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  1. /*
  2. * MPEG Audio decoder
  3. * Copyright (c) 2001, 2002 Fabrice Bellard
  4. *
  5. * This file is part of FFmpeg.
  6. *
  7. * FFmpeg is free software; you can redistribute it and/or
  8. * modify it under the terms of the GNU Lesser General Public
  9. * License as published by the Free Software Foundation; either
  10. * version 2.1 of the License, or (at your option) any later version.
  11. *
  12. * FFmpeg is distributed in the hope that it will be useful,
  13. * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14. * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
  15. * Lesser General Public License for more details.
  16. *
  17. * You should have received a copy of the GNU Lesser General Public
  18. * License along with FFmpeg; if not, write to the Free Software
  19. * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20. */
  21. /**
  22. * @file
  23. * MPEG Audio decoder
  24. */
  25. #include "libavutil/attributes.h"
  26. #include "libavutil/avassert.h"
  27. #include "libavutil/channel_layout.h"
  28. #include "libavutil/crc.h"
  29. #include "libavutil/float_dsp.h"
  30. #include "libavutil/libm.h"
  31. #include "avcodec.h"
  32. #include "get_bits.h"
  33. #include "internal.h"
  34. #include "mathops.h"
  35. #include "mpegaudiodsp.h"
  36. /*
  37. * TODO:
  38. * - test lsf / mpeg25 extensively.
  39. */
  40. #include "mpegaudio.h"
  41. #include "mpegaudiodecheader.h"
  42. #define BACKSTEP_SIZE 512
  43. #define EXTRABYTES 24
  44. #define LAST_BUF_SIZE 2 * BACKSTEP_SIZE + EXTRABYTES
  45. /* layer 3 "granule" */
  46. typedef struct GranuleDef {
  47. uint8_t scfsi;
  48. int part2_3_length;
  49. int big_values;
  50. int global_gain;
  51. int scalefac_compress;
  52. uint8_t block_type;
  53. uint8_t switch_point;
  54. int table_select[3];
  55. int subblock_gain[3];
  56. uint8_t scalefac_scale;
  57. uint8_t count1table_select;
  58. int region_size[3]; /* number of huffman codes in each region */
  59. int preflag;
  60. int short_start, long_end; /* long/short band indexes */
  61. uint8_t scale_factors[40];
  62. DECLARE_ALIGNED(16, INTFLOAT, sb_hybrid)[SBLIMIT * 18]; /* 576 samples */
  63. } GranuleDef;
  64. typedef struct MPADecodeContext {
  65. MPA_DECODE_HEADER
  66. uint8_t last_buf[LAST_BUF_SIZE];
  67. int last_buf_size;
  68. int extrasize;
  69. /* next header (used in free format parsing) */
  70. uint32_t free_format_next_header;
  71. GetBitContext gb;
  72. GetBitContext in_gb;
  73. DECLARE_ALIGNED(32, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512 * 2];
  74. int synth_buf_offset[MPA_MAX_CHANNELS];
  75. DECLARE_ALIGNED(32, INTFLOAT, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT];
  76. INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
  77. GranuleDef granules[2][2]; /* Used in Layer 3 */
  78. int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
  79. int dither_state;
  80. int err_recognition;
  81. AVCodecContext* avctx;
  82. MPADSPContext mpadsp;
  83. void (*butterflies_float)(float *av_restrict v1, float *av_restrict v2, int len);
  84. AVFrame *frame;
  85. uint32_t crc;
  86. } MPADecodeContext;
  87. #define HEADER_SIZE 4
  88. #include "mpegaudiodata.h"
  89. #include "mpegaudiodectab.h"
  90. /* vlc structure for decoding layer 3 huffman tables */
  91. static VLC huff_vlc[16];
  92. static VLC_TYPE huff_vlc_tables[
  93. 0 + 128 + 128 + 128 + 130 + 128 + 154 + 166 +
  94. 142 + 204 + 190 + 170 + 542 + 460 + 662 + 414
  95. ][2];
  96. static const int huff_vlc_tables_sizes[16] = {
  97. 0, 128, 128, 128, 130, 128, 154, 166,
  98. 142, 204, 190, 170, 542, 460, 662, 414
  99. };
  100. static VLC huff_quad_vlc[2];
  101. static VLC_TYPE huff_quad_vlc_tables[128+16][2];
  102. static const int huff_quad_vlc_tables_sizes[2] = { 128, 16 };
  103. /* computed from band_size_long */
  104. static uint16_t band_index_long[9][23];
  105. #include "mpegaudio_tablegen.h"
  106. /* intensity stereo coef table */
  107. static INTFLOAT is_table[2][16];
  108. static INTFLOAT is_table_lsf[2][2][16];
  109. static INTFLOAT csa_table[8][4];
  110. static int16_t division_tab3[1 << 6 ];
  111. static int16_t division_tab5[1 << 8 ];
  112. static int16_t division_tab9[1 << 11];
  113. static int16_t * const division_tabs[4] = {
  114. division_tab3, division_tab5, NULL, division_tab9
  115. };
  116. /* lower 2 bits: modulo 3, higher bits: shift */
  117. static uint16_t scale_factor_modshift[64];
  118. /* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
  119. static int32_t scale_factor_mult[15][3];
  120. /* mult table for layer 2 group quantization */
  121. #define SCALE_GEN(v) \
  122. { FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
  123. static const int32_t scale_factor_mult2[3][3] = {
  124. SCALE_GEN(4.0 / 3.0), /* 3 steps */
  125. SCALE_GEN(4.0 / 5.0), /* 5 steps */
  126. SCALE_GEN(4.0 / 9.0), /* 9 steps */
  127. };
  128. /**
  129. * Convert region offsets to region sizes and truncate
  130. * size to big_values.
  131. */
  132. static void region_offset2size(GranuleDef *g)
  133. {
  134. int i, k, j = 0;
  135. g->region_size[2] = 576 / 2;
  136. for (i = 0; i < 3; i++) {
  137. k = FFMIN(g->region_size[i], g->big_values);
  138. g->region_size[i] = k - j;
  139. j = k;
  140. }
  141. }
  142. static void init_short_region(MPADecodeContext *s, GranuleDef *g)
  143. {
  144. if (g->block_type == 2) {
  145. if (s->sample_rate_index != 8)
  146. g->region_size[0] = (36 / 2);
  147. else
  148. g->region_size[0] = (72 / 2);
  149. } else {
  150. if (s->sample_rate_index <= 2)
  151. g->region_size[0] = (36 / 2);
  152. else if (s->sample_rate_index != 8)
  153. g->region_size[0] = (54 / 2);
  154. else
  155. g->region_size[0] = (108 / 2);
  156. }
  157. g->region_size[1] = (576 / 2);
  158. }
  159. static void init_long_region(MPADecodeContext *s, GranuleDef *g,
  160. int ra1, int ra2)
  161. {
  162. int l;
  163. g->region_size[0] = band_index_long[s->sample_rate_index][ra1 + 1] >> 1;
  164. /* should not overflow */
  165. l = FFMIN(ra1 + ra2 + 2, 22);
  166. g->region_size[1] = band_index_long[s->sample_rate_index][ l] >> 1;
  167. }
  168. static void compute_band_indexes(MPADecodeContext *s, GranuleDef *g)
  169. {
  170. if (g->block_type == 2) {
  171. if (g->switch_point) {
  172. if(s->sample_rate_index == 8)
  173. avpriv_request_sample(s->avctx, "switch point in 8khz");
  174. /* if switched mode, we handle the 36 first samples as
  175. long blocks. For 8000Hz, we handle the 72 first
  176. exponents as long blocks */
  177. if (s->sample_rate_index <= 2)
  178. g->long_end = 8;
  179. else
  180. g->long_end = 6;
  181. g->short_start = 3;
  182. } else {
  183. g->long_end = 0;
  184. g->short_start = 0;
  185. }
  186. } else {
  187. g->short_start = 13;
  188. g->long_end = 22;
  189. }
  190. }
  191. /* layer 1 unscaling */
  192. /* n = number of bits of the mantissa minus 1 */
  193. static inline int l1_unscale(int n, int mant, int scale_factor)
  194. {
  195. int shift, mod;
  196. int64_t val;
  197. shift = scale_factor_modshift[scale_factor];
  198. mod = shift & 3;
  199. shift >>= 2;
  200. val = MUL64((int)(mant + (-1U << n) + 1), scale_factor_mult[n-1][mod]);
  201. shift += n;
  202. /* NOTE: at this point, 1 <= shift >= 21 + 15 */
  203. return (int)((val + (1LL << (shift - 1))) >> shift);
  204. }
  205. static inline int l2_unscale_group(int steps, int mant, int scale_factor)
  206. {
  207. int shift, mod, val;
  208. shift = scale_factor_modshift[scale_factor];
  209. mod = shift & 3;
  210. shift >>= 2;
  211. val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
  212. /* NOTE: at this point, 0 <= shift <= 21 */
  213. if (shift > 0)
  214. val = (val + (1 << (shift - 1))) >> shift;
  215. return val;
  216. }
  217. /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
  218. static inline int l3_unscale(int value, int exponent)
  219. {
  220. unsigned int m;
  221. int e;
  222. e = table_4_3_exp [4 * value + (exponent & 3)];
  223. m = table_4_3_value[4 * value + (exponent & 3)];
  224. e -= exponent >> 2;
  225. #ifdef DEBUG
  226. if(e < 1)
  227. av_log(NULL, AV_LOG_WARNING, "l3_unscale: e is %d\n", e);
  228. #endif
  229. if (e > (SUINT)31)
  230. return 0;
  231. m = (m + ((1U << e) >> 1)) >> e;
  232. return m;
  233. }
  234. static av_cold void decode_init_static(void)
  235. {
  236. int i, j, k;
  237. int offset;
  238. /* scale factors table for layer 1/2 */
  239. for (i = 0; i < 64; i++) {
  240. int shift, mod;
  241. /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
  242. shift = i / 3;
  243. mod = i % 3;
  244. scale_factor_modshift[i] = mod | (shift << 2);
  245. }
  246. /* scale factor multiply for layer 1 */
  247. for (i = 0; i < 15; i++) {
  248. int n, norm;
  249. n = i + 2;
  250. norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
  251. scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
  252. scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
  253. scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
  254. ff_dlog(NULL, "%d: norm=%x s=%"PRIx32" %"PRIx32" %"PRIx32"\n", i,
  255. (unsigned)norm,
  256. scale_factor_mult[i][0],
  257. scale_factor_mult[i][1],
  258. scale_factor_mult[i][2]);
  259. }
  260. /* huffman decode tables */
  261. offset = 0;
  262. for (i = 1; i < 16; i++) {
  263. const HuffTable *h = &mpa_huff_tables[i];
  264. int xsize, x, y;
  265. uint8_t tmp_bits [512] = { 0 };
  266. uint16_t tmp_codes[512] = { 0 };
  267. xsize = h->xsize;
  268. j = 0;
  269. for (x = 0; x < xsize; x++) {
  270. for (y = 0; y < xsize; y++) {
  271. tmp_bits [(x << 5) | y | ((x && y) << 4)]= h->bits [j ];
  272. tmp_codes[(x << 5) | y | ((x && y) << 4)]= h->codes[j++];
  273. }
  274. }
  275. /* XXX: fail test */
  276. huff_vlc[i].table = huff_vlc_tables+offset;
  277. huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
  278. init_vlc(&huff_vlc[i], 7, 512,
  279. tmp_bits, 1, 1, tmp_codes, 2, 2,
  280. INIT_VLC_USE_NEW_STATIC);
  281. offset += huff_vlc_tables_sizes[i];
  282. }
  283. av_assert0(offset == FF_ARRAY_ELEMS(huff_vlc_tables));
  284. offset = 0;
  285. for (i = 0; i < 2; i++) {
  286. huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
  287. huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
  288. init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
  289. mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
  290. INIT_VLC_USE_NEW_STATIC);
  291. offset += huff_quad_vlc_tables_sizes[i];
  292. }
  293. av_assert0(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));
  294. for (i = 0; i < 9; i++) {
  295. k = 0;
  296. for (j = 0; j < 22; j++) {
  297. band_index_long[i][j] = k;
  298. k += band_size_long[i][j];
  299. }
  300. band_index_long[i][22] = k;
  301. }
  302. /* compute n ^ (4/3) and store it in mantissa/exp format */
  303. mpegaudio_tableinit();
  304. for (i = 0; i < 4; i++) {
  305. if (ff_mpa_quant_bits[i] < 0) {
  306. for (j = 0; j < (1 << (-ff_mpa_quant_bits[i] + 1)); j++) {
  307. int val1, val2, val3, steps;
  308. int val = j;
  309. steps = ff_mpa_quant_steps[i];
  310. val1 = val % steps;
  311. val /= steps;
  312. val2 = val % steps;
  313. val3 = val / steps;
  314. division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);
  315. }
  316. }
  317. }
  318. for (i = 0; i < 7; i++) {
  319. float f;
  320. INTFLOAT v;
  321. if (i != 6) {
  322. f = tan((double)i * M_PI / 12.0);
  323. v = FIXR(f / (1.0 + f));
  324. } else {
  325. v = FIXR(1.0);
  326. }
  327. is_table[0][ i] = v;
  328. is_table[1][6 - i] = v;
  329. }
  330. /* invalid values */
  331. for (i = 7; i < 16; i++)
  332. is_table[0][i] = is_table[1][i] = 0.0;
  333. for (i = 0; i < 16; i++) {
  334. double f;
  335. int e, k;
  336. for (j = 0; j < 2; j++) {
  337. e = -(j + 1) * ((i + 1) >> 1);
  338. f = exp2(e / 4.0);
  339. k = i & 1;
  340. is_table_lsf[j][k ^ 1][i] = FIXR(f);
  341. is_table_lsf[j][k ][i] = FIXR(1.0);
  342. ff_dlog(NULL, "is_table_lsf %d %d: %f %f\n",
  343. i, j, (float) is_table_lsf[j][0][i],
  344. (float) is_table_lsf[j][1][i]);
  345. }
  346. }
  347. for (i = 0; i < 8; i++) {
  348. double ci, cs, ca;
  349. ci = ci_table[i];
  350. cs = 1.0 / sqrt(1.0 + ci * ci);
  351. ca = cs * ci;
  352. #if !USE_FLOATS
  353. csa_table[i][0] = FIXHR(cs/4);
  354. csa_table[i][1] = FIXHR(ca/4);
  355. csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);
  356. csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);
  357. #else
  358. csa_table[i][0] = cs;
  359. csa_table[i][1] = ca;
  360. csa_table[i][2] = ca + cs;
  361. csa_table[i][3] = ca - cs;
  362. #endif
  363. }
  364. RENAME(ff_mpa_synth_init)();
  365. }
  366. static av_cold int decode_init(AVCodecContext * avctx)
  367. {
  368. static int initialized_tables = 0;
  369. MPADecodeContext *s = avctx->priv_data;
  370. if (!initialized_tables) {
  371. decode_init_static();
  372. initialized_tables = 1;
  373. }
  374. s->avctx = avctx;
  375. #if USE_FLOATS
  376. {
  377. AVFloatDSPContext *fdsp;
  378. fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
  379. if (!fdsp)
  380. return AVERROR(ENOMEM);
  381. s->butterflies_float = fdsp->butterflies_float;
  382. av_free(fdsp);
  383. }
  384. #endif
  385. ff_mpadsp_init(&s->mpadsp);
  386. if (avctx->request_sample_fmt == OUT_FMT &&
  387. avctx->codec_id != AV_CODEC_ID_MP3ON4)
  388. avctx->sample_fmt = OUT_FMT;
  389. else
  390. avctx->sample_fmt = OUT_FMT_P;
  391. s->err_recognition = avctx->err_recognition;
  392. if (avctx->codec_id == AV_CODEC_ID_MP3ADU)
  393. s->adu_mode = 1;
  394. return 0;
  395. }
  396. #define C3 FIXHR(0.86602540378443864676/2)
  397. #define C4 FIXHR(0.70710678118654752439/2) //0.5 / cos(pi*(9)/36)
  398. #define C5 FIXHR(0.51763809020504152469/2) //0.5 / cos(pi*(5)/36)
  399. #define C6 FIXHR(1.93185165257813657349/4) //0.5 / cos(pi*(15)/36)
  400. /* 12 points IMDCT. We compute it "by hand" by factorizing obvious
  401. cases. */
  402. static void imdct12(INTFLOAT *out, SUINTFLOAT *in)
  403. {
  404. SUINTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
  405. in0 = in[0*3];
  406. in1 = in[1*3] + in[0*3];
  407. in2 = in[2*3] + in[1*3];
  408. in3 = in[3*3] + in[2*3];
  409. in4 = in[4*3] + in[3*3];
  410. in5 = in[5*3] + in[4*3];
  411. in5 += in3;
  412. in3 += in1;
  413. in2 = MULH3(in2, C3, 2);
  414. in3 = MULH3(in3, C3, 4);
  415. t1 = in0 - in4;
  416. t2 = MULH3(in1 - in5, C4, 2);
  417. out[ 7] =
  418. out[10] = t1 + t2;
  419. out[ 1] =
  420. out[ 4] = t1 - t2;
  421. in0 += SHR(in4, 1);
  422. in4 = in0 + in2;
  423. in5 += 2*in1;
  424. in1 = MULH3(in5 + in3, C5, 1);
  425. out[ 8] =
  426. out[ 9] = in4 + in1;
  427. out[ 2] =
  428. out[ 3] = in4 - in1;
  429. in0 -= in2;
  430. in5 = MULH3(in5 - in3, C6, 2);
  431. out[ 0] =
  432. out[ 5] = in0 - in5;
  433. out[ 6] =
  434. out[11] = in0 + in5;
  435. }
  436. static int handle_crc(MPADecodeContext *s, int sec_len)
  437. {
  438. if (s->error_protection && (s->err_recognition & AV_EF_CRCCHECK)) {
  439. const uint8_t *buf = s->gb.buffer - HEADER_SIZE;
  440. int sec_byte_len = sec_len >> 3;
  441. int sec_rem_bits = sec_len & 7;
  442. const AVCRC *crc_tab = av_crc_get_table(AV_CRC_16_ANSI);
  443. uint8_t tmp_buf[4];
  444. uint32_t crc_val = av_crc(crc_tab, UINT16_MAX, &buf[2], 2);
  445. crc_val = av_crc(crc_tab, crc_val, &buf[6], sec_byte_len);
  446. AV_WB32(tmp_buf,
  447. ((buf[6 + sec_byte_len] & (0xFF00 >> sec_rem_bits)) << 24) +
  448. ((s->crc << 16) >> sec_rem_bits));
  449. crc_val = av_crc(crc_tab, crc_val, tmp_buf, 3);
  450. if (crc_val) {
  451. av_log(s->avctx, AV_LOG_ERROR, "CRC mismatch %X!\n", crc_val);
  452. if (s->err_recognition & AV_EF_EXPLODE)
  453. return AVERROR_INVALIDDATA;
  454. }
  455. }
  456. return 0;
  457. }
  458. /* return the number of decoded frames */
  459. static int mp_decode_layer1(MPADecodeContext *s)
  460. {
  461. int bound, i, v, n, ch, j, mant;
  462. uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
  463. uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
  464. int ret;
  465. ret = handle_crc(s, (s->nb_channels == 1) ? 8*16 : 8*32);
  466. if (ret < 0)
  467. return ret;
  468. if (s->mode == MPA_JSTEREO)
  469. bound = (s->mode_ext + 1) * 4;
  470. else
  471. bound = SBLIMIT;
  472. /* allocation bits */
  473. for (i = 0; i < bound; i++) {
  474. for (ch = 0; ch < s->nb_channels; ch++) {
  475. allocation[ch][i] = get_bits(&s->gb, 4);
  476. }
  477. }
  478. for (i = bound; i < SBLIMIT; i++)
  479. allocation[0][i] = get_bits(&s->gb, 4);
  480. /* scale factors */
  481. for (i = 0; i < bound; i++) {
  482. for (ch = 0; ch < s->nb_channels; ch++) {
  483. if (allocation[ch][i])
  484. scale_factors[ch][i] = get_bits(&s->gb, 6);
  485. }
  486. }
  487. for (i = bound; i < SBLIMIT; i++) {
  488. if (allocation[0][i]) {
  489. scale_factors[0][i] = get_bits(&s->gb, 6);
  490. scale_factors[1][i] = get_bits(&s->gb, 6);
  491. }
  492. }
  493. /* compute samples */
  494. for (j = 0; j < 12; j++) {
  495. for (i = 0; i < bound; i++) {
  496. for (ch = 0; ch < s->nb_channels; ch++) {
  497. n = allocation[ch][i];
  498. if (n) {
  499. mant = get_bits(&s->gb, n + 1);
  500. v = l1_unscale(n, mant, scale_factors[ch][i]);
  501. } else {
  502. v = 0;
  503. }
  504. s->sb_samples[ch][j][i] = v;
  505. }
  506. }
  507. for (i = bound; i < SBLIMIT; i++) {
  508. n = allocation[0][i];
  509. if (n) {
  510. mant = get_bits(&s->gb, n + 1);
  511. v = l1_unscale(n, mant, scale_factors[0][i]);
  512. s->sb_samples[0][j][i] = v;
  513. v = l1_unscale(n, mant, scale_factors[1][i]);
  514. s->sb_samples[1][j][i] = v;
  515. } else {
  516. s->sb_samples[0][j][i] = 0;
  517. s->sb_samples[1][j][i] = 0;
  518. }
  519. }
  520. }
  521. return 12;
  522. }
  523. static int mp_decode_layer2(MPADecodeContext *s)
  524. {
  525. int sblimit; /* number of used subbands */
  526. const unsigned char *alloc_table;
  527. int table, bit_alloc_bits, i, j, ch, bound, v;
  528. unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
  529. unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
  530. unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
  531. int scale, qindex, bits, steps, k, l, m, b;
  532. int ret;
  533. /* select decoding table */
  534. table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
  535. s->sample_rate, s->lsf);
  536. sblimit = ff_mpa_sblimit_table[table];
  537. alloc_table = ff_mpa_alloc_tables[table];
  538. if (s->mode == MPA_JSTEREO)
  539. bound = (s->mode_ext + 1) * 4;
  540. else
  541. bound = sblimit;
  542. ff_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
  543. /* sanity check */
  544. if (bound > sblimit)
  545. bound = sblimit;
  546. /* parse bit allocation */
  547. j = 0;
  548. for (i = 0; i < bound; i++) {
  549. bit_alloc_bits = alloc_table[j];
  550. for (ch = 0; ch < s->nb_channels; ch++)
  551. bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
  552. j += 1 << bit_alloc_bits;
  553. }
  554. for (i = bound; i < sblimit; i++) {
  555. bit_alloc_bits = alloc_table[j];
  556. v = get_bits(&s->gb, bit_alloc_bits);
  557. bit_alloc[0][i] = v;
  558. bit_alloc[1][i] = v;
  559. j += 1 << bit_alloc_bits;
  560. }
  561. /* scale codes */
  562. for (i = 0; i < sblimit; i++) {
  563. for (ch = 0; ch < s->nb_channels; ch++) {
  564. if (bit_alloc[ch][i])
  565. scale_code[ch][i] = get_bits(&s->gb, 2);
  566. }
  567. }
  568. ret = handle_crc(s, get_bits_count(&s->gb) - 16);
  569. if (ret < 0)
  570. return ret;
  571. /* scale factors */
  572. for (i = 0; i < sblimit; i++) {
  573. for (ch = 0; ch < s->nb_channels; ch++) {
  574. if (bit_alloc[ch][i]) {
  575. sf = scale_factors[ch][i];
  576. switch (scale_code[ch][i]) {
  577. default:
  578. case 0:
  579. sf[0] = get_bits(&s->gb, 6);
  580. sf[1] = get_bits(&s->gb, 6);
  581. sf[2] = get_bits(&s->gb, 6);
  582. break;
  583. case 2:
  584. sf[0] = get_bits(&s->gb, 6);
  585. sf[1] = sf[0];
  586. sf[2] = sf[0];
  587. break;
  588. case 1:
  589. sf[0] = get_bits(&s->gb, 6);
  590. sf[2] = get_bits(&s->gb, 6);
  591. sf[1] = sf[0];
  592. break;
  593. case 3:
  594. sf[0] = get_bits(&s->gb, 6);
  595. sf[2] = get_bits(&s->gb, 6);
  596. sf[1] = sf[2];
  597. break;
  598. }
  599. }
  600. }
  601. }
  602. /* samples */
  603. for (k = 0; k < 3; k++) {
  604. for (l = 0; l < 12; l += 3) {
  605. j = 0;
  606. for (i = 0; i < bound; i++) {
  607. bit_alloc_bits = alloc_table[j];
  608. for (ch = 0; ch < s->nb_channels; ch++) {
  609. b = bit_alloc[ch][i];
  610. if (b) {
  611. scale = scale_factors[ch][i][k];
  612. qindex = alloc_table[j+b];
  613. bits = ff_mpa_quant_bits[qindex];
  614. if (bits < 0) {
  615. int v2;
  616. /* 3 values at the same time */
  617. v = get_bits(&s->gb, -bits);
  618. v2 = division_tabs[qindex][v];
  619. steps = ff_mpa_quant_steps[qindex];
  620. s->sb_samples[ch][k * 12 + l + 0][i] =
  621. l2_unscale_group(steps, v2 & 15, scale);
  622. s->sb_samples[ch][k * 12 + l + 1][i] =
  623. l2_unscale_group(steps, (v2 >> 4) & 15, scale);
  624. s->sb_samples[ch][k * 12 + l + 2][i] =
  625. l2_unscale_group(steps, v2 >> 8 , scale);
  626. } else {
  627. for (m = 0; m < 3; m++) {
  628. v = get_bits(&s->gb, bits);
  629. v = l1_unscale(bits - 1, v, scale);
  630. s->sb_samples[ch][k * 12 + l + m][i] = v;
  631. }
  632. }
  633. } else {
  634. s->sb_samples[ch][k * 12 + l + 0][i] = 0;
  635. s->sb_samples[ch][k * 12 + l + 1][i] = 0;
  636. s->sb_samples[ch][k * 12 + l + 2][i] = 0;
  637. }
  638. }
  639. /* next subband in alloc table */
  640. j += 1 << bit_alloc_bits;
  641. }
  642. /* XXX: find a way to avoid this duplication of code */
  643. for (i = bound; i < sblimit; i++) {
  644. bit_alloc_bits = alloc_table[j];
  645. b = bit_alloc[0][i];
  646. if (b) {
  647. int mant, scale0, scale1;
  648. scale0 = scale_factors[0][i][k];
  649. scale1 = scale_factors[1][i][k];
  650. qindex = alloc_table[j + b];
  651. bits = ff_mpa_quant_bits[qindex];
  652. if (bits < 0) {
  653. /* 3 values at the same time */
  654. v = get_bits(&s->gb, -bits);
  655. steps = ff_mpa_quant_steps[qindex];
  656. mant = v % steps;
  657. v = v / steps;
  658. s->sb_samples[0][k * 12 + l + 0][i] =
  659. l2_unscale_group(steps, mant, scale0);
  660. s->sb_samples[1][k * 12 + l + 0][i] =
  661. l2_unscale_group(steps, mant, scale1);
  662. mant = v % steps;
  663. v = v / steps;
  664. s->sb_samples[0][k * 12 + l + 1][i] =
  665. l2_unscale_group(steps, mant, scale0);
  666. s->sb_samples[1][k * 12 + l + 1][i] =
  667. l2_unscale_group(steps, mant, scale1);
  668. s->sb_samples[0][k * 12 + l + 2][i] =
  669. l2_unscale_group(steps, v, scale0);
  670. s->sb_samples[1][k * 12 + l + 2][i] =
  671. l2_unscale_group(steps, v, scale1);
  672. } else {
  673. for (m = 0; m < 3; m++) {
  674. mant = get_bits(&s->gb, bits);
  675. s->sb_samples[0][k * 12 + l + m][i] =
  676. l1_unscale(bits - 1, mant, scale0);
  677. s->sb_samples[1][k * 12 + l + m][i] =
  678. l1_unscale(bits - 1, mant, scale1);
  679. }
  680. }
  681. } else {
  682. s->sb_samples[0][k * 12 + l + 0][i] = 0;
  683. s->sb_samples[0][k * 12 + l + 1][i] = 0;
  684. s->sb_samples[0][k * 12 + l + 2][i] = 0;
  685. s->sb_samples[1][k * 12 + l + 0][i] = 0;
  686. s->sb_samples[1][k * 12 + l + 1][i] = 0;
  687. s->sb_samples[1][k * 12 + l + 2][i] = 0;
  688. }
  689. /* next subband in alloc table */
  690. j += 1 << bit_alloc_bits;
  691. }
  692. /* fill remaining samples to zero */
  693. for (i = sblimit; i < SBLIMIT; i++) {
  694. for (ch = 0; ch < s->nb_channels; ch++) {
  695. s->sb_samples[ch][k * 12 + l + 0][i] = 0;
  696. s->sb_samples[ch][k * 12 + l + 1][i] = 0;
  697. s->sb_samples[ch][k * 12 + l + 2][i] = 0;
  698. }
  699. }
  700. }
  701. }
  702. return 3 * 12;
  703. }
  704. #define SPLIT(dst,sf,n) \
  705. if (n == 3) { \
  706. int m = (sf * 171) >> 9; \
  707. dst = sf - 3 * m; \
  708. sf = m; \
  709. } else if (n == 4) { \
  710. dst = sf & 3; \
  711. sf >>= 2; \
  712. } else if (n == 5) { \
  713. int m = (sf * 205) >> 10; \
  714. dst = sf - 5 * m; \
  715. sf = m; \
  716. } else if (n == 6) { \
  717. int m = (sf * 171) >> 10; \
  718. dst = sf - 6 * m; \
  719. sf = m; \
  720. } else { \
  721. dst = 0; \
  722. }
  723. static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2,
  724. int n3)
  725. {
  726. SPLIT(slen[3], sf, n3)
  727. SPLIT(slen[2], sf, n2)
  728. SPLIT(slen[1], sf, n1)
  729. slen[0] = sf;
  730. }
  731. static void exponents_from_scale_factors(MPADecodeContext *s, GranuleDef *g,
  732. int16_t *exponents)
  733. {
  734. const uint8_t *bstab, *pretab;
  735. int len, i, j, k, l, v0, shift, gain, gains[3];
  736. int16_t *exp_ptr;
  737. exp_ptr = exponents;
  738. gain = g->global_gain - 210;
  739. shift = g->scalefac_scale + 1;
  740. bstab = band_size_long[s->sample_rate_index];
  741. pretab = mpa_pretab[g->preflag];
  742. for (i = 0; i < g->long_end; i++) {
  743. v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
  744. len = bstab[i];
  745. for (j = len; j > 0; j--)
  746. *exp_ptr++ = v0;
  747. }
  748. if (g->short_start < 13) {
  749. bstab = band_size_short[s->sample_rate_index];
  750. gains[0] = gain - (g->subblock_gain[0] << 3);
  751. gains[1] = gain - (g->subblock_gain[1] << 3);
  752. gains[2] = gain - (g->subblock_gain[2] << 3);
  753. k = g->long_end;
  754. for (i = g->short_start; i < 13; i++) {
  755. len = bstab[i];
  756. for (l = 0; l < 3; l++) {
  757. v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
  758. for (j = len; j > 0; j--)
  759. *exp_ptr++ = v0;
  760. }
  761. }
  762. }
  763. }
  764. static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos,
  765. int *end_pos2)
  766. {
  767. if (s->in_gb.buffer && *pos >= s->gb.size_in_bits - s->extrasize * 8) {
  768. s->gb = s->in_gb;
  769. s->in_gb.buffer = NULL;
  770. s->extrasize = 0;
  771. av_assert2((get_bits_count(&s->gb) & 7) == 0);
  772. skip_bits_long(&s->gb, *pos - *end_pos);
  773. *end_pos2 =
  774. *end_pos = *end_pos2 + get_bits_count(&s->gb) - *pos;
  775. *pos = get_bits_count(&s->gb);
  776. }
  777. }
  778. /* Following is an optimized code for
  779. INTFLOAT v = *src
  780. if(get_bits1(&s->gb))
  781. v = -v;
  782. *dst = v;
  783. */
  784. #if USE_FLOATS
  785. #define READ_FLIP_SIGN(dst,src) \
  786. v = AV_RN32A(src) ^ (get_bits1(&s->gb) << 31); \
  787. AV_WN32A(dst, v);
  788. #else
  789. #define READ_FLIP_SIGN(dst,src) \
  790. v = -get_bits1(&s->gb); \
  791. *(dst) = (*(src) ^ v) - v;
  792. #endif
  793. static int huffman_decode(MPADecodeContext *s, GranuleDef *g,
  794. int16_t *exponents, int end_pos2)
  795. {
  796. int s_index;
  797. int i;
  798. int last_pos, bits_left;
  799. VLC *vlc;
  800. int end_pos = FFMIN(end_pos2, s->gb.size_in_bits - s->extrasize * 8);
  801. /* low frequencies (called big values) */
  802. s_index = 0;
  803. for (i = 0; i < 3; i++) {
  804. int j, k, l, linbits;
  805. j = g->region_size[i];
  806. if (j == 0)
  807. continue;
  808. /* select vlc table */
  809. k = g->table_select[i];
  810. l = mpa_huff_data[k][0];
  811. linbits = mpa_huff_data[k][1];
  812. vlc = &huff_vlc[l];
  813. if (!l) {
  814. memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * 2 * j);
  815. s_index += 2 * j;
  816. continue;
  817. }
  818. /* read huffcode and compute each couple */
  819. for (; j > 0; j--) {
  820. int exponent, x, y;
  821. int v;
  822. int pos = get_bits_count(&s->gb);
  823. if (pos >= end_pos){
  824. switch_buffer(s, &pos, &end_pos, &end_pos2);
  825. if (pos >= end_pos)
  826. break;
  827. }
  828. y = get_vlc2(&s->gb, vlc->table, 7, 3);
  829. if (!y) {
  830. g->sb_hybrid[s_index ] =
  831. g->sb_hybrid[s_index + 1] = 0;
  832. s_index += 2;
  833. continue;
  834. }
  835. exponent= exponents[s_index];
  836. ff_dlog(s->avctx, "region=%d n=%d y=%d exp=%d\n",
  837. i, g->region_size[i] - j, y, exponent);
  838. if (y & 16) {
  839. x = y >> 5;
  840. y = y & 0x0f;
  841. if (x < 15) {
  842. READ_FLIP_SIGN(g->sb_hybrid + s_index, RENAME(expval_table)[exponent] + x)
  843. } else {
  844. x += get_bitsz(&s->gb, linbits);
  845. v = l3_unscale(x, exponent);
  846. if (get_bits1(&s->gb))
  847. v = -v;
  848. g->sb_hybrid[s_index] = v;
  849. }
  850. if (y < 15) {
  851. READ_FLIP_SIGN(g->sb_hybrid + s_index + 1, RENAME(expval_table)[exponent] + y)
  852. } else {
  853. y += get_bitsz(&s->gb, linbits);
  854. v = l3_unscale(y, exponent);
  855. if (get_bits1(&s->gb))
  856. v = -v;
  857. g->sb_hybrid[s_index + 1] = v;
  858. }
  859. } else {
  860. x = y >> 5;
  861. y = y & 0x0f;
  862. x += y;
  863. if (x < 15) {
  864. READ_FLIP_SIGN(g->sb_hybrid + s_index + !!y, RENAME(expval_table)[exponent] + x)
  865. } else {
  866. x += get_bitsz(&s->gb, linbits);
  867. v = l3_unscale(x, exponent);
  868. if (get_bits1(&s->gb))
  869. v = -v;
  870. g->sb_hybrid[s_index+!!y] = v;
  871. }
  872. g->sb_hybrid[s_index + !y] = 0;
  873. }
  874. s_index += 2;
  875. }
  876. }
  877. /* high frequencies */
  878. vlc = &huff_quad_vlc[g->count1table_select];
  879. last_pos = 0;
  880. while (s_index <= 572) {
  881. int pos, code;
  882. pos = get_bits_count(&s->gb);
  883. if (pos >= end_pos) {
  884. if (pos > end_pos2 && last_pos) {
  885. /* some encoders generate an incorrect size for this
  886. part. We must go back into the data */
  887. s_index -= 4;
  888. skip_bits_long(&s->gb, last_pos - pos);
  889. av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
  890. if(s->err_recognition & (AV_EF_BITSTREAM|AV_EF_COMPLIANT))
  891. s_index=0;
  892. break;
  893. }
  894. switch_buffer(s, &pos, &end_pos, &end_pos2);
  895. if (pos >= end_pos)
  896. break;
  897. }
  898. last_pos = pos;
  899. code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
  900. ff_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
  901. g->sb_hybrid[s_index + 0] =
  902. g->sb_hybrid[s_index + 1] =
  903. g->sb_hybrid[s_index + 2] =
  904. g->sb_hybrid[s_index + 3] = 0;
  905. while (code) {
  906. static const int idxtab[16] = { 3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0 };
  907. int v;
  908. int pos = s_index + idxtab[code];
  909. code ^= 8 >> idxtab[code];
  910. READ_FLIP_SIGN(g->sb_hybrid + pos, RENAME(exp_table)+exponents[pos])
  911. }
  912. s_index += 4;
  913. }
  914. /* skip extension bits */
  915. bits_left = end_pos2 - get_bits_count(&s->gb);
  916. if (bits_left < 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_COMPLIANT))) {
  917. av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
  918. s_index=0;
  919. } else if (bits_left > 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_AGGRESSIVE))) {
  920. av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
  921. s_index = 0;
  922. }
  923. memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * (576 - s_index));
  924. skip_bits_long(&s->gb, bits_left);
  925. i = get_bits_count(&s->gb);
  926. switch_buffer(s, &i, &end_pos, &end_pos2);
  927. return 0;
  928. }
  929. /* Reorder short blocks from bitstream order to interleaved order. It
  930. would be faster to do it in parsing, but the code would be far more
  931. complicated */
  932. static void reorder_block(MPADecodeContext *s, GranuleDef *g)
  933. {
  934. int i, j, len;
  935. INTFLOAT *ptr, *dst, *ptr1;
  936. INTFLOAT tmp[576];
  937. if (g->block_type != 2)
  938. return;
  939. if (g->switch_point) {
  940. if (s->sample_rate_index != 8)
  941. ptr = g->sb_hybrid + 36;
  942. else
  943. ptr = g->sb_hybrid + 72;
  944. } else {
  945. ptr = g->sb_hybrid;
  946. }
  947. for (i = g->short_start; i < 13; i++) {
  948. len = band_size_short[s->sample_rate_index][i];
  949. ptr1 = ptr;
  950. dst = tmp;
  951. for (j = len; j > 0; j--) {
  952. *dst++ = ptr[0*len];
  953. *dst++ = ptr[1*len];
  954. *dst++ = ptr[2*len];
  955. ptr++;
  956. }
  957. ptr += 2 * len;
  958. memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
  959. }
  960. }
  961. #define ISQRT2 FIXR(0.70710678118654752440)
  962. static void compute_stereo(MPADecodeContext *s, GranuleDef *g0, GranuleDef *g1)
  963. {
  964. int i, j, k, l;
  965. int sf_max, sf, len, non_zero_found;
  966. INTFLOAT (*is_tab)[16], *tab0, *tab1, v1, v2;
  967. SUINTFLOAT tmp0, tmp1;
  968. int non_zero_found_short[3];
  969. /* intensity stereo */
  970. if (s->mode_ext & MODE_EXT_I_STEREO) {
  971. if (!s->lsf) {
  972. is_tab = is_table;
  973. sf_max = 7;
  974. } else {
  975. is_tab = is_table_lsf[g1->scalefac_compress & 1];
  976. sf_max = 16;
  977. }
  978. tab0 = g0->sb_hybrid + 576;
  979. tab1 = g1->sb_hybrid + 576;
  980. non_zero_found_short[0] = 0;
  981. non_zero_found_short[1] = 0;
  982. non_zero_found_short[2] = 0;
  983. k = (13 - g1->short_start) * 3 + g1->long_end - 3;
  984. for (i = 12; i >= g1->short_start; i--) {
  985. /* for last band, use previous scale factor */
  986. if (i != 11)
  987. k -= 3;
  988. len = band_size_short[s->sample_rate_index][i];
  989. for (l = 2; l >= 0; l--) {
  990. tab0 -= len;
  991. tab1 -= len;
  992. if (!non_zero_found_short[l]) {
  993. /* test if non zero band. if so, stop doing i-stereo */
  994. for (j = 0; j < len; j++) {
  995. if (tab1[j] != 0) {
  996. non_zero_found_short[l] = 1;
  997. goto found1;
  998. }
  999. }
  1000. sf = g1->scale_factors[k + l];
  1001. if (sf >= sf_max)
  1002. goto found1;
  1003. v1 = is_tab[0][sf];
  1004. v2 = is_tab[1][sf];
  1005. for (j = 0; j < len; j++) {
  1006. tmp0 = tab0[j];
  1007. tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
  1008. tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
  1009. }
  1010. } else {
  1011. found1:
  1012. if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1013. /* lower part of the spectrum : do ms stereo
  1014. if enabled */
  1015. for (j = 0; j < len; j++) {
  1016. tmp0 = tab0[j];
  1017. tmp1 = tab1[j];
  1018. tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
  1019. tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
  1020. }
  1021. }
  1022. }
  1023. }
  1024. }
  1025. non_zero_found = non_zero_found_short[0] |
  1026. non_zero_found_short[1] |
  1027. non_zero_found_short[2];
  1028. for (i = g1->long_end - 1;i >= 0;i--) {
  1029. len = band_size_long[s->sample_rate_index][i];
  1030. tab0 -= len;
  1031. tab1 -= len;
  1032. /* test if non zero band. if so, stop doing i-stereo */
  1033. if (!non_zero_found) {
  1034. for (j = 0; j < len; j++) {
  1035. if (tab1[j] != 0) {
  1036. non_zero_found = 1;
  1037. goto found2;
  1038. }
  1039. }
  1040. /* for last band, use previous scale factor */
  1041. k = (i == 21) ? 20 : i;
  1042. sf = g1->scale_factors[k];
  1043. if (sf >= sf_max)
  1044. goto found2;
  1045. v1 = is_tab[0][sf];
  1046. v2 = is_tab[1][sf];
  1047. for (j = 0; j < len; j++) {
  1048. tmp0 = tab0[j];
  1049. tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
  1050. tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
  1051. }
  1052. } else {
  1053. found2:
  1054. if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1055. /* lower part of the spectrum : do ms stereo
  1056. if enabled */
  1057. for (j = 0; j < len; j++) {
  1058. tmp0 = tab0[j];
  1059. tmp1 = tab1[j];
  1060. tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
  1061. tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
  1062. }
  1063. }
  1064. }
  1065. }
  1066. } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1067. /* ms stereo ONLY */
  1068. /* NOTE: the 1/sqrt(2) normalization factor is included in the
  1069. global gain */
  1070. #if USE_FLOATS
  1071. s->butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);
  1072. #else
  1073. tab0 = g0->sb_hybrid;
  1074. tab1 = g1->sb_hybrid;
  1075. for (i = 0; i < 576; i++) {
  1076. tmp0 = tab0[i];
  1077. tmp1 = tab1[i];
  1078. tab0[i] = tmp0 + tmp1;
  1079. tab1[i] = tmp0 - tmp1;
  1080. }
  1081. #endif
  1082. }
  1083. }
  1084. #if USE_FLOATS
  1085. #if HAVE_MIPSFPU
  1086. # include "mips/compute_antialias_float.h"
  1087. #endif /* HAVE_MIPSFPU */
  1088. #else
  1089. #if HAVE_MIPSDSP
  1090. # include "mips/compute_antialias_fixed.h"
  1091. #endif /* HAVE_MIPSDSP */
  1092. #endif /* USE_FLOATS */
  1093. #ifndef compute_antialias
  1094. #if USE_FLOATS
  1095. #define AA(j) do { \
  1096. float tmp0 = ptr[-1-j]; \
  1097. float tmp1 = ptr[ j]; \
  1098. ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1]; \
  1099. ptr[ j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0]; \
  1100. } while (0)
  1101. #else
  1102. #define AA(j) do { \
  1103. SUINT tmp0 = ptr[-1-j]; \
  1104. SUINT tmp1 = ptr[ j]; \
  1105. SUINT tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]); \
  1106. ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2])); \
  1107. ptr[ j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3])); \
  1108. } while (0)
  1109. #endif
  1110. static void compute_antialias(MPADecodeContext *s, GranuleDef *g)
  1111. {
  1112. INTFLOAT *ptr;
  1113. int n, i;
  1114. /* we antialias only "long" bands */
  1115. if (g->block_type == 2) {
  1116. if (!g->switch_point)
  1117. return;
  1118. /* XXX: check this for 8000Hz case */
  1119. n = 1;
  1120. } else {
  1121. n = SBLIMIT - 1;
  1122. }
  1123. ptr = g->sb_hybrid + 18;
  1124. for (i = n; i > 0; i--) {
  1125. AA(0);
  1126. AA(1);
  1127. AA(2);
  1128. AA(3);
  1129. AA(4);
  1130. AA(5);
  1131. AA(6);
  1132. AA(7);
  1133. ptr += 18;
  1134. }
  1135. }
  1136. #endif /* compute_antialias */
  1137. static void compute_imdct(MPADecodeContext *s, GranuleDef *g,
  1138. INTFLOAT *sb_samples, INTFLOAT *mdct_buf)
  1139. {
  1140. INTFLOAT *win, *out_ptr, *ptr, *buf, *ptr1;
  1141. INTFLOAT out2[12];
  1142. int i, j, mdct_long_end, sblimit;
  1143. /* find last non zero block */
  1144. ptr = g->sb_hybrid + 576;
  1145. ptr1 = g->sb_hybrid + 2 * 18;
  1146. while (ptr >= ptr1) {
  1147. int32_t *p;
  1148. ptr -= 6;
  1149. p = (int32_t*)ptr;
  1150. if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
  1151. break;
  1152. }
  1153. sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
  1154. if (g->block_type == 2) {
  1155. /* XXX: check for 8000 Hz */
  1156. if (g->switch_point)
  1157. mdct_long_end = 2;
  1158. else
  1159. mdct_long_end = 0;
  1160. } else {
  1161. mdct_long_end = sblimit;
  1162. }
  1163. s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid,
  1164. mdct_long_end, g->switch_point,
  1165. g->block_type);
  1166. buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3);
  1167. ptr = g->sb_hybrid + 18 * mdct_long_end;
  1168. for (j = mdct_long_end; j < sblimit; j++) {
  1169. /* select frequency inversion */
  1170. win = RENAME(ff_mdct_win)[2 + (4 & -(j & 1))];
  1171. out_ptr = sb_samples + j;
  1172. for (i = 0; i < 6; i++) {
  1173. *out_ptr = buf[4*i];
  1174. out_ptr += SBLIMIT;
  1175. }
  1176. imdct12(out2, ptr + 0);
  1177. for (i = 0; i < 6; i++) {
  1178. *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*1)];
  1179. buf[4*(i + 6*2)] = MULH3(out2[i + 6], win[i + 6], 1);
  1180. out_ptr += SBLIMIT;
  1181. }
  1182. imdct12(out2, ptr + 1);
  1183. for (i = 0; i < 6; i++) {
  1184. *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*2)];
  1185. buf[4*(i + 6*0)] = MULH3(out2[i + 6], win[i + 6], 1);
  1186. out_ptr += SBLIMIT;
  1187. }
  1188. imdct12(out2, ptr + 2);
  1189. for (i = 0; i < 6; i++) {
  1190. buf[4*(i + 6*0)] = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*0)];
  1191. buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1);
  1192. buf[4*(i + 6*2)] = 0;
  1193. }
  1194. ptr += 18;
  1195. buf += (j&3) != 3 ? 1 : (4*18-3);
  1196. }
  1197. /* zero bands */
  1198. for (j = sblimit; j < SBLIMIT; j++) {
  1199. /* overlap */
  1200. out_ptr = sb_samples + j;
  1201. for (i = 0; i < 18; i++) {
  1202. *out_ptr = buf[4*i];
  1203. buf[4*i] = 0;
  1204. out_ptr += SBLIMIT;
  1205. }
  1206. buf += (j&3) != 3 ? 1 : (4*18-3);
  1207. }
  1208. }
  1209. /* main layer3 decoding function */
  1210. static int mp_decode_layer3(MPADecodeContext *s)
  1211. {
  1212. int nb_granules, main_data_begin;
  1213. int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
  1214. GranuleDef *g;
  1215. int16_t exponents[576]; //FIXME try INTFLOAT
  1216. int ret;
  1217. /* read side info */
  1218. if (s->lsf) {
  1219. ret = handle_crc(s, ((s->nb_channels == 1) ? 8*9 : 8*17));
  1220. main_data_begin = get_bits(&s->gb, 8);
  1221. skip_bits(&s->gb, s->nb_channels);
  1222. nb_granules = 1;
  1223. } else {
  1224. ret = handle_crc(s, ((s->nb_channels == 1) ? 8*17 : 8*32));
  1225. main_data_begin = get_bits(&s->gb, 9);
  1226. if (s->nb_channels == 2)
  1227. skip_bits(&s->gb, 3);
  1228. else
  1229. skip_bits(&s->gb, 5);
  1230. nb_granules = 2;
  1231. for (ch = 0; ch < s->nb_channels; ch++) {
  1232. s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
  1233. s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
  1234. }
  1235. }
  1236. if (ret < 0)
  1237. return ret;
  1238. for (gr = 0; gr < nb_granules; gr++) {
  1239. for (ch = 0; ch < s->nb_channels; ch++) {
  1240. ff_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
  1241. g = &s->granules[ch][gr];
  1242. g->part2_3_length = get_bits(&s->gb, 12);
  1243. g->big_values = get_bits(&s->gb, 9);
  1244. if (g->big_values > 288) {
  1245. av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
  1246. return AVERROR_INVALIDDATA;
  1247. }
  1248. g->global_gain = get_bits(&s->gb, 8);
  1249. /* if MS stereo only is selected, we precompute the
  1250. 1/sqrt(2) renormalization factor */
  1251. if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
  1252. MODE_EXT_MS_STEREO)
  1253. g->global_gain -= 2;
  1254. if (s->lsf)
  1255. g->scalefac_compress = get_bits(&s->gb, 9);
  1256. else
  1257. g->scalefac_compress = get_bits(&s->gb, 4);
  1258. blocksplit_flag = get_bits1(&s->gb);
  1259. if (blocksplit_flag) {
  1260. g->block_type = get_bits(&s->gb, 2);
  1261. if (g->block_type == 0) {
  1262. av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
  1263. return AVERROR_INVALIDDATA;
  1264. }
  1265. g->switch_point = get_bits1(&s->gb);
  1266. for (i = 0; i < 2; i++)
  1267. g->table_select[i] = get_bits(&s->gb, 5);
  1268. for (i = 0; i < 3; i++)
  1269. g->subblock_gain[i] = get_bits(&s->gb, 3);
  1270. init_short_region(s, g);
  1271. } else {
  1272. int region_address1, region_address2;
  1273. g->block_type = 0;
  1274. g->switch_point = 0;
  1275. for (i = 0; i < 3; i++)
  1276. g->table_select[i] = get_bits(&s->gb, 5);
  1277. /* compute huffman coded region sizes */
  1278. region_address1 = get_bits(&s->gb, 4);
  1279. region_address2 = get_bits(&s->gb, 3);
  1280. ff_dlog(s->avctx, "region1=%d region2=%d\n",
  1281. region_address1, region_address2);
  1282. init_long_region(s, g, region_address1, region_address2);
  1283. }
  1284. region_offset2size(g);
  1285. compute_band_indexes(s, g);
  1286. g->preflag = 0;
  1287. if (!s->lsf)
  1288. g->preflag = get_bits1(&s->gb);
  1289. g->scalefac_scale = get_bits1(&s->gb);
  1290. g->count1table_select = get_bits1(&s->gb);
  1291. ff_dlog(s->avctx, "block_type=%d switch_point=%d\n",
  1292. g->block_type, g->switch_point);
  1293. }
  1294. }
  1295. if (!s->adu_mode) {
  1296. int skip;
  1297. const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb) >> 3);
  1298. s->extrasize = av_clip((get_bits_left(&s->gb) >> 3) - s->extrasize, 0,
  1299. FFMAX(0, LAST_BUF_SIZE - s->last_buf_size));
  1300. av_assert1((get_bits_count(&s->gb) & 7) == 0);
  1301. /* now we get bits from the main_data_begin offset */
  1302. ff_dlog(s->avctx, "seekback:%d, lastbuf:%d\n",
  1303. main_data_begin, s->last_buf_size);
  1304. memcpy(s->last_buf + s->last_buf_size, ptr, s->extrasize);
  1305. s->in_gb = s->gb;
  1306. init_get_bits(&s->gb, s->last_buf, (s->last_buf_size + s->extrasize) * 8);
  1307. s->last_buf_size <<= 3;
  1308. for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) {
  1309. for (ch = 0; ch < s->nb_channels; ch++) {
  1310. g = &s->granules[ch][gr];
  1311. s->last_buf_size += g->part2_3_length;
  1312. memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
  1313. compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
  1314. }
  1315. }
  1316. skip = s->last_buf_size - 8 * main_data_begin;
  1317. if (skip >= s->gb.size_in_bits - s->extrasize * 8 && s->in_gb.buffer) {
  1318. skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits + s->extrasize * 8);
  1319. s->gb = s->in_gb;
  1320. s->in_gb.buffer = NULL;
  1321. s->extrasize = 0;
  1322. } else {
  1323. skip_bits_long(&s->gb, skip);
  1324. }
  1325. } else {
  1326. gr = 0;
  1327. s->extrasize = 0;
  1328. }
  1329. for (; gr < nb_granules; gr++) {
  1330. for (ch = 0; ch < s->nb_channels; ch++) {
  1331. g = &s->granules[ch][gr];
  1332. bits_pos = get_bits_count(&s->gb);
  1333. if (!s->lsf) {
  1334. uint8_t *sc;
  1335. int slen, slen1, slen2;
  1336. /* MPEG-1 scale factors */
  1337. slen1 = slen_table[0][g->scalefac_compress];
  1338. slen2 = slen_table[1][g->scalefac_compress];
  1339. ff_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
  1340. if (g->block_type == 2) {
  1341. n = g->switch_point ? 17 : 18;
  1342. j = 0;
  1343. if (slen1) {
  1344. for (i = 0; i < n; i++)
  1345. g->scale_factors[j++] = get_bits(&s->gb, slen1);
  1346. } else {
  1347. for (i = 0; i < n; i++)
  1348. g->scale_factors[j++] = 0;
  1349. }
  1350. if (slen2) {
  1351. for (i = 0; i < 18; i++)
  1352. g->scale_factors[j++] = get_bits(&s->gb, slen2);
  1353. for (i = 0; i < 3; i++)
  1354. g->scale_factors[j++] = 0;
  1355. } else {
  1356. for (i = 0; i < 21; i++)
  1357. g->scale_factors[j++] = 0;
  1358. }
  1359. } else {
  1360. sc = s->granules[ch][0].scale_factors;
  1361. j = 0;
  1362. for (k = 0; k < 4; k++) {
  1363. n = k == 0 ? 6 : 5;
  1364. if ((g->scfsi & (0x8 >> k)) == 0) {
  1365. slen = (k < 2) ? slen1 : slen2;
  1366. if (slen) {
  1367. for (i = 0; i < n; i++)
  1368. g->scale_factors[j++] = get_bits(&s->gb, slen);
  1369. } else {
  1370. for (i = 0; i < n; i++)
  1371. g->scale_factors[j++] = 0;
  1372. }
  1373. } else {
  1374. /* simply copy from last granule */
  1375. for (i = 0; i < n; i++) {
  1376. g->scale_factors[j] = sc[j];
  1377. j++;
  1378. }
  1379. }
  1380. }
  1381. g->scale_factors[j++] = 0;
  1382. }
  1383. } else {
  1384. int tindex, tindex2, slen[4], sl, sf;
  1385. /* LSF scale factors */
  1386. if (g->block_type == 2)
  1387. tindex = g->switch_point ? 2 : 1;
  1388. else
  1389. tindex = 0;
  1390. sf = g->scalefac_compress;
  1391. if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
  1392. /* intensity stereo case */
  1393. sf >>= 1;
  1394. if (sf < 180) {
  1395. lsf_sf_expand(slen, sf, 6, 6, 0);
  1396. tindex2 = 3;
  1397. } else if (sf < 244) {
  1398. lsf_sf_expand(slen, sf - 180, 4, 4, 0);
  1399. tindex2 = 4;
  1400. } else {
  1401. lsf_sf_expand(slen, sf - 244, 3, 0, 0);
  1402. tindex2 = 5;
  1403. }
  1404. } else {
  1405. /* normal case */
  1406. if (sf < 400) {
  1407. lsf_sf_expand(slen, sf, 5, 4, 4);
  1408. tindex2 = 0;
  1409. } else if (sf < 500) {
  1410. lsf_sf_expand(slen, sf - 400, 5, 4, 0);
  1411. tindex2 = 1;
  1412. } else {
  1413. lsf_sf_expand(slen, sf - 500, 3, 0, 0);
  1414. tindex2 = 2;
  1415. g->preflag = 1;
  1416. }
  1417. }
  1418. j = 0;
  1419. for (k = 0; k < 4; k++) {
  1420. n = lsf_nsf_table[tindex2][tindex][k];
  1421. sl = slen[k];
  1422. if (sl) {
  1423. for (i = 0; i < n; i++)
  1424. g->scale_factors[j++] = get_bits(&s->gb, sl);
  1425. } else {
  1426. for (i = 0; i < n; i++)
  1427. g->scale_factors[j++] = 0;
  1428. }
  1429. }
  1430. /* XXX: should compute exact size */
  1431. for (; j < 40; j++)
  1432. g->scale_factors[j] = 0;
  1433. }
  1434. exponents_from_scale_factors(s, g, exponents);
  1435. /* read Huffman coded residue */
  1436. huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
  1437. } /* ch */
  1438. if (s->mode == MPA_JSTEREO)
  1439. compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
  1440. for (ch = 0; ch < s->nb_channels; ch++) {
  1441. g = &s->granules[ch][gr];
  1442. reorder_block(s, g);
  1443. compute_antialias(s, g);
  1444. compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
  1445. }
  1446. } /* gr */
  1447. if (get_bits_count(&s->gb) < 0)
  1448. skip_bits_long(&s->gb, -get_bits_count(&s->gb));
  1449. return nb_granules * 18;
  1450. }
  1451. static int mp_decode_frame(MPADecodeContext *s, OUT_INT **samples,
  1452. const uint8_t *buf, int buf_size)
  1453. {
  1454. int i, nb_frames, ch, ret;
  1455. OUT_INT *samples_ptr;
  1456. init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);
  1457. if (s->error_protection)
  1458. s->crc = get_bits(&s->gb, 16);
  1459. switch(s->layer) {
  1460. case 1:
  1461. s->avctx->frame_size = 384;
  1462. nb_frames = mp_decode_layer1(s);
  1463. break;
  1464. case 2:
  1465. s->avctx->frame_size = 1152;
  1466. nb_frames = mp_decode_layer2(s);
  1467. break;
  1468. case 3:
  1469. s->avctx->frame_size = s->lsf ? 576 : 1152;
  1470. default:
  1471. nb_frames = mp_decode_layer3(s);
  1472. s->last_buf_size=0;
  1473. if (s->in_gb.buffer) {
  1474. align_get_bits(&s->gb);
  1475. i = (get_bits_left(&s->gb) >> 3) - s->extrasize;
  1476. if (i >= 0 && i <= BACKSTEP_SIZE) {
  1477. memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb) >> 3), i);
  1478. s->last_buf_size=i;
  1479. } else
  1480. av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
  1481. s->gb = s->in_gb;
  1482. s->in_gb.buffer = NULL;
  1483. s->extrasize = 0;
  1484. }
  1485. align_get_bits(&s->gb);
  1486. av_assert1((get_bits_count(&s->gb) & 7) == 0);
  1487. i = (get_bits_left(&s->gb) >> 3) - s->extrasize;
  1488. if (i < 0 || i > BACKSTEP_SIZE || nb_frames < 0) {
  1489. if (i < 0)
  1490. av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
  1491. i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
  1492. }
  1493. av_assert1(i <= buf_size - HEADER_SIZE && i >= 0);
  1494. memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
  1495. s->last_buf_size += i;
  1496. }
  1497. if(nb_frames < 0)
  1498. return nb_frames;
  1499. /* get output buffer */
  1500. if (!samples) {
  1501. av_assert0(s->frame);
  1502. s->frame->nb_samples = s->avctx->frame_size;
  1503. if ((ret = ff_get_buffer(s->avctx, s->frame, 0)) < 0)
  1504. return ret;
  1505. samples = (OUT_INT **)s->frame->extended_data;
  1506. }
  1507. /* apply the synthesis filter */
  1508. for (ch = 0; ch < s->nb_channels; ch++) {
  1509. int sample_stride;
  1510. if (s->avctx->sample_fmt == OUT_FMT_P) {
  1511. samples_ptr = samples[ch];
  1512. sample_stride = 1;
  1513. } else {
  1514. samples_ptr = samples[0] + ch;
  1515. sample_stride = s->nb_channels;
  1516. }
  1517. for (i = 0; i < nb_frames; i++) {
  1518. RENAME(ff_mpa_synth_filter)(&s->mpadsp, s->synth_buf[ch],
  1519. &(s->synth_buf_offset[ch]),
  1520. RENAME(ff_mpa_synth_window),
  1521. &s->dither_state, samples_ptr,
  1522. sample_stride, s->sb_samples[ch][i]);
  1523. samples_ptr += 32 * sample_stride;
  1524. }
  1525. }
  1526. return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
  1527. }
  1528. static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,
  1529. AVPacket *avpkt)
  1530. {
  1531. const uint8_t *buf = avpkt->data;
  1532. int buf_size = avpkt->size;
  1533. MPADecodeContext *s = avctx->priv_data;
  1534. uint32_t header;
  1535. int ret;
  1536. int skipped = 0;
  1537. while(buf_size && !*buf){
  1538. buf++;
  1539. buf_size--;
  1540. skipped++;
  1541. }
  1542. if (buf_size < HEADER_SIZE)
  1543. return AVERROR_INVALIDDATA;
  1544. header = AV_RB32(buf);
  1545. if (header >> 8 == AV_RB32("TAG") >> 8) {
  1546. av_log(avctx, AV_LOG_DEBUG, "discarding ID3 tag\n");
  1547. return buf_size + skipped;
  1548. }
  1549. ret = avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);
  1550. if (ret < 0) {
  1551. av_log(avctx, AV_LOG_ERROR, "Header missing\n");
  1552. return AVERROR_INVALIDDATA;
  1553. } else if (ret == 1) {
  1554. /* free format: prepare to compute frame size */
  1555. s->frame_size = -1;
  1556. return AVERROR_INVALIDDATA;
  1557. }
  1558. /* update codec info */
  1559. avctx->channels = s->nb_channels;
  1560. avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
  1561. if (!avctx->bit_rate)
  1562. avctx->bit_rate = s->bit_rate;
  1563. if (s->frame_size <= 0) {
  1564. av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
  1565. return AVERROR_INVALIDDATA;
  1566. } else if (s->frame_size < buf_size) {
  1567. av_log(avctx, AV_LOG_DEBUG, "incorrect frame size - multiple frames in buffer?\n");
  1568. buf_size= s->frame_size;
  1569. }
  1570. s->frame = data;
  1571. ret = mp_decode_frame(s, NULL, buf, buf_size);
  1572. if (ret >= 0) {
  1573. s->frame->nb_samples = avctx->frame_size;
  1574. *got_frame_ptr = 1;
  1575. avctx->sample_rate = s->sample_rate;
  1576. //FIXME maybe move the other codec info stuff from above here too
  1577. } else {
  1578. av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
  1579. /* Only return an error if the bad frame makes up the whole packet or
  1580. * the error is related to buffer management.
  1581. * If there is more data in the packet, just consume the bad frame
  1582. * instead of returning an error, which would discard the whole
  1583. * packet. */
  1584. *got_frame_ptr = 0;
  1585. if (buf_size == avpkt->size || ret != AVERROR_INVALIDDATA)
  1586. return ret;
  1587. }
  1588. s->frame_size = 0;
  1589. return buf_size + skipped;
  1590. }
  1591. static void mp_flush(MPADecodeContext *ctx)
  1592. {
  1593. memset(ctx->synth_buf, 0, sizeof(ctx->synth_buf));
  1594. memset(ctx->mdct_buf, 0, sizeof(ctx->mdct_buf));
  1595. ctx->last_buf_size = 0;
  1596. ctx->dither_state = 0;
  1597. }
  1598. static void flush(AVCodecContext *avctx)
  1599. {
  1600. mp_flush(avctx->priv_data);
  1601. }
  1602. #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
  1603. static int decode_frame_adu(AVCodecContext *avctx, void *data,
  1604. int *got_frame_ptr, AVPacket *avpkt)
  1605. {
  1606. const uint8_t *buf = avpkt->data;
  1607. int buf_size = avpkt->size;
  1608. MPADecodeContext *s = avctx->priv_data;
  1609. uint32_t header;
  1610. int len, ret;
  1611. int av_unused out_size;
  1612. len = buf_size;
  1613. // Discard too short frames
  1614. if (buf_size < HEADER_SIZE) {
  1615. av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
  1616. return AVERROR_INVALIDDATA;
  1617. }
  1618. if (len > MPA_MAX_CODED_FRAME_SIZE)
  1619. len = MPA_MAX_CODED_FRAME_SIZE;
  1620. // Get header and restore sync word
  1621. header = AV_RB32(buf) | 0xffe00000;
  1622. ret = avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);
  1623. if (ret < 0) {
  1624. av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n");
  1625. return ret;
  1626. }
  1627. /* update codec info */
  1628. avctx->sample_rate = s->sample_rate;
  1629. avctx->channels = s->nb_channels;
  1630. avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
  1631. if (!avctx->bit_rate)
  1632. avctx->bit_rate = s->bit_rate;
  1633. s->frame_size = len;
  1634. s->frame = data;
  1635. ret = mp_decode_frame(s, NULL, buf, buf_size);
  1636. if (ret < 0) {
  1637. av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
  1638. return ret;
  1639. }
  1640. *got_frame_ptr = 1;
  1641. return buf_size;
  1642. }
  1643. #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
  1644. #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
  1645. /**
  1646. * Context for MP3On4 decoder
  1647. */
  1648. typedef struct MP3On4DecodeContext {
  1649. int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
  1650. int syncword; ///< syncword patch
  1651. const uint8_t *coff; ///< channel offsets in output buffer
  1652. MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
  1653. } MP3On4DecodeContext;
  1654. #include "mpeg4audio.h"
  1655. /* Next 3 arrays are indexed by channel config number (passed via codecdata) */
  1656. /* number of mp3 decoder instances */
  1657. static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 };
  1658. /* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */
  1659. static const uint8_t chan_offset[8][5] = {
  1660. { 0 },
  1661. { 0 }, // C
  1662. { 0 }, // FLR
  1663. { 2, 0 }, // C FLR
  1664. { 2, 0, 3 }, // C FLR BS
  1665. { 2, 0, 3 }, // C FLR BLRS
  1666. { 2, 0, 4, 3 }, // C FLR BLRS LFE
  1667. { 2, 0, 6, 4, 3 }, // C FLR BLRS BLR LFE
  1668. };
  1669. /* mp3on4 channel layouts */
  1670. static const int16_t chan_layout[8] = {
  1671. 0,
  1672. AV_CH_LAYOUT_MONO,
  1673. AV_CH_LAYOUT_STEREO,
  1674. AV_CH_LAYOUT_SURROUND,
  1675. AV_CH_LAYOUT_4POINT0,
  1676. AV_CH_LAYOUT_5POINT0,
  1677. AV_CH_LAYOUT_5POINT1,
  1678. AV_CH_LAYOUT_7POINT1
  1679. };
  1680. static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
  1681. {
  1682. MP3On4DecodeContext *s = avctx->priv_data;
  1683. int i;
  1684. for (i = 0; i < s->frames; i++)
  1685. av_freep(&s->mp3decctx[i]);
  1686. return 0;
  1687. }
  1688. static av_cold int decode_init_mp3on4(AVCodecContext * avctx)
  1689. {
  1690. MP3On4DecodeContext *s = avctx->priv_data;
  1691. MPEG4AudioConfig cfg;
  1692. int i, ret;
  1693. if ((avctx->extradata_size < 2) || !avctx->extradata) {
  1694. av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
  1695. return AVERROR_INVALIDDATA;
  1696. }
  1697. avpriv_mpeg4audio_get_config2(&cfg, avctx->extradata,
  1698. avctx->extradata_size, 1, avctx);
  1699. if (!cfg.chan_config || cfg.chan_config > 7) {
  1700. av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
  1701. return AVERROR_INVALIDDATA;
  1702. }
  1703. s->frames = mp3Frames[cfg.chan_config];
  1704. s->coff = chan_offset[cfg.chan_config];
  1705. avctx->channels = ff_mpeg4audio_channels[cfg.chan_config];
  1706. avctx->channel_layout = chan_layout[cfg.chan_config];
  1707. if (cfg.sample_rate < 16000)
  1708. s->syncword = 0xffe00000;
  1709. else
  1710. s->syncword = 0xfff00000;
  1711. /* Init the first mp3 decoder in standard way, so that all tables get builded
  1712. * We replace avctx->priv_data with the context of the first decoder so that
  1713. * decode_init() does not have to be changed.
  1714. * Other decoders will be initialized here copying data from the first context
  1715. */
  1716. // Allocate zeroed memory for the first decoder context
  1717. s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
  1718. if (!s->mp3decctx[0])
  1719. return AVERROR(ENOMEM);
  1720. // Put decoder context in place to make init_decode() happy
  1721. avctx->priv_data = s->mp3decctx[0];
  1722. ret = decode_init(avctx);
  1723. // Restore mp3on4 context pointer
  1724. avctx->priv_data = s;
  1725. if (ret < 0)
  1726. return ret;
  1727. s->mp3decctx[0]->adu_mode = 1; // Set adu mode
  1728. /* Create a separate codec/context for each frame (first is already ok).
  1729. * Each frame is 1 or 2 channels - up to 5 frames allowed
  1730. */
  1731. for (i = 1; i < s->frames; i++) {
  1732. s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
  1733. if (!s->mp3decctx[i])
  1734. return AVERROR(ENOMEM);
  1735. s->mp3decctx[i]->adu_mode = 1;
  1736. s->mp3decctx[i]->avctx = avctx;
  1737. s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;
  1738. s->mp3decctx[i]->butterflies_float = s->mp3decctx[0]->butterflies_float;
  1739. }
  1740. return 0;
  1741. }
  1742. static void flush_mp3on4(AVCodecContext *avctx)
  1743. {
  1744. int i;
  1745. MP3On4DecodeContext *s = avctx->priv_data;
  1746. for (i = 0; i < s->frames; i++)
  1747. mp_flush(s->mp3decctx[i]);
  1748. }
  1749. static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
  1750. int *got_frame_ptr, AVPacket *avpkt)
  1751. {
  1752. AVFrame *frame = data;
  1753. const uint8_t *buf = avpkt->data;
  1754. int buf_size = avpkt->size;
  1755. MP3On4DecodeContext *s = avctx->priv_data;
  1756. MPADecodeContext *m;
  1757. int fsize, len = buf_size, out_size = 0;
  1758. uint32_t header;
  1759. OUT_INT **out_samples;
  1760. OUT_INT *outptr[2];
  1761. int fr, ch, ret;
  1762. /* get output buffer */
  1763. frame->nb_samples = MPA_FRAME_SIZE;
  1764. if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
  1765. return ret;
  1766. out_samples = (OUT_INT **)frame->extended_data;
  1767. // Discard too short frames
  1768. if (buf_size < HEADER_SIZE)
  1769. return AVERROR_INVALIDDATA;
  1770. avctx->bit_rate = 0;
  1771. ch = 0;
  1772. for (fr = 0; fr < s->frames; fr++) {
  1773. fsize = AV_RB16(buf) >> 4;
  1774. fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
  1775. m = s->mp3decctx[fr];
  1776. av_assert1(m);
  1777. if (fsize < HEADER_SIZE) {
  1778. av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n");
  1779. return AVERROR_INVALIDDATA;
  1780. }
  1781. header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
  1782. ret = avpriv_mpegaudio_decode_header((MPADecodeHeader *)m, header);
  1783. if (ret < 0) {
  1784. av_log(avctx, AV_LOG_ERROR, "Bad header, discard block\n");
  1785. return AVERROR_INVALIDDATA;
  1786. }
  1787. if (ch + m->nb_channels > avctx->channels ||
  1788. s->coff[fr] + m->nb_channels > avctx->channels) {
  1789. av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "
  1790. "channel count\n");
  1791. return AVERROR_INVALIDDATA;
  1792. }
  1793. ch += m->nb_channels;
  1794. outptr[0] = out_samples[s->coff[fr]];
  1795. if (m->nb_channels > 1)
  1796. outptr[1] = out_samples[s->coff[fr] + 1];
  1797. if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0) {
  1798. av_log(avctx, AV_LOG_ERROR, "failed to decode channel %d\n", ch);
  1799. memset(outptr[0], 0, MPA_FRAME_SIZE*sizeof(OUT_INT));
  1800. if (m->nb_channels > 1)
  1801. memset(outptr[1], 0, MPA_FRAME_SIZE*sizeof(OUT_INT));
  1802. ret = m->nb_channels * MPA_FRAME_SIZE*sizeof(OUT_INT);
  1803. }
  1804. out_size += ret;
  1805. buf += fsize;
  1806. len -= fsize;
  1807. avctx->bit_rate += m->bit_rate;
  1808. }
  1809. if (ch != avctx->channels) {
  1810. av_log(avctx, AV_LOG_ERROR, "failed to decode all channels\n");
  1811. return AVERROR_INVALIDDATA;
  1812. }
  1813. /* update codec info */
  1814. avctx->sample_rate = s->mp3decctx[0]->sample_rate;
  1815. frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));
  1816. *got_frame_ptr = 1;
  1817. return buf_size;
  1818. }
  1819. #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */