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

1931 lines
63KB

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