;
  942. tab1 -= len;
  943. if (!non_zero_found_short[l]) {
  944. /* test if non zero band. if so, stop doing i-stereo */
  945. for (j = 0; j < len; j++) {
  946. if (tab1[j] != 0) {
  947. non_zero_found_short[l] = 1;
  948. goto found1;
  949. }
  950. }
  951. sf = g1->scale_factors[k + l];
  952. if (sf >= sf_max)
  953. goto found1;
  954. v1 = is_tab[0][sf];
  955. v2 = is_tab[1][sf];
  956. for (j = 0; j < len; j++) {
  957. tmp0 = tab0[j];
  958. tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
  959. tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
  960. }
  961. } else {
  962. found1:
  963. if (s->mode_ext & MODE_EXT_MS_STEREO) {
  964. /* lower part of the spectrum : do ms stereo
  965. if enabled */
  966. for (j = 0; j < len; j++) {
  967. tmp0 = tab0[j];
  968. tmp1 = tab1[j];
  969. tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
  970. tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
  971. }
  972. }
  973. }
  974. }
  975. }
  976. non_zero_found = non_zero_found_short[0] |
  977. non_zero_found_short[1] |
  978. non_zero_found_short[2];
  979. for (i = g1->long_end - 1;i >= 0;i--) {
  980. len = band_size_long[s->sample_rate_index][i];
  981. tab0 -= len;
  982. tab1 -= len;
  983. /* test if non zero band. if so, stop doing i-stereo */
  984. if (!non_zero_found) {
  985. for (j = 0; j < len; j++) {
  986. if (tab1[j] != 0) {
  987. non_zero_found = 1;
  988. goto found2;
  989. }
  990. }
  991. /* for last band, use previous scale factor */
  992. k = (i == 21) ? 20 : i;
  993. sf = g1->scale_factors[k];
  994. if (sf >= sf_max)
  995. goto found2;
  996. v1 = is_tab[0][sf];
  997. v2 = is_tab[1][sf];
  998. for (j = 0; j < len; j++) {
  999. tmp0 = tab0[j];
  1000. tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
  1001. tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
  1002. }
  1003. } else {
  1004. found2:
  1005. if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1006. /* lower part of the spectrum : do ms stereo
  1007. if enabled */
  1008. for (j = 0; j < len; j++) {
  1009. tmp0 = tab0[j];
  1010. tmp1 = tab1[j];
  1011. tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
  1012. tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
  1013. }
  1014. }
  1015. }
  1016. }
  1017. } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1018. /* ms stereo ONLY */
  1019. /* NOTE: the 1/sqrt(2) normalization factor is included in the
  1020. global gain */
  1021. #if CONFIG_FLOAT
  1022. s->fdsp.butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);
  1023. #else
  1024. tab0 = g0->sb_hybrid;
  1025. tab1 = g1->sb_hybrid;
  1026. for (i = 0; i < 576; i++) {
  1027. tmp0 = tab0[i];
  1028. tmp1 = tab1[i];
  1029. tab0[i] = tmp0 + tmp1;
  1030. tab1[i] = tmp0 - tmp1;
  1031. }
  1032. #endif
  1033. }
  1034. }
  1035. #if CONFIG_FLOAT
  1036. #define AA(j) do { \
  1037. float tmp0 = ptr[-1-j]; \
  1038. float tmp1 = ptr[ j]; \
  1039. ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1]; \
  1040. ptr[ j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0]; \
  1041. } while (0)
  1042. #else
  1043. #define AA(j) do { \
  1044. int tmp0 = ptr[-1-j]; \
  1045. int tmp1 = ptr[ j]; \
  1046. int tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]); \
  1047. ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2])); \
  1048. ptr[ j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3])); \
  1049. } while (0)
  1050. #endif
  1051. static void compute_antialias(MPADecodeContext *s, GranuleDef *g)
  1052. {
  1053. INTFLOAT *ptr;
  1054. int n, i;
  1055. /* we antialias only "long" bands */
  1056. if (g->block_type == 2) {
  1057. if (!g->switch_point)
  1058. return;
  1059. /* XXX: check this for 8000Hz case */
  1060. n = 1;
  1061. } else {
  1062. n = SBLIMIT - 1;
  1063. }
  1064. ptr = g->sb_hybrid + 18;
  1065. for (i = n; i > 0; i--) {
  1066. AA(0);
  1067. AA(1);
  1068. AA(2);
  1069. AA(3);
  1070. AA(4);
  1071. AA(5);
  1072. AA(6);
  1073. AA(7);
  1074. ptr += 18;
  1075. }
  1076. }
  1077. static void compute_imdct(MPADecodeContext *s, GranuleDef *g,
  1078. INTFLOAT *sb_samples, INTFLOAT *mdct_buf)
  1079. {
  1080. INTFLOAT *win, *out_ptr, *ptr, *buf, *ptr1;
  1081. INTFLOAT out2[12];
  1082. int i, j, mdct_long_end, sblimit;
  1083. /* find last non zero block */
  1084. ptr = g->sb_hybrid + 576;
  1085. ptr1 = g->sb_hybrid + 2 * 18;
  1086. while (ptr >= ptr1) {
  1087. int32_t *p;
  1088. ptr -= 6;
  1089. p = (int32_t*)ptr;
  1090. if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
  1091. break;
  1092. }
  1093. sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
  1094. if (g->block_type == 2) {
  1095. /* XXX: check for 8000 Hz */
  1096. if (g->switch_point)
  1097. mdct_long_end = 2;
  1098. else
  1099. mdct_long_end = 0;
  1100. } else {
  1101. mdct_long_end = sblimit;
  1102. }
  1103. s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid,
  1104. mdct_long_end, g->switch_point,
  1105. g->block_type);
  1106. buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3);
  1107. ptr = g->sb_hybrid + 18 * mdct_long_end;
  1108. for (j = mdct_long_end; j < sblimit; j++) {
  1109. /* select frequency inversion */
  1110. win = RENAME(ff_mdct_win)[2 + (4 & -(j & 1))];
  1111. out_ptr = sb_samples + j;
  1112. for (i = 0; i < 6; i++) {
  1113. *out_ptr = buf[4*i];
  1114. out_ptr += SBLIMIT;
  1115. }
  1116. imdct12(out2, ptr + 0);
  1117. for (i = 0; i < 6; i++) {
  1118. *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*1)];
  1119. buf[4*(i + 6*2)] = MULH3(out2[i + 6], win[i + 6], 1);
  1120. out_ptr += SBLIMIT;
  1121. }
  1122. imdct12(out2, ptr + 1);
  1123. for (i = 0; i < 6; i++) {
  1124. *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*2)];
  1125. buf[4*(i + 6*0)] = MULH3(out2[i + 6], win[i + 6], 1);
  1126. out_ptr += SBLIMIT;
  1127. }
  1128. imdct12(out2, ptr + 2);
  1129. for (i = 0; i < 6; i++) {
  1130. buf[4*(i + 6*0)] = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*0)];
  1131. buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1);
  1132. buf[4*(i + 6*2)] = 0;
  1133. }
  1134. ptr += 18;
  1135. buf += (j&3) != 3 ? 1 : (4*18-3);
  1136. }
  1137. /* zero bands */
  1138. for (j = sblimit; j < SBLIMIT; j++) {
  1139. /* overlap */
  1140. out_ptr = sb_samples + j;
  1141. for (i = 0; i < 18; i++) {
  1142. *out_ptr = buf[4*i];
  1143. buf[4*i] = 0;
  1144. out_ptr += SBLIMIT;
  1145. }
  1146. buf += (j&3) != 3 ? 1 : (4*18-3);
  1147. }
  1148. }
  1149. /* main layer3 decoding function */
  1150. static int mp_decode_layer3(MPADecodeContext *s)
  1151. {
  1152. int nb_granules, main_data_begin;
  1153. int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
  1154. GranuleDef *g;
  1155. int16_t exponents[576]; //FIXME try INTFLOAT
  1156. /* read side info */
  1157. if (s->lsf) {
  1158. main_data_begin = get_bits(&s->gb, 8);
  1159. skip_bits(&s->gb, s->nb_channels);
  1160. nb_granules = 1;
  1161. } else {
  1162. main_data_begin = get_bits(&s->gb, 9);
  1163. if (s->nb_channels == 2)
  1164. skip_bits(&s->gb, 3);
  1165. else
  1166. skip_bits(&s->gb, 5);
  1167. nb_granules = 2;
  1168. for (ch = 0; ch < s->nb_channels; ch++) {
  1169. s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
  1170. s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
  1171. }
  1172. }
  1173. for (gr = 0; gr < nb_granules; gr++) {
  1174. for (ch = 0; ch < s->nb_channels; ch++) {
  1175. ff_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
  1176. g = &s->granules[ch][gr];
  1177. g->part2_3_length = get_bits(&s->gb, 12);
  1178. g->big_values = get_bits(&s->gb, 9);
  1179. if (g->big_values > 288) {
  1180. av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
  1181. return AVERROR_INVALIDDATA;
  1182. }
  1183. g->global_gain = get_bits(&s->gb, 8);
  1184. /* if MS stereo only is selected, we precompute the
  1185. 1/sqrt(2) renormalization factor */
  1186. if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
  1187. MODE_EXT_MS_STEREO)
  1188. g->global_gain -= 2;
  1189. if (s->lsf)
  1190. g->scalefac_compress = get_bits(&s->gb, 9);
  1191. else
  1192. g->scalefac_compress = get_bits(&s->gb, 4);
  1193. blocksplit_flag = get_bits1(&s->gb);
  1194. if (blocksplit_flag) {
  1195. g->block_type = get_bits(&s->gb, 2);
  1196. if (g->block_type == 0) {
  1197. av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
  1198. return AVERROR_INVALIDDATA;
  1199. }
  1200. g->switch_point = get_bits1(&s->gb);
  1201. for (i = 0; i < 2; i++)
  1202. g->table_select[i] = get_bits(&s->gb, 5);
  1203. for (i = 0; i < 3; i++)
  1204. g->subblock_gain[i] = get_bits(&s->gb, 3);
  1205. init_short_region(s, g);
  1206. } else {
  1207. int region_address1, region_address2;
  1208. g->block_type = 0;
  1209. g->switch_point = 0;
  1210. for (i = 0; i < 3; i++)
  1211. g->table_select[i] = get_bits(&s->gb, 5);
  1212. /* compute huffman coded region sizes */
  1213. region_address1 = get_bits(&s->gb, 4);
  1214. region_address2 = get_bits(&s->gb, 3);
  1215. ff_dlog(s->avctx, "region1=%d region2=%d\n",
  1216. region_address1, region_address2);
  1217. init_long_region(s, g, region_address1, region_address2);
  1218. }
  1219. region_offset2size(g);
  1220. compute_band_indexes(s, g);
  1221. g->preflag = 0;
  1222. if (!s->lsf)
  1223. g->preflag = get_bits1(&s->gb);
  1224. g->scalefac_scale = get_bits1(&s->gb);
  1225. g->count1table_select = get_bits1(&s->gb);
  1226. ff_dlog(s->avctx, "block_type=%d switch_point=%d\n",
  1227. g->block_type, g->switch_point);
  1228. }
  1229. }
  1230. if (!s->adu_mode) {
  1231. int skip;
  1232. const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
  1233. s->extrasize = av_clip((get_bits_left(&s->gb) >> 3) - s->extrasize, 0,
  1234. FFMAX(0, LAST_BUF_SIZE - s->last_buf_size));
  1235. assert((get_bits_count(&s->gb) & 7) == 0);
  1236. /* now we get bits from the main_data_begin offset */
  1237. ff_dlog(s->avctx, "seekback:%d, lastbuf:%d\n",
  1238. main_data_begin, s->last_buf_size);
  1239. memcpy(s->last_buf + s->last_buf_size, ptr, s->extrasize);
  1240. s->in_gb = s->gb;
  1241. init_get_bits(&s->gb, s->last_buf, (s->last_buf_size + s->extrasize) * 8);
  1242. s->last_buf_size <<= 3;
  1243. for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) {
  1244. for (ch = 0; ch < s->nb_channels; ch++) {
  1245. g = &s->granules[ch][gr];
  1246. s->last_buf_size += g->part2_3_length;
  1247. memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
  1248. compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
  1249. }
  1250. }
  1251. skip = s->last_buf_size - 8 * main_data_begin;
  1252. if (skip >= s->gb.size_in_bits - s->extrasize * 8 && s->in_gb.buffer) {
  1253. skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits + s->extrasize * 8);
  1254. s->gb = s->in_gb;
  1255. s->in_gb.buffer = NULL;
  1256. s->extrasize = 0;
  1257. } else {
  1258. skip_bits_long(&s->gb, skip);
  1259. }
  1260. } else {
  1261. gr = 0;
  1262. s->extrasize = 0;
  1263. }
  1264. for (; gr < nb_granules; gr++) {
  1265. for (ch = 0; ch < s->nb_channels; ch++) {
  1266. g = &s->granules[ch][gr];
  1267. bits_pos = get_bits_count(&s->gb);
  1268. if (!s->lsf) {
  1269. uint8_t *sc;
  1270. int slen, slen1, slen2;
  1271. /* MPEG-1 scale factors */
  1272. slen1 = slen_table[0][g->scalefac_compress];
  1273. slen2 = slen_table[1][g->scalefac_compress];
  1274. ff_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
  1275. if (g->block_type == 2) {
  1276. n = g->switch_point ? 17 : 18;
  1277. j = 0;
  1278. if (slen1) {
  1279. for (i = 0; i < n; i++)
  1280. g->scale_factors[j++] = get_bits(&s->gb, slen1);
  1281. } else {
  1282. for (i = 0; i < n; i++)
  1283. g->scale_factors[j++] = 0;
  1284. }
  1285. if (slen2) {
  1286. for (i = 0; i < 18; i++)
  1287. g->scale_factors[j++] = get_bits(&s->gb, slen2);
  1288. for (i = 0; i < 3; i++)
  1289. g->scale_factors[j++] = 0;
  1290. } else {
  1291. for (i = 0; i < 21; i++)
  1292. g->scale_factors[j++] = 0;
  1293. }
  1294. } else {
  1295. sc = s->granules[ch][0].scale_factors;
  1296. j = 0;
  1297. for (k = 0; k < 4; k++) {
  1298. n = k == 0 ? 6 : 5;
  1299. if ((g->scfsi & (0x8 >> k)) == 0) {
  1300. slen = (k < 2) ? slen1 : slen2;
  1301. if (slen) {
  1302. for (i = 0; i < n; i++)
  1303. g->scale_factors[j++] = get_bits(&s->gb, slen);
  1304. } else {
  1305. for (i = 0; i < n; i++)
  1306. g->scale_factors[j++] = 0;
  1307. }
  1308. } else {
  1309. /* simply copy from last granule */
  1310. for (i = 0; i < n; i++) {
  1311. g->scale_factors[j] = sc[j];
  1312. j++;
  1313. }
  1314. }
  1315. }
  1316. g->scale_factors[j++] = 0;
  1317. }
  1318. } else {
  1319. int tindex, tindex2, slen[4], sl, sf;
  1320. /* LSF scale factors */
  1321. if (g->block_type == 2)
  1322. tindex = g->switch_point ? 2 : 1;
  1323. else
  1324. tindex = 0;
  1325. sf = g->scalefac_compress;
  1326. if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
  1327. /* intensity stereo case */
  1328. sf >>= 1;
  1329. if (sf < 180) {
  1330. lsf_sf_expand(slen, sf, 6, 6, 0);
  1331. tindex2 = 3;
  1332. } else if (sf < 244) {
  1333. lsf_sf_expand(slen, sf - 180, 4, 4, 0);
  1334. tindex2 = 4;
  1335. } else {
  1336. lsf_sf_expand(slen, sf - 244, 3, 0, 0);
  1337. tindex2 = 5;
  1338. }
  1339. } else {
  1340. /* normal case */
  1341. if (sf < 400) {
  1342. lsf_sf_expand(slen, sf, 5, 4, 4);
  1343. tindex2 = 0;
  1344. } else if (sf < 500) {
  1345. lsf_sf_expand(slen, sf - 400, 5, 4, 0);
  1346. tindex2 = 1;
  1347. } else {
  1348. lsf_sf_expand(slen, sf - 500, 3, 0, 0);
  1349. tindex2 = 2;
  1350. g->preflag = 1;
  1351. }
  1352. }
  1353. j = 0;
  1354. for (k = 0; k < 4; k++) {
  1355. n = lsf_nsf_table[tindex2][tindex][k];
  1356. sl = slen[k];
  1357. if (sl) {
  1358. for (i = 0; i < n; i++)
  1359. g->scale_factors[j++] = get_bits(&s->gb, sl);
  1360. } else {
  1361. for (i = 0; i < n; i++)
  1362. g->scale_factors[j++] = 0;
  1363. }
  1364. }
  1365. /* XXX: should compute exact size */
  1366. for (; j < 40; j++)
  1367. g->scale_factors[j] = 0;
  1368. }
  1369. exponents_from_scale_factors(s, g, exponents);
  1370. /* read Huffman coded residue */
  1371. huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
  1372. } /* ch */
  1373. if (s->mode == MPA_JSTEREO)
  1374. compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
  1375. for (ch = 0; ch < s->nb_channels; ch++) {
  1376. g = &s->granules[ch][gr];
  1377. reorder_block(s, g);
  1378. compute_antialias(s, g);
  1379. compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
  1380. }
  1381. } /* gr */
  1382. if (get_bits_count(&s->gb) < 0)
  1383. skip_bits_long(&s->gb, -get_bits_count(&s->gb));
  1384. return nb_granules * 18;
  1385. }
  1386. static int mp_decode_frame(MPADecodeContext *s, OUT_INT **samples,
  1387. const uint8_t *buf, int buf_size)
  1388. {
  1389. int i, nb_frames, ch, ret;
  1390. OUT_INT *samples_ptr;
  1391. init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);
  1392. /* skip error protection field */
  1393. if (s->error_protection)
  1394. skip_bits(&s->gb, 16);
  1395. switch(s->layer) {
  1396. case 1:
  1397. s->avctx->frame_size = 384;
  1398. nb_frames = mp_decode_layer1(s);
  1399. break;
  1400. case 2:
  1401. s->avctx->frame_size = 1152;
  1402. nb_frames = mp_decode_layer2(s);
  1403. break;
  1404. case 3:
  1405. s->avctx->frame_size = s->lsf ? 576 : 1152;
  1406. default:
  1407. nb_frames = mp_decode_layer3(s);
  1408. if (nb_frames < 0)
  1409. return nb_frames;
  1410. s->last_buf_size=0;
  1411. if (s->in_gb.buffer) {
  1412. align_get_bits(&s->gb);
  1413. i = (get_bits_left(&s->gb) >> 3) - s->extrasize;
  1414. if (i >= 0 && i <= BACKSTEP_SIZE) {
  1415. memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);
  1416. s->last_buf_size=i;
  1417. } else
  1418. av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
  1419. s->gb = s->in_gb;
  1420. s->in_gb.buffer = NULL;
  1421. s->extrasize = 0;
  1422. }
  1423. align_get_bits(&s->gb);
  1424. assert((get_bits_count(&s->gb) & 7) == 0);
  1425. i = (get_bits_left(&s->gb) >> 3) - s->extrasize;
  1426. if (i < 0 || i > BACKSTEP_SIZE || nb_frames < 0) {
  1427. if (i < 0)
  1428. av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
  1429. i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
  1430. }
  1431. assert(i <= buf_size - HEADER_SIZE && i >= 0);
  1432. memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
  1433. s->last_buf_size += i;
  1434. }
  1435. /* get output buffer */
  1436. if (!samples) {
  1437. av_assert0(s->frame != NULL);
  1438. s->frame->nb_samples = s->avctx->frame_size;
  1439. if ((ret = ff_get_buffer(s->avctx, s->frame, 0)) < 0) {
  1440. av_log(s->avctx, AV_LOG_ERROR, "get_buffer() failed\n");
  1441. return ret;
  1442. }
  1443. samples = (OUT_INT **)s->frame->extended_data;
  1444. }
  1445. /* apply the synthesis filter */
  1446. for (ch = 0; ch < s->nb_channels; ch++) {
  1447. int sample_stride;
  1448. if (s->avctx->sample_fmt == OUT_FMT_P) {
  1449. samples_ptr = samples[ch];
  1450. sample_stride = 1;
  1451. } else {
  1452. samples_ptr = samples[0] + ch;
  1453. sample_stride = s->nb_channels;
  1454. }
  1455. for (i = 0; i < nb_frames; i++) {
  1456. RENAME(ff_mpa_synth_filter)(&s->mpadsp, s->synth_buf[ch],
  1457. &(s->synth_buf_offset[ch]),
  1458. RENAME(ff_mpa_synth_window),
  1459. &s->dither_state, samples_ptr,
  1460. sample_stride, s->sb_samples[ch][i]);
  1461. samples_ptr += 32 * sample_stride;
  1462. }
  1463. }
  1464. return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
  1465. }
  1466. static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,
  1467. AVPacket *avpkt)
  1468. {
  1469. const uint8_t *buf = avpkt->data;
  1470. int buf_size = avpkt->size;
  1471. MPADecodeContext *s = avctx->priv_data;
  1472. uint32_t header;
  1473. int ret;
  1474. if (buf_size < HEADER_SIZE)
  1475. return AVERROR_INVALIDDATA;
  1476. header = AV_RB32(buf);
  1477. ret = avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);
  1478. if (ret < 0) {
  1479. av_log(avctx, AV_LOG_ERROR, "Header missing\n");
  1480. return AVERROR_INVALIDDATA;
  1481. } else if (ret == 1) {
  1482. /* free format: prepare to compute frame size */
  1483. s->frame_size = -1;
  1484. return AVERROR_INVALIDDATA;
  1485. }
  1486. /* update codec info */
  1487. avctx->channels = s->nb_channels;
  1488. avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
  1489. if (!avctx->bit_rate)
  1490. avctx->bit_rate = s->bit_rate;
  1491. s->frame = data;
  1492. ret = mp_decode_frame(s, NULL, buf, buf_size);
  1493. if (ret >= 0) {
  1494. s->frame->nb_samples = avctx->frame_size;
  1495. *got_frame_ptr = 1;
  1496. avctx->sample_rate = s->sample_rate;
  1497. //FIXME maybe move the other codec info stuff from above here too
  1498. } else {
  1499. av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
  1500. /* Only return an error if the bad frame makes up the whole packet or
  1501. * the error is related to buffer management.
  1502. * If there is more data in the packet, just consume the bad frame
  1503. * instead of returning an error, which would discard the whole
  1504. * packet. */
  1505. *got_frame_ptr = 0;
  1506. if (buf_size == avpkt->size || ret != AVERROR_INVALIDDATA)
  1507. return ret;
  1508. }
  1509. s->frame_size = 0;
  1510. return buf_size;
  1511. }
  1512. static void mp_flush(MPADecodeContext *ctx)
  1513. {
  1514. memset(ctx->synth_buf, 0, sizeof(ctx->synth_buf));
  1515. ctx->last_buf_size = 0;
  1516. }
  1517. static void flush(AVCodecContext *avctx)
  1518. {
  1519. mp_flush(avctx->priv_data);
  1520. }
  1521. #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
  1522. static int decode_frame_adu(AVCodecContext *avctx, void *data,
  1523. int *got_frame_ptr, AVPacket *avpkt)
  1524. {
  1525. const uint8_t *buf = avpkt->data;
  1526. int buf_size = avpkt->size;
  1527. MPADecodeContext *s = avctx->priv_data;
  1528. uint32_t header;
  1529. int len, ret;
  1530. len = buf_size;
  1531. // Discard too short frames
  1532. if (buf_size < HEADER_SIZE) {
  1533. av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
  1534. return AVERROR_INVALIDDATA;
  1535. }
  1536. if (len > MPA_MAX_CODED_FRAME_SIZE)
  1537. len = MPA_MAX_CODED_FRAME_SIZE;
  1538. // Get header and restore sync word
  1539. header = AV_RB32(buf) | 0xffe00000;
  1540. ret = avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);
  1541. if (ret < 0) {
  1542. av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n");
  1543. return ret;
  1544. }
  1545. /* update codec info */
  1546. avctx->sample_rate = s->sample_rate;
  1547. avctx->channels = s->nb_channels;
  1548. avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
  1549. if (!avctx->bit_rate)
  1550. avctx->bit_rate = s->bit_rate;
  1551. s->frame_size = len;
  1552. s->frame = data;
  1553. ret = mp_decode_frame(s, NULL, buf, buf_size);
  1554. if (ret < 0) {
  1555. av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
  1556. return ret;
  1557. }
  1558. *got_frame_ptr = 1;
  1559. return buf_size;
  1560. }
  1561. #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
  1562. #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
  1563. /**
  1564. * Context for MP3On4 decoder
  1565. */
  1566. typedef struct MP3On4DecodeContext {
  1567. int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
  1568. int syncword; ///< syncword patch
  1569. const uint8_t *coff; ///< channel offsets in output buffer
  1570. MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
  1571. } MP3On4DecodeContext;
  1572. #include "mpeg4audio.h"
  1573. /* Next 3 arrays are indexed by channel config number (passed via codecdata) */
  1574. /* number of mp3 decoder instances */
  1575. static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 };
  1576. /* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */
  1577. static const uint8_t chan_offset[8][5] = {
  1578. { 0 },
  1579. { 0 }, // C
  1580. { 0 }, // FLR
  1581. { 2, 0 }, // C FLR
  1582. { 2, 0, 3 }, // C FLR BS
  1583. { 2, 0, 3 }, // C FLR BLRS
  1584. { 2, 0, 4, 3 }, // C FLR BLRS LFE
  1585. { 2, 0, 6, 4, 3 }, // C FLR BLRS BLR LFE
  1586. };
  1587. /* mp3on4 channel layouts */
  1588. static const int16_t chan_layout[8] = {
  1589. 0,
  1590. AV_CH_LAYOUT_MONO,
  1591. AV_CH_LAYOUT_STEREO,
  1592. AV_CH_LAYOUT_SURROUND,
  1593. AV_CH_LAYOUT_4POINT0,
  1594. AV_CH_LAYOUT_5POINT0,
  1595. AV_CH_LAYOUT_5POINT1,
  1596. AV_CH_LAYOUT_7POINT1
  1597. };
  1598. static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
  1599. {
  1600. MP3On4DecodeContext *s = avctx->priv_data;
  1601. int i;
  1602. for (i = 0; i < s->frames; i++)
  1603. av_free(s->mp3decctx[i]);
  1604. return 0;
  1605. }
  1606. static av_cold int decode_init_mp3on4(AVCodecContext * avctx)
  1607. {
  1608. MP3On4DecodeContext *s = avctx->priv_data;
  1609. MPEG4AudioConfig cfg;
  1610. int i;
  1611. if ((avctx->extradata_size < 2) || !avctx->extradata) {
  1612. av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
  1613. return AVERROR_INVALIDDATA;
  1614. }
  1615. avpriv_mpeg4audio_get_config(&cfg, avctx->extradata,
  1616. avctx->extradata_size * 8, 1);
  1617. if (!cfg.chan_config || cfg.chan_config > 7) {
  1618. av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
  1619. return AVERROR_INVALIDDATA;
  1620. }
  1621. s->frames = mp3Frames[cfg.chan_config];
  1622. s->coff = chan_offset[cfg.chan_config];
  1623. avctx->channels = ff_mpeg4audio_channels[cfg.chan_config];
  1624. avctx->channel_layout = chan_layout[cfg.chan_config];
  1625. if (cfg.sample_rate < 16000)
  1626. s->syncword = 0xffe00000;
  1627. else
  1628. s->syncword = 0xfff00000;
  1629. /* Init the first mp3 decoder in standard way, so that all tables get builded
  1630. * We replace avctx->priv_data with the context of the first decoder so that
  1631. * decode_init() does not have to be changed.
  1632. * Other decoders will be initialized here copying data from the first context
  1633. */
  1634. // Allocate zeroed memory for the first decoder context
  1635. s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
  1636. if (!s->mp3decctx[0])
  1637. goto alloc_fail;
  1638. // Put decoder context in place to make init_decode() happy
  1639. avctx->priv_data = s->mp3decctx[0];
  1640. decode_init(avctx);
  1641. // Restore mp3on4 context pointer
  1642. avctx->priv_data = s;
  1643. s->mp3decctx[0]->adu_mode = 1; // Set adu mode
  1644. /* Create a separate codec/context for each frame (first is already ok).
  1645. * Each frame is 1 or 2 channels - up to 5 frames allowed
  1646. */
  1647. for (i = 1; i < s->frames; i++) {
  1648. s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
  1649. if (!s->mp3decctx[i])
  1650. goto alloc_fail;
  1651. s->mp3decctx[i]->adu_mode = 1;
  1652. s->mp3decctx[i]->avctx = avctx;
  1653. s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;
  1654. }
  1655. return 0;
  1656. alloc_fail:
  1657. decode_close_mp3on4(avctx);
  1658. return AVERROR(ENOMEM);
  1659. }
  1660. static void flush_mp3on4(AVCodecContext *avctx)
  1661. {
  1662. int i;
  1663. MP3On4DecodeContext *s = avctx->priv_data;
  1664. for (i = 0; i < s->frames; i++)
  1665. mp_flush(s->mp3decctx[i]);
  1666. }
  1667. static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
  1668. int *got_frame_ptr, AVPacket *avpkt)
  1669. {
  1670. AVFrame *frame = data;
  1671. const uint8_t *buf = avpkt->data;
  1672. int buf_size = avpkt->size;
  1673. MP3On4DecodeContext *s = avctx->priv_data;
  1674. MPADecodeContext *m;
  1675. int fsize, len = buf_size, out_size = 0;
  1676. uint32_t header;
  1677. OUT_INT **out_samples;
  1678. OUT_INT *outptr[2];
  1679. int fr, ch, ret;
  1680. /* get output buffer */
  1681. frame->nb_samples = MPA_FRAME_SIZE;
  1682. if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
  1683. av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
  1684. return ret;
  1685. }
  1686. out_samples = (OUT_INT **)frame->extended_data;
  1687. // Discard too short frames
  1688. if (buf_size < HEADER_SIZE)
  1689. return AVERROR_INVALIDDATA;
  1690. avctx->bit_rate = 0;
  1691. ch = 0;
  1692. for (fr = 0; fr < s->frames; fr++) {
  1693. fsize = AV_RB16(buf) >> 4;
  1694. fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
  1695. m = s->mp3decctx[fr];
  1696. assert(m != NULL);
  1697. if (fsize < HEADER_SIZE) {
  1698. av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n");
  1699. return AVERROR_INVALIDDATA;
  1700. }
  1701. header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
  1702. ret = avpriv_mpegaudio_decode_header((MPADecodeHeader *)m, header);
  1703. if (ret < 0) // Bad header, discard block
  1704. break;
  1705. if (ch + m->nb_channels > avctx->channels ||
  1706. s->coff[fr] + m->nb_channels > avctx->channels) {
  1707. av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "
  1708. "channel count\n");
  1709. return AVERROR_INVALIDDATA;
  1710. }
  1711. ch += m->nb_channels;
  1712. outptr[0] = out_samples[s->coff[fr]];
  1713. if (m->nb_channels > 1)
  1714. outptr[1] = out_samples[s->coff[fr] + 1];
  1715. if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0)
  1716. return ret;
  1717. out_size += ret;
  1718. buf += fsize;
  1719. len -= fsize;
  1720. avctx->bit_rate += m->bit_rate;
  1721. }
  1722. /* update codec info */
  1723. avctx->sample_rate = s->mp3decctx[0]->sample_rate;
  1724. frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));
  1725. *got_frame_ptr = 1;
  1726. return buf_size;
  1727. }
  1728. #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */