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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/channel_layout.h"
  26. #include "avcodec.h"
  27. #include "get_bits.h"
  28. #include "mathops.h"
  29. #include "mpegaudiodsp.h"
  30. #include "dsputil.h"
  31. /*
  32. * TODO:
  33. * - test lsf / mpeg25 extensively.
  34. */
  35. #include "mpegaudio.h"
  36. #include "mpegaudiodecheader.h"
  37. #define BACKSTEP_SIZE 512
  38. #define EXTRABYTES 24
  39. #define LAST_BUF_SIZE 2 * BACKSTEP_SIZE + EXTRABYTES
  40. /* layer 3 "granule" */
  41. typedef struct GranuleDef {
  42. uint8_t scfsi;
  43. int part2_3_length;
  44. int big_values;
  45. int global_gain;
  46. int scalefac_compress;
  47. uint8_t block_type;
  48. uint8_t switch_point;
  49. int table_select[3];
  50. int subblock_gain[3];
  51. uint8_t scalefac_scale;
  52. uint8_t count1table_select;
  53. int region_size[3]; /* number of huffman codes in each region */
  54. int preflag;
  55. int short_start, long_end; /* long/short band indexes */
  56. uint8_t scale_factors[40];
  57. DECLARE_ALIGNED(16, INTFLOAT, sb_hybrid)[SBLIMIT * 18]; /* 576 samples */
  58. } GranuleDef;
  59. typedef struct MPADecodeContext {
  60. MPA_DECODE_HEADER
  61. uint8_t last_buf[LAST_BUF_SIZE];
  62. int last_buf_size;
  63. /* next header (used in free format parsing) */
  64. uint32_t free_format_next_header;
  65. GetBitContext gb;
  66. GetBitContext in_gb;
  67. DECLARE_ALIGNED(32, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512 * 2];
  68. int synth_buf_offset[MPA_MAX_CHANNELS];
  69. DECLARE_ALIGNED(32, INTFLOAT, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT];
  70. INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
  71. GranuleDef granules[2][2]; /* Used in Layer 3 */
  72. int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
  73. int dither_state;
  74. int err_recognition;
  75. AVCodecContext* avctx;
  76. MPADSPContext mpadsp;
  77. DSPContext dsp;
  78. AVFrame frame;
  79. } MPADecodeContext;
  80. #if CONFIG_FLOAT
  81. # define SHR(a,b) ((a)*(1.0f/(1<<(b))))
  82. # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
  83. # define FIXR(x) ((float)(x))
  84. # define FIXHR(x) ((float)(x))
  85. # define MULH3(x, y, s) ((s)*(y)*(x))
  86. # define MULLx(x, y, s) ((y)*(x))
  87. # define RENAME(a) a ## _float
  88. # define OUT_FMT AV_SAMPLE_FMT_FLT
  89. #else
  90. # define SHR(a,b) ((a)>>(b))
  91. /* WARNING: only correct for positive numbers */
  92. # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
  93. # define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
  94. # define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
  95. # define MULH3(x, y, s) MULH((s)*(x), y)
  96. # define MULLx(x, y, s) MULL(x,y,s)
  97. # define RENAME(a) a ## _fixed
  98. # define OUT_FMT AV_SAMPLE_FMT_S16
  99. #endif
  100. /****************/
  101. #define HEADER_SIZE 4
  102. #include "mpegaudiodata.h"
  103. #include "mpegaudiodectab.h"
  104. /* vlc structure for decoding layer 3 huffman tables */
  105. static VLC huff_vlc[16];
  106. static VLC_TYPE huff_vlc_tables[
  107. 0 + 128 + 128 + 128 + 130 + 128 + 154 + 166 +
  108. 142 + 204 + 190 + 170 + 542 + 460 + 662 + 414
  109. ][2];
  110. static const int huff_vlc_tables_sizes[16] = {
  111. 0, 128, 128, 128, 130, 128, 154, 166,
  112. 142, 204, 190, 170, 542, 460, 662, 414
  113. };
  114. static VLC huff_quad_vlc[2];
  115. static VLC_TYPE huff_quad_vlc_tables[128+16][2];
  116. static const int huff_quad_vlc_tables_sizes[2] = { 128, 16 };
  117. /* computed from band_size_long */
  118. static uint16_t band_index_long[9][23];
  119. #include "mpegaudio_tablegen.h"
  120. /* intensity stereo coef table */
  121. static INTFLOAT is_table[2][16];
  122. static INTFLOAT is_table_lsf[2][2][16];
  123. static INTFLOAT csa_table[8][4];
  124. static int16_t division_tab3[1<<6 ];
  125. static int16_t division_tab5[1<<8 ];
  126. static int16_t division_tab9[1<<11];
  127. static int16_t * const division_tabs[4] = {
  128. division_tab3, division_tab5, NULL, division_tab9
  129. };
  130. /* lower 2 bits: modulo 3, higher bits: shift */
  131. static uint16_t scale_factor_modshift[64];
  132. /* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
  133. static int32_t scale_factor_mult[15][3];
  134. /* mult table for layer 2 group quantization */
  135. #define SCALE_GEN(v) \
  136. { FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
  137. static const int32_t scale_factor_mult2[3][3] = {
  138. SCALE_GEN(4.0 / 3.0), /* 3 steps */
  139. SCALE_GEN(4.0 / 5.0), /* 5 steps */
  140. SCALE_GEN(4.0 / 9.0), /* 9 steps */
  141. };
  142. /**
  143. * Convert region offsets to region sizes and truncate
  144. * size to big_values.
  145. */
  146. static void ff_region_offset2size(GranuleDef *g)
  147. {
  148. int i, k, j = 0;
  149. g->region_size[2] = 576 / 2;
  150. for (i = 0; i < 3; i++) {
  151. k = FFMIN(g->region_size[i], g->big_values);
  152. g->region_size[i] = k - j;
  153. j = k;
  154. }
  155. }
  156. static void ff_init_short_region(MPADecodeContext *s, GranuleDef *g)
  157. {
  158. if (g->block_type == 2) {
  159. if (s->sample_rate_index != 8)
  160. g->region_size[0] = (36 / 2);
  161. else
  162. g->region_size[0] = (72 / 2);
  163. } else {
  164. if (s->sample_rate_index <= 2)
  165. g->region_size[0] = (36 / 2);
  166. else if (s->sample_rate_index != 8)
  167. g->region_size[0] = (54 / 2);
  168. else
  169. g->region_size[0] = (108 / 2);
  170. }
  171. g->region_size[1] = (576 / 2);
  172. }
  173. static void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2)
  174. {
  175. int l;
  176. g->region_size[0] = band_index_long[s->sample_rate_index][ra1 + 1] >> 1;
  177. /* should not overflow */
  178. l = FFMIN(ra1 + ra2 + 2, 22);
  179. g->region_size[1] = band_index_long[s->sample_rate_index][ l] >> 1;
  180. }
  181. static void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g)
  182. {
  183. if (g->block_type == 2) {
  184. if (g->switch_point) {
  185. /* if switched mode, we handle the 36 first samples as
  186. long blocks. For 8000Hz, we handle the 72 first
  187. exponents as long blocks */
  188. if (s->sample_rate_index <= 2)
  189. g->long_end = 8;
  190. else
  191. g->long_end = 6;
  192. g->short_start = 3;
  193. } else {
  194. g->long_end = 0;
  195. g->short_start = 0;
  196. }
  197. } else {
  198. g->short_start = 13;
  199. g->long_end = 22;
  200. }
  201. }
  202. /* layer 1 unscaling */
  203. /* n = number of bits of the mantissa minus 1 */
  204. static inline int l1_unscale(int n, int mant, int scale_factor)
  205. {
  206. int shift, mod;
  207. int64_t val;
  208. shift = scale_factor_modshift[scale_factor];
  209. mod = shift & 3;
  210. shift >>= 2;
  211. val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]);
  212. shift += n;
  213. /* NOTE: at this point, 1 <= shift >= 21 + 15 */
  214. return (int)((val + (1LL << (shift - 1))) >> shift);
  215. }
  216. static inline int l2_unscale_group(int steps, int mant, int scale_factor)
  217. {
  218. int shift, mod, val;
  219. shift = scale_factor_modshift[scale_factor];
  220. mod = shift & 3;
  221. shift >>= 2;
  222. val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
  223. /* NOTE: at this point, 0 <= shift <= 21 */
  224. if (shift > 0)
  225. val = (val + (1 << (shift - 1))) >> shift;
  226. return val;
  227. }
  228. /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
  229. static inline int l3_unscale(int value, int exponent)
  230. {
  231. unsigned int m;
  232. int e;
  233. e = table_4_3_exp [4 * value + (exponent & 3)];
  234. m = table_4_3_value[4 * value + (exponent & 3)];
  235. e -= exponent >> 2;
  236. assert(e >= 1);
  237. if (e > 31)
  238. return 0;
  239. m = (m + (1 << (e - 1))) >> e;
  240. return m;
  241. }
  242. static av_cold void decode_init_static(void)
  243. {
  244. int i, j, k;
  245. int offset;
  246. /* scale factors table for layer 1/2 */
  247. for (i = 0; i < 64; i++) {
  248. int shift, mod;
  249. /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
  250. shift = i / 3;
  251. mod = i % 3;
  252. scale_factor_modshift[i] = mod | (shift << 2);
  253. }
  254. /* scale factor multiply for layer 1 */
  255. for (i = 0; i < 15; i++) {
  256. int n, norm;
  257. n = i + 2;
  258. norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
  259. scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
  260. scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
  261. scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
  262. av_dlog(NULL, "%d: norm=%x s=%x %x %x\n", i, norm,
  263. scale_factor_mult[i][0],
  264. scale_factor_mult[i][1],
  265. scale_factor_mult[i][2]);
  266. }
  267. RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));
  268. /* huffman decode tables */
  269. offset = 0;
  270. for (i = 1; i < 16; i++) {
  271. const HuffTable *h = &mpa_huff_tables[i];
  272. int xsize, x, y;
  273. uint8_t tmp_bits [512] = { 0 };
  274. uint16_t tmp_codes[512] = { 0 };
  275. xsize = h->xsize;
  276. j = 0;
  277. for (x = 0; x < xsize; x++) {
  278. for (y = 0; y < xsize; y++) {
  279. tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ];
  280. tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];
  281. }
  282. }
  283. /* XXX: fail test */
  284. huff_vlc[i].table = huff_vlc_tables+offset;
  285. huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
  286. init_vlc(&huff_vlc[i], 7, 512,
  287. tmp_bits, 1, 1, tmp_codes, 2, 2,
  288. INIT_VLC_USE_NEW_STATIC);
  289. offset += huff_vlc_tables_sizes[i];
  290. }
  291. assert(offset == FF_ARRAY_ELEMS(huff_vlc_tables));
  292. offset = 0;
  293. for (i = 0; i < 2; i++) {
  294. huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
  295. huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
  296. init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
  297. mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
  298. INIT_VLC_USE_NEW_STATIC);
  299. offset += huff_quad_vlc_tables_sizes[i];
  300. }
  301. assert(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));
  302. for (i = 0; i < 9; i++) {
  303. k = 0;
  304. for (j = 0; j < 22; j++) {
  305. band_index_long[i][j] = k;
  306. k += band_size_long[i][j];
  307. }
  308. band_index_long[i][22] = k;
  309. }
  310. /* compute n ^ (4/3) and store it in mantissa/exp format */
  311. mpegaudio_tableinit();
  312. for (i = 0; i < 4; i++) {
  313. if (ff_mpa_quant_bits[i] < 0) {
  314. for (j = 0; j < (1 << (-ff_mpa_quant_bits[i]+1)); j++) {
  315. int val1, val2, val3, steps;
  316. int val = j;
  317. steps = ff_mpa_quant_steps[i];
  318. val1 = val % steps;
  319. val /= steps;
  320. val2 = val % steps;
  321. val3 = val / steps;
  322. division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);
  323. }
  324. }
  325. }
  326. for (i = 0; i < 7; i++) {
  327. float f;
  328. INTFLOAT v;
  329. if (i != 6) {
  330. f = tan((double)i * M_PI / 12.0);
  331. v = FIXR(f / (1.0 + f));
  332. } else {
  333. v = FIXR(1.0);
  334. }
  335. is_table[0][ i] = v;
  336. is_table[1][6 - i] = v;
  337. }
  338. /* invalid values */
  339. for (i = 7; i < 16; i++)
  340. is_table[0][i] = is_table[1][i] = 0.0;
  341. for (i = 0; i < 16; i++) {
  342. double f;
  343. int e, k;
  344. for (j = 0; j < 2; j++) {
  345. e = -(j + 1) * ((i + 1) >> 1);
  346. f = pow(2.0, e / 4.0);
  347. k = i & 1;
  348. is_table_lsf[j][k ^ 1][i] = FIXR(f);
  349. is_table_lsf[j][k ][i] = FIXR(1.0);
  350. av_dlog(NULL, "is_table_lsf %d %d: %f %f\n",
  351. i, j, (float) is_table_lsf[j][0][i],
  352. (float) is_table_lsf[j][1][i]);
  353. }
  354. }
  355. for (i = 0; i < 8; i++) {
  356. float ci, cs, ca;
  357. ci = ci_table[i];
  358. cs = 1.0 / sqrt(1.0 + ci * ci);
  359. ca = cs * ci;
  360. #if !CONFIG_FLOAT
  361. csa_table[i][0] = FIXHR(cs/4);
  362. csa_table[i][1] = FIXHR(ca/4);
  363. csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);
  364. csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);
  365. #else
  366. csa_table[i][0] = cs;
  367. csa_table[i][1] = ca;
  368. csa_table[i][2] = ca + cs;
  369. csa_table[i][3] = ca - cs;
  370. #endif
  371. }
  372. }
  373. static av_cold int decode_init(AVCodecContext * avctx)
  374. {
  375. static int initialized_tables = 0;
  376. MPADecodeContext *s = avctx->priv_data;
  377. if (!initialized_tables) {
  378. decode_init_static();
  379. initialized_tables = 1;
  380. }
  381. s->avctx = avctx;
  382. ff_mpadsp_init(&s->mpadsp);
  383. ff_dsputil_init(&s->dsp, avctx);
  384. avctx->sample_fmt= OUT_FMT;
  385. s->err_recognition = avctx->err_recognition;
  386. if (avctx->codec_id == AV_CODEC_ID_MP3ADU)
  387. s->adu_mode = 1;
  388. avcodec_get_frame_defaults(&s->frame);
  389. avctx->coded_frame = &s->frame;
  390. return 0;
  391. }
  392. #define C3 FIXHR(0.86602540378443864676/2)
  393. #define C4 FIXHR(0.70710678118654752439/2) //0.5 / cos(pi*(9)/36)
  394. #define C5 FIXHR(0.51763809020504152469/2) //0.5 / cos(pi*(5)/36)
  395. #define C6 FIXHR(1.93185165257813657349/4) //0.5 / cos(pi*(15)/36)
  396. /* 12 points IMDCT. We compute it "by hand" by factorizing obvious
  397. cases. */
  398. static void imdct12(INTFLOAT *out, INTFLOAT *in)
  399. {
  400. INTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
  401. in0 = in[0*3];
  402. in1 = in[1*3] + in[0*3];
  403. in2 = in[2*3] + in[1*3];
  404. in3 = in[3*3] + in[2*3];
  405. in4 = in[4*3] + in[3*3];
  406. in5 = in[5*3] + in[4*3];
  407. in5 += in3;
  408. in3 += in1;
  409. in2 = MULH3(in2, C3, 2);
  410. in3 = MULH3(in3, C3, 4);
  411. t1 = in0 - in4;
  412. t2 = MULH3(in1 - in5, C4, 2);
  413. out[ 7] =
  414. out[10] = t1 + t2;
  415. out[ 1] =
  416. out[ 4] = t1 - t2;
  417. in0 += SHR(in4, 1);
  418. in4 = in0 + in2;
  419. in5 += 2*in1;
  420. in1 = MULH3(in5 + in3, C5, 1);
  421. out[ 8] =
  422. out[ 9] = in4 + in1;
  423. out[ 2] =
  424. out[ 3] = in4 - in1;
  425. in0 -= in2;
  426. in5 = MULH3(in5 - in3, C6, 2);
  427. out[ 0] =
  428. out[ 5] = in0 - in5;
  429. out[ 6] =
  430. out[11] = in0 + in5;
  431. }
  432. /* return the number of decoded frames */
  433. static int mp_decode_layer1(MPADecodeContext *s)
  434. {
  435. int bound, i, v, n, ch, j, mant;
  436. uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
  437. uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
  438. if (s->mode == MPA_JSTEREO)
  439. bound = (s->mode_ext + 1) * 4;
  440. else
  441. bound = SBLIMIT;
  442. /* allocation bits */
  443. for (i = 0; i < bound; i++) {
  444. for (ch = 0; ch < s->nb_channels; ch++) {
  445. allocation[ch][i] = get_bits(&s->gb, 4);
  446. }
  447. }
  448. for (i = bound; i < SBLIMIT; i++)
  449. allocation[0][i] = get_bits(&s->gb, 4);
  450. /* scale factors */
  451. for (i = 0; i < bound; i++) {
  452. for (ch = 0; ch < s->nb_channels; ch++) {
  453. if (allocation[ch][i])
  454. scale_factors[ch][i] = get_bits(&s->gb, 6);
  455. }
  456. }
  457. for (i = bound; i < SBLIMIT; i++) {
  458. if (allocation[0][i]) {
  459. scale_factors[0][i] = get_bits(&s->gb, 6);
  460. scale_factors[1][i] = get_bits(&s->gb, 6);
  461. }
  462. }
  463. /* compute samples */
  464. for (j = 0; j < 12; j++) {
  465. for (i = 0; i < bound; i++) {
  466. for (ch = 0; ch < s->nb_channels; ch++) {
  467. n = allocation[ch][i];
  468. if (n) {
  469. mant = get_bits(&s->gb, n + 1);
  470. v = l1_unscale(n, mant, scale_factors[ch][i]);
  471. } else {
  472. v = 0;
  473. }
  474. s->sb_samples[ch][j][i] = v;
  475. }
  476. }
  477. for (i = bound; i < SBLIMIT; i++) {
  478. n = allocation[0][i];
  479. if (n) {
  480. mant = get_bits(&s->gb, n + 1);
  481. v = l1_unscale(n, mant, scale_factors[0][i]);
  482. s->sb_samples[0][j][i] = v;
  483. v = l1_unscale(n, mant, scale_factors[1][i]);
  484. s->sb_samples[1][j][i] = v;
  485. } else {
  486. s->sb_samples[0][j][i] = 0;
  487. s->sb_samples[1][j][i] = 0;
  488. }
  489. }
  490. }
  491. return 12;
  492. }
  493. static int mp_decode_layer2(MPADecodeContext *s)
  494. {
  495. int sblimit; /* number of used subbands */
  496. const unsigned char *alloc_table;
  497. int table, bit_alloc_bits, i, j, ch, bound, v;
  498. unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
  499. unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
  500. unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
  501. int scale, qindex, bits, steps, k, l, m, b;
  502. /* select decoding table */
  503. table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
  504. s->sample_rate, s->lsf);
  505. sblimit = ff_mpa_sblimit_table[table];
  506. alloc_table = ff_mpa_alloc_tables[table];
  507. if (s->mode == MPA_JSTEREO)
  508. bound = (s->mode_ext + 1) * 4;
  509. else
  510. bound = sblimit;
  511. av_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
  512. /* sanity check */
  513. if (bound > sblimit)
  514. bound = sblimit;
  515. /* parse bit allocation */
  516. j = 0;
  517. for (i = 0; i < bound; i++) {
  518. bit_alloc_bits = alloc_table[j];
  519. for (ch = 0; ch < s->nb_channels; ch++)
  520. bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
  521. j += 1 << bit_alloc_bits;
  522. }
  523. for (i = bound; i < sblimit; i++) {
  524. bit_alloc_bits = alloc_table[j];
  525. v = get_bits(&s->gb, bit_alloc_bits);
  526. bit_alloc[0][i] = v;
  527. bit_alloc[1][i] = v;
  528. j += 1 << bit_alloc_bits;
  529. }
  530. /* scale codes */
  531. for (i = 0; i < sblimit; i++) {
  532. for (ch = 0; ch < s->nb_channels; ch++) {
  533. if (bit_alloc[ch][i])
  534. scale_code[ch][i] = get_bits(&s->gb, 2);
  535. }
  536. }
  537. /* scale factors */
  538. for (i = 0; i < sblimit; i++) {
  539. for (ch = 0; ch < s->nb_channels; ch++) {
  540. if (bit_alloc[ch][i]) {
  541. sf = scale_factors[ch][i];
  542. switch (scale_code[ch][i]) {
  543. default:
  544. case 0:
  545. sf[0] = get_bits(&s->gb, 6);
  546. sf[1] = get_bits(&s->gb, 6);
  547. sf[2] = get_bits(&s->gb, 6);
  548. break;
  549. case 2:
  550. sf[0] = get_bits(&s->gb, 6);
  551. sf[1] = sf[0];
  552. sf[2] = sf[0];
  553. break;
  554. case 1:
  555. sf[0] = get_bits(&s->gb, 6);
  556. sf[2] = get_bits(&s->gb, 6);
  557. sf[1] = sf[0];
  558. break;
  559. case 3:
  560. sf[0] = get_bits(&s->gb, 6);
  561. sf[2] = get_bits(&s->gb, 6);
  562. sf[1] = sf[2];
  563. break;
  564. }
  565. }
  566. }
  567. }
  568. /* samples */
  569. for (k = 0; k < 3; k++) {
  570. for (l = 0; l < 12; l += 3) {
  571. j = 0;
  572. for (i = 0; i < bound; i++) {
  573. bit_alloc_bits = alloc_table[j];
  574. for (ch = 0; ch < s->nb_channels; ch++) {
  575. b = bit_alloc[ch][i];
  576. if (b) {
  577. scale = scale_factors[ch][i][k];
  578. qindex = alloc_table[j+b];
  579. bits = ff_mpa_quant_bits[qindex];
  580. if (bits < 0) {
  581. int v2;
  582. /* 3 values at the same time */
  583. v = get_bits(&s->gb, -bits);
  584. v2 = division_tabs[qindex][v];
  585. steps = ff_mpa_quant_steps[qindex];
  586. s->sb_samples[ch][k * 12 + l + 0][i] =
  587. l2_unscale_group(steps, v2 & 15, scale);
  588. s->sb_samples[ch][k * 12 + l + 1][i] =
  589. l2_unscale_group(steps, (v2 >> 4) & 15, scale);
  590. s->sb_samples[ch][k * 12 + l + 2][i] =
  591. l2_unscale_group(steps, v2 >> 8 , scale);
  592. } else {
  593. for (m = 0; m < 3; m++) {
  594. v = get_bits(&s->gb, bits);
  595. v = l1_unscale(bits - 1, v, scale);
  596. s->sb_samples[ch][k * 12 + l + m][i] = v;
  597. }
  598. }
  599. } else {
  600. s->sb_samples[ch][k * 12 + l + 0][i] = 0;
  601. s->sb_samples[ch][k * 12 + l + 1][i] = 0;
  602. s->sb_samples[ch][k * 12 + l + 2][i] = 0;
  603. }
  604. }
  605. /* next subband in alloc table */
  606. j += 1 << bit_alloc_bits;
  607. }
  608. /* XXX: find a way to avoid this duplication of code */
  609. for (i = bound; i < sblimit; i++) {
  610. bit_alloc_bits = alloc_table[j];
  611. b = bit_alloc[0][i];
  612. if (b) {
  613. int mant, scale0, scale1;
  614. scale0 = scale_factors[0][i][k];
  615. scale1 = scale_factors[1][i][k];
  616. qindex = alloc_table[j+b];
  617. bits = ff_mpa_quant_bits[qindex];
  618. if (bits < 0) {
  619. /* 3 values at the same time */
  620. v = get_bits(&s->gb, -bits);
  621. steps = ff_mpa_quant_steps[qindex];
  622. mant = v % steps;
  623. v = v / steps;
  624. s->sb_samples[0][k * 12 + l + 0][i] =
  625. l2_unscale_group(steps, mant, scale0);
  626. s->sb_samples[1][k * 12 + l + 0][i] =
  627. l2_unscale_group(steps, mant, scale1);
  628. mant = v % steps;
  629. v = v / steps;
  630. s->sb_samples[0][k * 12 + l + 1][i] =
  631. l2_unscale_group(steps, mant, scale0);
  632. s->sb_samples[1][k * 12 + l + 1][i] =
  633. l2_unscale_group(steps, mant, scale1);
  634. s->sb_samples[0][k * 12 + l + 2][i] =
  635. l2_unscale_group(steps, v, scale0);
  636. s->sb_samples[1][k * 12 + l + 2][i] =
  637. l2_unscale_group(steps, v, scale1);
  638. } else {
  639. for (m = 0; m < 3; m++) {
  640. mant = get_bits(&s->gb, bits);
  641. s->sb_samples[0][k * 12 + l + m][i] =
  642. l1_unscale(bits - 1, mant, scale0);
  643. s->sb_samples[1][k * 12 + l + m][i] =
  644. l1_unscale(bits - 1, mant, scale1);
  645. }
  646. }
  647. } else {
  648. s->sb_samples[0][k * 12 + l + 0][i] = 0;
  649. s->sb_samples[0][k * 12 + l + 1][i] = 0;
  650. s->sb_samples[0][k * 12 + l + 2][i] = 0;
  651. s->sb_samples[1][k * 12 + l + 0][i] = 0;
  652. s->sb_samples[1][k * 12 + l + 1][i] = 0;
  653. s->sb_samples[1][k * 12 + l + 2][i] = 0;
  654. }
  655. /* next subband in alloc table */
  656. j += 1 << bit_alloc_bits;
  657. }
  658. /* fill remaining samples to zero */
  659. for (i = sblimit; i < SBLIMIT; i++) {
  660. for (ch = 0; ch < s->nb_channels; ch++) {
  661. s->sb_samples[ch][k * 12 + l + 0][i] = 0;
  662. s->sb_samples[ch][k * 12 + l + 1][i] = 0;
  663. s->sb_samples[ch][k * 12 + l + 2][i] = 0;
  664. }
  665. }
  666. }
  667. }
  668. return 3 * 12;
  669. }
  670. #define SPLIT(dst,sf,n) \
  671. if (n == 3) { \
  672. int m = (sf * 171) >> 9; \
  673. dst = sf - 3 * m; \
  674. sf = m; \
  675. } else if (n == 4) { \
  676. dst = sf & 3; \
  677. sf >>= 2; \
  678. } else if (n == 5) { \
  679. int m = (sf * 205) >> 10; \
  680. dst = sf - 5 * m; \
  681. sf = m; \
  682. } else if (n == 6) { \
  683. int m = (sf * 171) >> 10; \
  684. dst = sf - 6 * m; \
  685. sf = m; \
  686. } else { \
  687. dst = 0; \
  688. }
  689. static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2,
  690. int n3)
  691. {
  692. SPLIT(slen[3], sf, n3)
  693. SPLIT(slen[2], sf, n2)
  694. SPLIT(slen[1], sf, n1)
  695. slen[0] = sf;
  696. }
  697. static void exponents_from_scale_factors(MPADecodeContext *s, GranuleDef *g,
  698. int16_t *exponents)
  699. {
  700. const uint8_t *bstab, *pretab;
  701. int len, i, j, k, l, v0, shift, gain, gains[3];
  702. int16_t *exp_ptr;
  703. exp_ptr = exponents;
  704. gain = g->global_gain - 210;
  705. shift = g->scalefac_scale + 1;
  706. bstab = band_size_long[s->sample_rate_index];
  707. pretab = mpa_pretab[g->preflag];
  708. for (i = 0; i < g->long_end; i++) {
  709. v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
  710. len = bstab[i];
  711. for (j = len; j > 0; j--)
  712. *exp_ptr++ = v0;
  713. }
  714. if (g->short_start < 13) {
  715. bstab = band_size_short[s->sample_rate_index];
  716. gains[0] = gain - (g->subblock_gain[0] << 3);
  717. gains[1] = gain - (g->subblock_gain[1] << 3);
  718. gains[2] = gain - (g->subblock_gain[2] << 3);
  719. k = g->long_end;
  720. for (i = g->short_start; i < 13; i++) {
  721. len = bstab[i];
  722. for (l = 0; l < 3; l++) {
  723. v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
  724. for (j = len; j > 0; j--)
  725. *exp_ptr++ = v0;
  726. }
  727. }
  728. }
  729. }
  730. /* handle n = 0 too */
  731. static inline int get_bitsz(GetBitContext *s, int n)
  732. {
  733. return n ? get_bits(s, n) : 0;
  734. }
  735. static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos,
  736. int *end_pos2)
  737. {
  738. if (s->in_gb.buffer && *pos >= s->gb.size_in_bits) {
  739. s->gb = s->in_gb;
  740. s->in_gb.buffer = NULL;
  741. assert((get_bits_count(&s->gb) & 7) == 0);
  742. skip_bits_long(&s->gb, *pos - *end_pos);
  743. *end_pos2 =
  744. *end_pos = *end_pos2 + get_bits_count(&s->gb) - *pos;
  745. *pos = get_bits_count(&s->gb);
  746. }
  747. }
  748. /* Following is a optimized code for
  749. INTFLOAT v = *src
  750. if(get_bits1(&s->gb))
  751. v = -v;
  752. *dst = v;
  753. */
  754. #if CONFIG_FLOAT
  755. #define READ_FLIP_SIGN(dst,src) \
  756. v = AV_RN32A(src) ^ (get_bits1(&s->gb) << 31); \
  757. AV_WN32A(dst, v);
  758. #else
  759. #define READ_FLIP_SIGN(dst,src) \
  760. v = -get_bits1(&s->gb); \
  761. *(dst) = (*(src) ^ v) - v;
  762. #endif
  763. static int huffman_decode(MPADecodeContext *s, GranuleDef *g,
  764. int16_t *exponents, int end_pos2)
  765. {
  766. int s_index;
  767. int i;
  768. int last_pos, bits_left;
  769. VLC *vlc;
  770. int end_pos = FFMIN(end_pos2, s->gb.size_in_bits);
  771. /* low frequencies (called big values) */
  772. s_index = 0;
  773. for (i = 0; i < 3; i++) {
  774. int j, k, l, linbits;
  775. j = g->region_size[i];
  776. if (j == 0)
  777. continue;
  778. /* select vlc table */
  779. k = g->table_select[i];
  780. l = mpa_huff_data[k][0];
  781. linbits = mpa_huff_data[k][1];
  782. vlc = &huff_vlc[l];
  783. if (!l) {
  784. memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * 2 * j);
  785. s_index += 2 * j;
  786. continue;
  787. }
  788. /* read huffcode and compute each couple */
  789. for (; j > 0; j--) {
  790. int exponent, x, y;
  791. int v;
  792. int pos = get_bits_count(&s->gb);
  793. if (pos >= end_pos){
  794. switch_buffer(s, &pos, &end_pos, &end_pos2);
  795. if (pos >= end_pos)
  796. break;
  797. }
  798. y = get_vlc2(&s->gb, vlc->table, 7, 3);
  799. if (!y) {
  800. g->sb_hybrid[s_index ] =
  801. g->sb_hybrid[s_index+1] = 0;
  802. s_index += 2;
  803. continue;
  804. }
  805. exponent= exponents[s_index];
  806. av_dlog(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n",
  807. i, g->region_size[i] - j, x, y, exponent);
  808. if (y & 16) {
  809. x = y >> 5;
  810. y = y & 0x0f;
  811. if (x < 15) {
  812. READ_FLIP_SIGN(g->sb_hybrid + s_index, RENAME(expval_table)[exponent] + x)
  813. } else {
  814. x += get_bitsz(&s->gb, linbits);
  815. v = l3_unscale(x, exponent);
  816. if (get_bits1(&s->gb))
  817. v = -v;
  818. g->sb_hybrid[s_index] = v;
  819. }
  820. if (y < 15) {
  821. READ_FLIP_SIGN(g->sb_hybrid + s_index + 1, RENAME(expval_table)[exponent] + y)
  822. } else {
  823. y += get_bitsz(&s->gb, linbits);
  824. v = l3_unscale(y, exponent);
  825. if (get_bits1(&s->gb))
  826. v = -v;
  827. g->sb_hybrid[s_index+1] = v;
  828. }
  829. } else {
  830. x = y >> 5;
  831. y = y & 0x0f;
  832. x += y;
  833. if (x < 15) {
  834. READ_FLIP_SIGN(g->sb_hybrid + s_index + !!y, RENAME(expval_table)[exponent] + x)
  835. } else {
  836. x += get_bitsz(&s->gb, linbits);
  837. v = l3_unscale(x, exponent);
  838. if (get_bits1(&s->gb))
  839. v = -v;
  840. g->sb_hybrid[s_index+!!y] = v;
  841. }
  842. g->sb_hybrid[s_index + !y] = 0;
  843. }
  844. s_index += 2;
  845. }
  846. }
  847. /* high frequencies */
  848. vlc = &huff_quad_vlc[g->count1table_select];
  849. last_pos = 0;
  850. while (s_index <= 572) {
  851. int pos, code;
  852. pos = get_bits_count(&s->gb);
  853. if (pos >= end_pos) {
  854. if (pos > end_pos2 && last_pos) {
  855. /* some encoders generate an incorrect size for this
  856. part. We must go back into the data */
  857. s_index -= 4;
  858. skip_bits_long(&s->gb, last_pos - pos);
  859. av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
  860. if(s->err_recognition & AV_EF_BITSTREAM)
  861. s_index=0;
  862. break;
  863. }
  864. switch_buffer(s, &pos, &end_pos, &end_pos2);
  865. if (pos >= end_pos)
  866. break;
  867. }
  868. last_pos = pos;
  869. code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
  870. av_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
  871. g->sb_hybrid[s_index+0] =
  872. g->sb_hybrid[s_index+1] =
  873. g->sb_hybrid[s_index+2] =
  874. g->sb_hybrid[s_index+3] = 0;
  875. while (code) {
  876. static const int idxtab[16] = { 3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0 };
  877. int v;
  878. int pos = s_index + idxtab[code];
  879. code ^= 8 >> idxtab[code];
  880. READ_FLIP_SIGN(g->sb_hybrid + pos, RENAME(exp_table)+exponents[pos])
  881. }
  882. s_index += 4;
  883. }
  884. /* skip extension bits */
  885. bits_left = end_pos2 - get_bits_count(&s->gb);
  886. if (bits_left < 0 && (s->err_recognition & AV_EF_BUFFER)) {
  887. av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
  888. s_index=0;
  889. } else if (bits_left > 0 && (s->err_recognition & AV_EF_BUFFER)) {
  890. av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
  891. s_index = 0;
  892. }
  893. memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * (576 - s_index));
  894. skip_bits_long(&s->gb, bits_left);
  895. i = get_bits_count(&s->gb);
  896. switch_buffer(s, &i, &end_pos, &end_pos2);
  897. return 0;
  898. }
  899. /* Reorder short blocks from bitstream order to interleaved order. It
  900. would be faster to do it in parsing, but the code would be far more
  901. complicated */
  902. static void reorder_block(MPADecodeContext *s, GranuleDef *g)
  903. {
  904. int i, j, len;
  905. INTFLOAT *ptr, *dst, *ptr1;
  906. INTFLOAT tmp[576];
  907. if (g->block_type != 2)
  908. return;
  909. if (g->switch_point) {
  910. if (s->sample_rate_index != 8)
  911. ptr = g->sb_hybrid + 36;
  912. else
  913. ptr = g->sb_hybrid + 72;
  914. } else {
  915. ptr = g->sb_hybrid;
  916. }
  917. for (i = g->short_start; i < 13; i++) {
  918. len = band_size_short[s->sample_rate_index][i];
  919. ptr1 = ptr;
  920. dst = tmp;
  921. for (j = len; j > 0; j--) {
  922. *dst++ = ptr[0*len];
  923. *dst++ = ptr[1*len];
  924. *dst++ = ptr[2*len];
  925. ptr++;
  926. }
  927. ptr += 2 * len;
  928. memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
  929. }
  930. }
  931. #define ISQRT2 FIXR(0.70710678118654752440)
  932. static void compute_stereo(MPADecodeContext *s, GranuleDef *g0, GranuleDef *g1)
  933. {
  934. int i, j, k, l;
  935. int sf_max, sf, len, non_zero_found;
  936. INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;
  937. int non_zero_found_short[3];
  938. /* intensity stereo */
  939. if (s->mode_ext & MODE_EXT_I_STEREO) {
  940. if (!s->lsf) {
  941. is_tab = is_table;
  942. sf_max = 7;
  943. } else {
  944. is_tab = is_table_lsf[g1->scalefac_compress & 1];
  945. sf_max = 16;
  946. }
  947. tab0 = g0->sb_hybrid + 576;
  948. tab1 = g1->sb_hybrid + 576;
  949. non_zero_found_short[0] = 0;
  950. non_zero_found_short[1] = 0;
  951. non_zero_found_short[2] = 0;
  952. k = (13 - g1->short_start) * 3 + g1->long_end - 3;
  953. for (i = 12; i >= g1->short_start; i--) {
  954. /* for last band, use previous scale factor */
  955. if (i != 11)
  956. k -= 3;
  957. len = band_size_short[s->sample_rate_index][i];
  958. for (l = 2; l >= 0; l--) {
  959. tab0 -= len;
  960. tab1 -= len;
  961. if (!non_zero_found_short[l]) {
  962. /* test if non zero band. if so, stop doing i-stereo */
  963. for (j = 0; j < len; j++) {
  964. if (tab1[j] != 0) {
  965. non_zero_found_short[l] = 1;
  966. goto found1;
  967. }
  968. }
  969. sf = g1->scale_factors[k + l];
  970. if (sf >= sf_max)
  971. goto found1;
  972. v1 = is_tab[0][sf];
  973. v2 = is_tab[1][sf];
  974. for (j = 0; j < len; j++) {
  975. tmp0 = tab0[j];
  976. tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
  977. tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
  978. }
  979. } else {
  980. found1:
  981. if (s->mode_ext & MODE_EXT_MS_STEREO) {
  982. /* lower part of the spectrum : do ms stereo
  983. if enabled */
  984. for (j = 0; j < len; j++) {
  985. tmp0 = tab0[j];
  986. tmp1 = tab1[j];
  987. tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
  988. tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
  989. }
  990. }
  991. }
  992. }
  993. }
  994. non_zero_found = non_zero_found_short[0] |
  995. non_zero_found_short[1] |
  996. non_zero_found_short[2];
  997. for (i = g1->long_end - 1;i >= 0;i--) {
  998. len = band_size_long[s->sample_rate_index][i];
  999. tab0 -= len;
  1000. tab1 -= len;
  1001. /* test if non zero band. if so, stop doing i-stereo */
  1002. if (!non_zero_found) {
  1003. for (j = 0; j < len; j++) {
  1004. if (tab1[j] != 0) {
  1005. non_zero_found = 1;
  1006. goto found2;
  1007. }
  1008. }
  1009. /* for last band, use previous scale factor */
  1010. k = (i == 21) ? 20 : i;
  1011. sf = g1->scale_factors[k];
  1012. if (sf >= sf_max)
  1013. goto found2;
  1014. v1 = is_tab[0][sf];
  1015. v2 = is_tab[1][sf];
  1016. for (j = 0; j < len; j++) {
  1017. tmp0 = tab0[j];
  1018. tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
  1019. tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
  1020. }
  1021. } else {
  1022. found2:
  1023. if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1024. /* lower part of the spectrum : do ms stereo
  1025. if enabled */
  1026. for (j = 0; j < len; j++) {
  1027. tmp0 = tab0[j];
  1028. tmp1 = tab1[j];
  1029. tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
  1030. tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
  1031. }
  1032. }
  1033. }
  1034. }
  1035. } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1036. /* ms stereo ONLY */
  1037. /* NOTE: the 1/sqrt(2) normalization factor is included in the
  1038. global gain */
  1039. #if CONFIG_FLOAT
  1040. s-> dsp.butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);
  1041. #else
  1042. tab0 = g0->sb_hybrid;
  1043. tab1 = g1->sb_hybrid;
  1044. for (i = 0; i < 576; i++) {
  1045. tmp0 = tab0[i];
  1046. tmp1 = tab1[i];
  1047. tab0[i] = tmp0 + tmp1;
  1048. tab1[i] = tmp0 - tmp1;
  1049. }
  1050. #endif
  1051. }
  1052. }
  1053. #if CONFIG_FLOAT
  1054. #define AA(j) do { \
  1055. float tmp0 = ptr[-1-j]; \
  1056. float tmp1 = ptr[ j]; \
  1057. ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1]; \
  1058. ptr[ j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0]; \
  1059. } while (0)
  1060. #else
  1061. #define AA(j) do { \
  1062. int tmp0 = ptr[-1-j]; \
  1063. int tmp1 = ptr[ j]; \
  1064. int tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]); \
  1065. ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2])); \
  1066. ptr[ j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3])); \
  1067. } while (0)
  1068. #endif
  1069. static void compute_antialias(MPADecodeContext *s, GranuleDef *g)
  1070. {
  1071. INTFLOAT *ptr;
  1072. int n, i;
  1073. /* we antialias only "long" bands */
  1074. if (g->block_type == 2) {
  1075. if (!g->switch_point)
  1076. return;
  1077. /* XXX: check this for 8000Hz case */
  1078. n = 1;
  1079. } else {
  1080. n = SBLIMIT - 1;
  1081. }
  1082. ptr = g->sb_hybrid + 18;
  1083. for (i = n; i > 0; i--) {
  1084. AA(0);
  1085. AA(1);
  1086. AA(2);
  1087. AA(3);
  1088. AA(4);
  1089. AA(5);
  1090. AA(6);
  1091. AA(7);
  1092. ptr += 18;
  1093. }
  1094. }
  1095. static void compute_imdct(MPADecodeContext *s, GranuleDef *g,
  1096. INTFLOAT *sb_samples, INTFLOAT *mdct_buf)
  1097. {
  1098. INTFLOAT *win, *out_ptr, *ptr, *buf, *ptr1;
  1099. INTFLOAT out2[12];
  1100. int i, j, mdct_long_end, sblimit;
  1101. /* find last non zero block */
  1102. ptr = g->sb_hybrid + 576;
  1103. ptr1 = g->sb_hybrid + 2 * 18;
  1104. while (ptr >= ptr1) {
  1105. int32_t *p;
  1106. ptr -= 6;
  1107. p = (int32_t*)ptr;
  1108. if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
  1109. break;
  1110. }
  1111. sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
  1112. if (g->block_type == 2) {
  1113. /* XXX: check for 8000 Hz */
  1114. if (g->switch_point)
  1115. mdct_long_end = 2;
  1116. else
  1117. mdct_long_end = 0;
  1118. } else {
  1119. mdct_long_end = sblimit;
  1120. }
  1121. s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid,
  1122. mdct_long_end, g->switch_point,
  1123. g->block_type);
  1124. buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3);
  1125. ptr = g->sb_hybrid + 18 * mdct_long_end;
  1126. for (j = mdct_long_end; j < sblimit; j++) {
  1127. /* select frequency inversion */
  1128. win = RENAME(ff_mdct_win)[2 + (4 & -(j & 1))];
  1129. out_ptr = sb_samples + j;
  1130. for (i = 0; i < 6; i++) {
  1131. *out_ptr = buf[4*i];
  1132. out_ptr += SBLIMIT;
  1133. }
  1134. imdct12(out2, ptr + 0);
  1135. for (i = 0; i < 6; i++) {
  1136. *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*1)];
  1137. buf[4*(i + 6*2)] = MULH3(out2[i + 6], win[i + 6], 1);
  1138. out_ptr += SBLIMIT;
  1139. }
  1140. imdct12(out2, ptr + 1);
  1141. for (i = 0; i < 6; i++) {
  1142. *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*2)];
  1143. buf[4*(i + 6*0)] = MULH3(out2[i + 6], win[i + 6], 1);
  1144. out_ptr += SBLIMIT;
  1145. }
  1146. imdct12(out2, ptr + 2);
  1147. for (i = 0; i < 6; i++) {
  1148. buf[4*(i + 6*0)] = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*0)];
  1149. buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1);
  1150. buf[4*(i + 6*2)] = 0;
  1151. }
  1152. ptr += 18;
  1153. buf += (j&3) != 3 ? 1 : (4*18-3);
  1154. }
  1155. /* zero bands */
  1156. for (j = sblimit; j < SBLIMIT; j++) {
  1157. /* overlap */
  1158. out_ptr = sb_samples + j;
  1159. for (i = 0; i < 18; i++) {
  1160. *out_ptr = buf[4*i];
  1161. buf[4*i] = 0;
  1162. out_ptr += SBLIMIT;
  1163. }
  1164. buf += (j&3) != 3 ? 1 : (4*18-3);
  1165. }
  1166. }
  1167. /* main layer3 decoding function */
  1168. static int mp_decode_layer3(MPADecodeContext *s)
  1169. {
  1170. int nb_granules, main_data_begin;
  1171. int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
  1172. GranuleDef *g;
  1173. int16_t exponents[576]; //FIXME try INTFLOAT
  1174. /* read side info */
  1175. if (s->lsf) {
  1176. main_data_begin = get_bits(&s->gb, 8);
  1177. skip_bits(&s->gb, s->nb_channels);
  1178. nb_granules = 1;
  1179. } else {
  1180. main_data_begin = get_bits(&s->gb, 9);
  1181. if (s->nb_channels == 2)
  1182. skip_bits(&s->gb, 3);
  1183. else
  1184. skip_bits(&s->gb, 5);
  1185. nb_granules = 2;
  1186. for (ch = 0; ch < s->nb_channels; ch++) {
  1187. s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
  1188. s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
  1189. }
  1190. }
  1191. for (gr = 0; gr < nb_granules; gr++) {
  1192. for (ch = 0; ch < s->nb_channels; ch++) {
  1193. av_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
  1194. g = &s->granules[ch][gr];
  1195. g->part2_3_length = get_bits(&s->gb, 12);
  1196. g->big_values = get_bits(&s->gb, 9);
  1197. if (g->big_values > 288) {
  1198. av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
  1199. return AVERROR_INVALIDDATA;
  1200. }
  1201. g->global_gain = get_bits(&s->gb, 8);
  1202. /* if MS stereo only is selected, we precompute the
  1203. 1/sqrt(2) renormalization factor */
  1204. if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
  1205. MODE_EXT_MS_STEREO)
  1206. g->global_gain -= 2;
  1207. if (s->lsf)
  1208. g->scalefac_compress = get_bits(&s->gb, 9);
  1209. else
  1210. g->scalefac_compress = get_bits(&s->gb, 4);
  1211. blocksplit_flag = get_bits1(&s->gb);
  1212. if (blocksplit_flag) {
  1213. g->block_type = get_bits(&s->gb, 2);
  1214. if (g->block_type == 0) {
  1215. av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
  1216. return AVERROR_INVALIDDATA;
  1217. }
  1218. g->switch_point = get_bits1(&s->gb);
  1219. for (i = 0; i < 2; i++)
  1220. g->table_select[i] = get_bits(&s->gb, 5);
  1221. for (i = 0; i < 3; i++)
  1222. g->subblock_gain[i] = get_bits(&s->gb, 3);
  1223. ff_init_short_region(s, g);
  1224. } else {
  1225. int region_address1, region_address2;
  1226. g->block_type = 0;
  1227. g->switch_point = 0;
  1228. for (i = 0; i < 3; i++)
  1229. g->table_select[i] = get_bits(&s->gb, 5);
  1230. /* compute huffman coded region sizes */
  1231. region_address1 = get_bits(&s->gb, 4);
  1232. region_address2 = get_bits(&s->gb, 3);
  1233. av_dlog(s->avctx, "region1=%d region2=%d\n",
  1234. region_address1, region_address2);
  1235. ff_init_long_region(s, g, region_address1, region_address2);
  1236. }
  1237. ff_region_offset2size(g);
  1238. ff_compute_band_indexes(s, g);
  1239. g->preflag = 0;
  1240. if (!s->lsf)
  1241. g->preflag = get_bits1(&s->gb);
  1242. g->scalefac_scale = get_bits1(&s->gb);
  1243. g->count1table_select = get_bits1(&s->gb);
  1244. av_dlog(s->avctx, "block_type=%d switch_point=%d\n",
  1245. g->block_type, g->switch_point);
  1246. }
  1247. }
  1248. if (!s->adu_mode) {
  1249. int skip;
  1250. const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
  1251. int extrasize = av_clip(get_bits_left(&s->gb) >> 3, 0,
  1252. FFMAX(0, LAST_BUF_SIZE - s->last_buf_size));
  1253. assert((get_bits_count(&s->gb) & 7) == 0);
  1254. /* now we get bits from the main_data_begin offset */
  1255. av_dlog(s->avctx, "seekback:%d, lastbuf:%d\n",
  1256. main_data_begin, s->last_buf_size);
  1257. memcpy(s->last_buf + s->last_buf_size, ptr, extrasize);
  1258. s->in_gb = s->gb;
  1259. init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);
  1260. #if !UNCHECKED_BITSTREAM_READER
  1261. s->gb.size_in_bits_plus8 += extrasize * 8;
  1262. #endif
  1263. s->last_buf_size <<= 3;
  1264. for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) {
  1265. for (ch = 0; ch < s->nb_channels; ch++) {
  1266. g = &s->granules[ch][gr];
  1267. s->last_buf_size += g->part2_3_length;
  1268. memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
  1269. compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
  1270. }
  1271. }
  1272. skip = s->last_buf_size - 8 * main_data_begin;
  1273. if (skip >= s->gb.size_in_bits && s->in_gb.buffer) {
  1274. skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits);
  1275. s->gb = s->in_gb;
  1276. s->in_gb.buffer = NULL;
  1277. } else {
  1278. skip_bits_long(&s->gb, skip);
  1279. }
  1280. } else {
  1281. gr = 0;
  1282. }
  1283. for (; gr < nb_granules; gr++) {
  1284. for (ch = 0; ch < s->nb_channels; ch++) {
  1285. g = &s->granules[ch][gr];
  1286. bits_pos = get_bits_count(&s->gb);
  1287. if (!s->lsf) {
  1288. uint8_t *sc;
  1289. int slen, slen1, slen2;
  1290. /* MPEG1 scale factors */
  1291. slen1 = slen_table[0][g->scalefac_compress];
  1292. slen2 = slen_table[1][g->scalefac_compress];
  1293. av_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
  1294. if (g->block_type == 2) {
  1295. n = g->switch_point ? 17 : 18;
  1296. j = 0;
  1297. if (slen1) {
  1298. for (i = 0; i < n; i++)
  1299. g->scale_factors[j++] = get_bits(&s->gb, slen1);
  1300. } else {
  1301. for (i = 0; i < n; i++)
  1302. g->scale_factors[j++] = 0;
  1303. }
  1304. if (slen2) {
  1305. for (i = 0; i < 18; i++)
  1306. g->scale_factors[j++] = get_bits(&s->gb, slen2);
  1307. for (i = 0; i < 3; i++)
  1308. g->scale_factors[j++] = 0;
  1309. } else {
  1310. for (i = 0; i < 21; i++)
  1311. g->scale_factors[j++] = 0;
  1312. }
  1313. } else {
  1314. sc = s->granules[ch][0].scale_factors;
  1315. j = 0;
  1316. for (k = 0; k < 4; k++) {
  1317. n = k == 0 ? 6 : 5;
  1318. if ((g->scfsi & (0x8 >> k)) == 0) {
  1319. slen = (k < 2) ? slen1 : slen2;
  1320. if (slen) {
  1321. for (i = 0; i < n; i++)
  1322. g->scale_factors[j++] = get_bits(&s->gb, slen);
  1323. } else {
  1324. for (i = 0; i < n; i++)
  1325. g->scale_factors[j++] = 0;
  1326. }
  1327. } else {
  1328. /* simply copy from last granule */
  1329. for (i = 0; i < n; i++) {
  1330. g->scale_factors[j] = sc[j];
  1331. j++;
  1332. }
  1333. }
  1334. }
  1335. g->scale_factors[j++] = 0;
  1336. }
  1337. } else {
  1338. int tindex, tindex2, slen[4], sl, sf;
  1339. /* LSF scale factors */
  1340. if (g->block_type == 2)
  1341. tindex = g->switch_point ? 2 : 1;
  1342. else
  1343. tindex = 0;
  1344. sf = g->scalefac_compress;
  1345. if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
  1346. /* intensity stereo case */
  1347. sf >>= 1;
  1348. if (sf < 180) {
  1349. lsf_sf_expand(slen, sf, 6, 6, 0);
  1350. tindex2 = 3;
  1351. } else if (sf < 244) {
  1352. lsf_sf_expand(slen, sf - 180, 4, 4, 0);
  1353. tindex2 = 4;
  1354. } else {
  1355. lsf_sf_expand(slen, sf - 244, 3, 0, 0);
  1356. tindex2 = 5;
  1357. }
  1358. } else {
  1359. /* normal case */
  1360. if (sf < 400) {
  1361. lsf_sf_expand(slen, sf, 5, 4, 4);
  1362. tindex2 = 0;
  1363. } else if (sf < 500) {
  1364. lsf_sf_expand(slen, sf - 400, 5, 4, 0);
  1365. tindex2 = 1;
  1366. } else {
  1367. lsf_sf_expand(slen, sf - 500, 3, 0, 0);
  1368. tindex2 = 2;
  1369. g->preflag = 1;
  1370. }
  1371. }
  1372. j = 0;
  1373. for (k = 0; k < 4; k++) {
  1374. n = lsf_nsf_table[tindex2][tindex][k];
  1375. sl = slen[k];
  1376. if (sl) {
  1377. for (i = 0; i < n; i++)
  1378. g->scale_factors[j++] = get_bits(&s->gb, sl);
  1379. } else {
  1380. for (i = 0; i < n; i++)
  1381. g->scale_factors[j++] = 0;
  1382. }
  1383. }
  1384. /* XXX: should compute exact size */
  1385. for (; j < 40; j++)
  1386. g->scale_factors[j] = 0;
  1387. }
  1388. exponents_from_scale_factors(s, g, exponents);
  1389. /* read Huffman coded residue */
  1390. huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
  1391. } /* ch */
  1392. if (s->mode == MPA_JSTEREO)
  1393. compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
  1394. for (ch = 0; ch < s->nb_channels; ch++) {
  1395. g = &s->granules[ch][gr];
  1396. reorder_block(s, g);
  1397. compute_antialias(s, g);
  1398. compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
  1399. }
  1400. } /* gr */
  1401. if (get_bits_count(&s->gb) < 0)
  1402. skip_bits_long(&s->gb, -get_bits_count(&s->gb));
  1403. return nb_granules * 18;
  1404. }
  1405. static int mp_decode_frame(MPADecodeContext *s, OUT_INT *samples,
  1406. const uint8_t *buf, int buf_size)
  1407. {
  1408. int i, nb_frames, ch, ret;
  1409. OUT_INT *samples_ptr;
  1410. init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);
  1411. /* skip error protection field */
  1412. if (s->error_protection)
  1413. skip_bits(&s->gb, 16);
  1414. switch(s->layer) {
  1415. case 1:
  1416. s->avctx->frame_size = 384;
  1417. nb_frames = mp_decode_layer1(s);
  1418. break;
  1419. case 2:
  1420. s->avctx->frame_size = 1152;
  1421. nb_frames = mp_decode_layer2(s);
  1422. break;
  1423. case 3:
  1424. s->avctx->frame_size = s->lsf ? 576 : 1152;
  1425. default:
  1426. nb_frames = mp_decode_layer3(s);
  1427. if (nb_frames < 0)
  1428. return nb_frames;
  1429. s->last_buf_size=0;
  1430. if (s->in_gb.buffer) {
  1431. align_get_bits(&s->gb);
  1432. i = get_bits_left(&s->gb)>>3;
  1433. if (i >= 0 && i <= BACKSTEP_SIZE) {
  1434. memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);
  1435. s->last_buf_size=i;
  1436. } else
  1437. av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
  1438. s->gb = s->in_gb;
  1439. s->in_gb.buffer = NULL;
  1440. }
  1441. align_get_bits(&s->gb);
  1442. assert((get_bits_count(&s->gb) & 7) == 0);
  1443. i = get_bits_left(&s->gb) >> 3;
  1444. if (i < 0 || i > BACKSTEP_SIZE || nb_frames < 0) {
  1445. if (i < 0)
  1446. av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
  1447. i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
  1448. }
  1449. assert(i <= buf_size - HEADER_SIZE && i >= 0);
  1450. memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
  1451. s->last_buf_size += i;
  1452. }
  1453. /* get output buffer */
  1454. if (!samples) {
  1455. s->frame.nb_samples = s->avctx->frame_size;
  1456. if ((ret = s->avctx->get_buffer(s->avctx, &s->frame)) < 0) {
  1457. av_log(s->avctx, AV_LOG_ERROR, "get_buffer() failed\n");
  1458. return ret;
  1459. }
  1460. samples = (OUT_INT *)s->frame.data[0];
  1461. }
  1462. /* apply the synthesis filter */
  1463. for (ch = 0; ch < s->nb_channels; ch++) {
  1464. samples_ptr = samples + ch;
  1465. for (i = 0; i < nb_frames; i++) {
  1466. RENAME(ff_mpa_synth_filter)(
  1467. &s->mpadsp,
  1468. s->synth_buf[ch], &(s->synth_buf_offset[ch]),
  1469. RENAME(ff_mpa_synth_window), &s->dither_state,
  1470. samples_ptr, s->nb_channels,
  1471. s->sb_samples[ch][i]);
  1472. samples_ptr += 32 * s->nb_channels;
  1473. }
  1474. }
  1475. return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
  1476. }
  1477. static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,
  1478. AVPacket *avpkt)
  1479. {
  1480. const uint8_t *buf = avpkt->data;
  1481. int buf_size = avpkt->size;
  1482. MPADecodeContext *s = avctx->priv_data;
  1483. uint32_t header;
  1484. int ret;
  1485. if (buf_size < HEADER_SIZE)
  1486. return AVERROR_INVALIDDATA;
  1487. header = AV_RB32(buf);
  1488. if (ff_mpa_check_header(header) < 0) {
  1489. av_log(avctx, AV_LOG_ERROR, "Header missing\n");
  1490. return AVERROR_INVALIDDATA;
  1491. }
  1492. if (avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
  1493. /* free format: prepare to compute frame size */
  1494. s->frame_size = -1;
  1495. return AVERROR_INVALIDDATA;
  1496. }
  1497. /* update codec info */
  1498. avctx->channels = s->nb_channels;
  1499. avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
  1500. if (!avctx->bit_rate)
  1501. avctx->bit_rate = s->bit_rate;
  1502. if (s->frame_size <= 0 || s->frame_size > buf_size) {
  1503. av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
  1504. return AVERROR_INVALIDDATA;
  1505. } else if (s->frame_size < buf_size) {
  1506. buf_size= s->frame_size;
  1507. }
  1508. ret = mp_decode_frame(s, NULL, buf, buf_size);
  1509. if (ret >= 0) {
  1510. *got_frame_ptr = 1;
  1511. *(AVFrame *)data = s->frame;
  1512. avctx->sample_rate = s->sample_rate;
  1513. //FIXME maybe move the other codec info stuff from above here too
  1514. } else {
  1515. av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
  1516. /* Only return an error if the bad frame makes up the whole packet or
  1517. * the error is related to buffer management.
  1518. * If there is more data in the packet, just consume the bad frame
  1519. * instead of returning an error, which would discard the whole
  1520. * packet. */
  1521. *got_frame_ptr = 0;
  1522. if (buf_size == avpkt->size || ret != AVERROR_INVALIDDATA)
  1523. return ret;
  1524. }
  1525. s->frame_size = 0;
  1526. return buf_size;
  1527. }
  1528. static void mp_flush(MPADecodeContext *ctx)
  1529. {
  1530. memset(ctx->synth_buf, 0, sizeof(ctx->synth_buf));
  1531. ctx->last_buf_size = 0;
  1532. }
  1533. static void flush(AVCodecContext *avctx)
  1534. {
  1535. mp_flush(avctx->priv_data);
  1536. }
  1537. #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
  1538. static int decode_frame_adu(AVCodecContext *avctx, void *data,
  1539. int *got_frame_ptr, AVPacket *avpkt)
  1540. {
  1541. const uint8_t *buf = avpkt->data;
  1542. int buf_size = avpkt->size;
  1543. MPADecodeContext *s = avctx->priv_data;
  1544. uint32_t header;
  1545. int len, ret;
  1546. len = buf_size;
  1547. // Discard too short frames
  1548. if (buf_size < HEADER_SIZE) {
  1549. av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
  1550. return AVERROR_INVALIDDATA;
  1551. }
  1552. if (len > MPA_MAX_CODED_FRAME_SIZE)
  1553. len = MPA_MAX_CODED_FRAME_SIZE;
  1554. // Get header and restore sync word
  1555. header = AV_RB32(buf) | 0xffe00000;
  1556. if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame
  1557. av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n");
  1558. return AVERROR_INVALIDDATA;
  1559. }
  1560. avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);
  1561. /* update codec info */
  1562. avctx->sample_rate = s->sample_rate;
  1563. avctx->channels = s->nb_channels;
  1564. if (!avctx->bit_rate)
  1565. avctx->bit_rate = s->bit_rate;
  1566. s->frame_size = len;
  1567. ret = mp_decode_frame(s, NULL, buf, buf_size);
  1568. if (ret < 0) {
  1569. av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
  1570. return ret;
  1571. }
  1572. *got_frame_ptr = 1;
  1573. *(AVFrame *)data = s->frame;
  1574. return buf_size;
  1575. }
  1576. #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
  1577. #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
  1578. /**
  1579. * Context for MP3On4 decoder
  1580. */
  1581. typedef struct MP3On4DecodeContext {
  1582. AVFrame *frame;
  1583. int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
  1584. int syncword; ///< syncword patch
  1585. const uint8_t *coff; ///< channel offsets in output buffer
  1586. MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
  1587. OUT_INT *decoded_buf; ///< output buffer for decoded samples
  1588. } MP3On4DecodeContext;
  1589. #include "mpeg4audio.h"
  1590. /* Next 3 arrays are indexed by channel config number (passed via codecdata) */
  1591. /* number of mp3 decoder instances */
  1592. static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 };
  1593. /* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */
  1594. static const uint8_t chan_offset[8][5] = {
  1595. { 0 },
  1596. { 0 }, // C
  1597. { 0 }, // FLR
  1598. { 2, 0 }, // C FLR
  1599. { 2, 0, 3 }, // C FLR BS
  1600. { 2, 0, 3 }, // C FLR BLRS
  1601. { 2, 0, 4, 3 }, // C FLR BLRS LFE
  1602. { 2, 0, 6, 4, 3 }, // C FLR BLRS BLR LFE
  1603. };
  1604. /* mp3on4 channel layouts */
  1605. static const int16_t chan_layout[8] = {
  1606. 0,
  1607. AV_CH_LAYOUT_MONO,
  1608. AV_CH_LAYOUT_STEREO,
  1609. AV_CH_LAYOUT_SURROUND,
  1610. AV_CH_LAYOUT_4POINT0,
  1611. AV_CH_LAYOUT_5POINT0,
  1612. AV_CH_LAYOUT_5POINT1,
  1613. AV_CH_LAYOUT_7POINT1
  1614. };
  1615. static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
  1616. {
  1617. MP3On4DecodeContext *s = avctx->priv_data;
  1618. int i;
  1619. for (i = 0; i < s->frames; i++)
  1620. av_free(s->mp3decctx[i]);
  1621. av_freep(&s->decoded_buf);
  1622. return 0;
  1623. }
  1624. static int decode_init_mp3on4(AVCodecContext * avctx)
  1625. {
  1626. MP3On4DecodeContext *s = avctx->priv_data;
  1627. MPEG4AudioConfig cfg;
  1628. int i;
  1629. if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) {
  1630. av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
  1631. return AVERROR_INVALIDDATA;
  1632. }
  1633. avpriv_mpeg4audio_get_config(&cfg, avctx->extradata,
  1634. avctx->extradata_size * 8, 1);
  1635. if (!cfg.chan_config || cfg.chan_config > 7) {
  1636. av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
  1637. return AVERROR_INVALIDDATA;
  1638. }
  1639. s->frames = mp3Frames[cfg.chan_config];
  1640. s->coff = chan_offset[cfg.chan_config];
  1641. avctx->channels = ff_mpeg4audio_channels[cfg.chan_config];
  1642. avctx->channel_layout = chan_layout[cfg.chan_config];
  1643. if (cfg.sample_rate < 16000)
  1644. s->syncword = 0xffe00000;
  1645. else
  1646. s->syncword = 0xfff00000;
  1647. /* Init the first mp3 decoder in standard way, so that all tables get builded
  1648. * We replace avctx->priv_data with the context of the first decoder so that
  1649. * decode_init() does not have to be changed.
  1650. * Other decoders will be initialized here copying data from the first context
  1651. */
  1652. // Allocate zeroed memory for the first decoder context
  1653. s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
  1654. if (!s->mp3decctx[0])
  1655. goto alloc_fail;
  1656. // Put decoder context in place to make init_decode() happy
  1657. avctx->priv_data = s->mp3decctx[0];
  1658. decode_init(avctx);
  1659. s->frame = avctx->coded_frame;
  1660. // Restore mp3on4 context pointer
  1661. avctx->priv_data = s;
  1662. s->mp3decctx[0]->adu_mode = 1; // Set adu mode
  1663. /* Create a separate codec/context for each frame (first is already ok).
  1664. * Each frame is 1 or 2 channels - up to 5 frames allowed
  1665. */
  1666. for (i = 1; i < s->frames; i++) {
  1667. s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
  1668. if (!s->mp3decctx[i])
  1669. goto alloc_fail;
  1670. s->mp3decctx[i]->adu_mode = 1;
  1671. s->mp3decctx[i]->avctx = avctx;
  1672. s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;
  1673. }
  1674. /* Allocate buffer for multi-channel output if needed */
  1675. if (s->frames > 1) {
  1676. s->decoded_buf = av_malloc(MPA_FRAME_SIZE * MPA_MAX_CHANNELS *
  1677. sizeof(*s->decoded_buf));
  1678. if (!s->decoded_buf)
  1679. goto alloc_fail;
  1680. }
  1681. return 0;
  1682. alloc_fail:
  1683. decode_close_mp3on4(avctx);
  1684. return AVERROR(ENOMEM);
  1685. }
  1686. static void flush_mp3on4(AVCodecContext *avctx)
  1687. {
  1688. int i;
  1689. MP3On4DecodeContext *s = avctx->priv_data;
  1690. for (i = 0; i < s->frames; i++)
  1691. mp_flush(s->mp3decctx[i]);
  1692. }
  1693. static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
  1694. int *got_frame_ptr, AVPacket *avpkt)
  1695. {
  1696. const uint8_t *buf = avpkt->data;
  1697. int buf_size = avpkt->size;
  1698. MP3On4DecodeContext *s = avctx->priv_data;
  1699. MPADecodeContext *m;
  1700. int fsize, len = buf_size, out_size = 0;
  1701. uint32_t header;
  1702. OUT_INT *out_samples;
  1703. OUT_INT *outptr, *bp;
  1704. int fr, j, n, ch, ret;
  1705. /* get output buffer */
  1706. s->frame->nb_samples = MPA_FRAME_SIZE;
  1707. if ((ret = avctx->get_buffer(avctx, s->frame)) < 0) {
  1708. av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
  1709. return ret;
  1710. }
  1711. out_samples = (OUT_INT *)s->frame->data[0];
  1712. // Discard too short frames
  1713. if (buf_size < HEADER_SIZE)
  1714. return AVERROR_INVALIDDATA;
  1715. // If only one decoder interleave is not needed
  1716. outptr = s->frames == 1 ? out_samples : s->decoded_buf;
  1717. avctx->bit_rate = 0;
  1718. ch = 0;
  1719. for (fr = 0; fr < s->frames; fr++) {
  1720. fsize = AV_RB16(buf) >> 4;
  1721. fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
  1722. m = s->mp3decctx[fr];
  1723. assert(m != NULL);
  1724. if (fsize < HEADER_SIZE) {
  1725. av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n");
  1726. return AVERROR_INVALIDDATA;
  1727. }
  1728. header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
  1729. if (ff_mpa_check_header(header) < 0) // Bad header, discard block
  1730. break;
  1731. avpriv_mpegaudio_decode_header((MPADecodeHeader *)m, header);
  1732. if (ch + m->nb_channels > avctx->channels) {
  1733. av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "
  1734. "channel count\n");
  1735. return AVERROR_INVALIDDATA;
  1736. }
  1737. ch += m->nb_channels;
  1738. if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0)
  1739. return ret;
  1740. out_size += ret;
  1741. buf += fsize;
  1742. len -= fsize;
  1743. if (s->frames > 1) {
  1744. n = m->avctx->frame_size*m->nb_channels;
  1745. /* interleave output data */
  1746. bp = out_samples + s->coff[fr];
  1747. if (m->nb_channels == 1) {
  1748. for (j = 0; j < n; j++) {
  1749. *bp = s->decoded_buf[j];
  1750. bp += avctx->channels;
  1751. }
  1752. } else {
  1753. for (j = 0; j < n; j++) {
  1754. bp[0] = s->decoded_buf[j++];
  1755. bp[1] = s->decoded_buf[j];
  1756. bp += avctx->channels;
  1757. }
  1758. }
  1759. }
  1760. avctx->bit_rate += m->bit_rate;
  1761. }
  1762. /* update codec info */
  1763. avctx->sample_rate = s->mp3decctx[0]->sample_rate;
  1764. s->frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));
  1765. *got_frame_ptr = 1;
  1766. *(AVFrame *)data = *s->frame;
  1767. return buf_size;
  1768. }
  1769. #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
  1770. #if !CONFIG_FLOAT
  1771. #if CONFIG_MP1_DECODER
  1772. AVCodec ff_mp1_decoder = {
  1773. .name = "mp1",
  1774. .type = AVMEDIA_TYPE_AUDIO,
  1775. .id = AV_CODEC_ID_MP1,
  1776. .priv_data_size = sizeof(MPADecodeContext),
  1777. .init = decode_init,
  1778. .decode = decode_frame,
  1779. .capabilities = CODEC_CAP_DR1,
  1780. .flush = flush,
  1781. .long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
  1782. };
  1783. #endif
  1784. #if CONFIG_MP2_DECODER
  1785. AVCodec ff_mp2_decoder = {
  1786. .name = "mp2",
  1787. .type = AVMEDIA_TYPE_AUDIO,
  1788. .id = AV_CODEC_ID_MP2,
  1789. .priv_data_size = sizeof(MPADecodeContext),
  1790. .init = decode_init,
  1791. .decode = decode_frame,
  1792. .capabilities = CODEC_CAP_DR1,
  1793. .flush = flush,
  1794. .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
  1795. };
  1796. #endif
  1797. #if CONFIG_MP3_DECODER
  1798. AVCodec ff_mp3_decoder = {
  1799. .name = "mp3",
  1800. .type = AVMEDIA_TYPE_AUDIO,
  1801. .id = AV_CODEC_ID_MP3,
  1802. .priv_data_size = sizeof(MPADecodeContext),
  1803. .init = decode_init,
  1804. .decode = decode_frame,
  1805. .capabilities = CODEC_CAP_DR1,
  1806. .flush = flush,
  1807. .long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
  1808. };
  1809. #endif
  1810. #if CONFIG_MP3ADU_DECODER
  1811. AVCodec ff_mp3adu_decoder = {
  1812. .name = "mp3adu",
  1813. .type = AVMEDIA_TYPE_AUDIO,
  1814. .id = AV_CODEC_ID_MP3ADU,
  1815. .priv_data_size = sizeof(MPADecodeContext),
  1816. .init = decode_init,
  1817. .decode = decode_frame_adu,
  1818. .capabilities = CODEC_CAP_DR1,
  1819. .flush = flush,
  1820. .long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
  1821. };
  1822. #endif
  1823. #if CONFIG_MP3ON4_DECODER
  1824. AVCodec ff_mp3on4_decoder = {
  1825. .name = "mp3on4",
  1826. .type = AVMEDIA_TYPE_AUDIO,
  1827. .id = AV_CODEC_ID_MP3ON4,
  1828. .priv_data_size = sizeof(MP3On4DecodeContext),
  1829. .init = decode_init_mp3on4,
  1830. .close = decode_close_mp3on4,
  1831. .decode = decode_frame_mp3on4,
  1832. .capabilities = CODEC_CAP_DR1,
  1833. .flush = flush_mp3on4,
  1834. .long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
  1835. };
  1836. #endif
  1837. #endif