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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. AVFloatDSPContext *fdsp;
  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. RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));
  261. /* huffman decode tables */
  262. offset = 0;
  263. for (i = 1; i < 16; i++) {
  264. const HuffTable *h = &mpa_huff_tables[i];
  265. int xsize, x, y;
  266. uint8_t tmp_bits [512] = { 0 };
  267. uint16_t tmp_codes[512] = { 0 };
  268. xsize = h->xsize;
  269. j = 0;
  270. for (x = 0; x < xsize; x++) {
  271. for (y = 0; y < xsize; y++) {
  272. tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ];
  273. tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];
  274. }
  275. }
  276. /* XXX: fail test */
  277. huff_vlc[i].table = huff_vlc_tables+offset;
  278. huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
  279. init_vlc(&huff_vlc[i], 7, 512,
  280. tmp_bits, 1, 1, tmp_codes, 2, 2,
  281. INIT_VLC_USE_NEW_STATIC);
  282. offset += huff_vlc_tables_sizes[i];
  283. }
  284. av_assert0(offset == FF_ARRAY_ELEMS(huff_vlc_tables));
  285. offset = 0;
  286. for (i = 0; i < 2; i++) {
  287. huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
  288. huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
  289. init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
  290. mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
  291. INIT_VLC_USE_NEW_STATIC);
  292. offset += huff_quad_vlc_tables_sizes[i];
  293. }
  294. av_assert0(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));
  295. for (i = 0; i < 9; i++) {
  296. k = 0;
  297. for (j = 0; j < 22; j++) {
  298. band_index_long[i][j] = k;
  299. k += band_size_long[i][j];
  300. }
  301. band_index_long[i][22] = k;
  302. }
  303. /* compute n ^ (4/3) and store it in mantissa/exp format */
  304. mpegaudio_tableinit();
  305. for (i = 0; i < 4; i++) {
  306. if (ff_mpa_quant_bits[i] < 0) {
  307. for (j = 0; j < (1 << (-ff_mpa_quant_bits[i]+1)); j++) {
  308. int val1, val2, val3, steps;
  309. int val = j;
  310. steps = ff_mpa_quant_steps[i];
  311. val1 = val % steps;
  312. val /= steps;
  313. val2 = val % steps;
  314. val3 = val / steps;
  315. division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);
  316. }
  317. }
  318. }
  319. for (i = 0; i < 7; i++) {
  320. float f;
  321. INTFLOAT v;
  322. if (i != 6) {
  323. f = tan((double)i * M_PI / 12.0);
  324. v = FIXR(f / (1.0 + f));
  325. } else {
  326. v = FIXR(1.0);
  327. }
  328. is_table[0][ i] = v;
  329. is_table[1][6 - i] = v;
  330. }
  331. /* invalid values */
  332. for (i = 7; i < 16; i++)
  333. is_table[0][i] = is_table[1][i] = 0.0;
  334. for (i = 0; i < 16; i++) {
  335. double f;
  336. int e, k;
  337. for (j = 0; j < 2; j++) {
  338. e = -(j + 1) * ((i + 1) >> 1);
  339. f = exp2(e / 4.0);
  340. k = i & 1;
  341. is_table_lsf[j][k ^ 1][i] = FIXR(f);
  342. is_table_lsf[j][k ][i] = FIXR(1.0);
  343. ff_dlog(NULL, "is_table_lsf %d %d: %f %f\n",
  344. i, j, (float) is_table_lsf[j][0][i],
  345. (float) is_table_lsf[j][1][i]);
  346. }
  347. }
  348. for (i = 0; i < 8; i++) {
  349. double ci, cs, ca;
  350. ci = ci_table[i];
  351. cs = 1.0 / sqrt(1.0 + ci * ci);
  352. ca = cs * ci;
  353. #if !USE_FLOATS
  354. csa_table[i][0] = FIXHR(cs/4);
  355. csa_table[i][1] = FIXHR(ca/4);
  356. csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);
  357. csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);
  358. #else
  359. csa_table[i][0] = cs;
  360. csa_table[i][1] = ca;
  361. csa_table[i][2] = ca + cs;
  362. csa_table[i][3] = ca - cs;
  363. #endif
  364. }
  365. }
  366. #if USE_FLOATS
  367. static av_cold int decode_close(AVCodecContext * avctx)
  368. {
  369. MPADecodeContext *s = avctx->priv_data;
  370. av_freep(&s->fdsp);
  371. return 0;
  372. }
  373. #endif
  374. static av_cold int decode_init(AVCodecContext * avctx)
  375. {
  376. static int initialized_tables = 0;
  377. MPADecodeContext *s = avctx->priv_data;
  378. if (!initialized_tables) {
  379. decode_init_static();
  380. initialized_tables = 1;
  381. }
  382. s->avctx = avctx;
  383. #if USE_FLOATS
  384. s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
  385. if (!s->fdsp)
  386. return AVERROR(ENOMEM);
  387. #endif
  388. ff_mpadsp_init(&s->mpadsp);
  389. if (avctx->request_sample_fmt == OUT_FMT &&
  390. avctx->codec_id != AV_CODEC_ID_MP3ON4)
  391. avctx->sample_fmt = OUT_FMT;
  392. else
  393. avctx->sample_fmt = OUT_FMT_P;
  394. s->err_recognition = avctx->err_recognition;
  395. if (avctx->codec_id == AV_CODEC_ID_MP3ADU)
  396. s->adu_mode = 1;
  397. return 0;
  398. }
  399. #define C3 FIXHR(0.86602540378443864676/2)
  400. #define C4 FIXHR(0.70710678118654752439/2) //0.5 / cos(pi*(9)/36)
  401. #define C5 FIXHR(0.51763809020504152469/2) //0.5 / cos(pi*(5)/36)
  402. #define C6 FIXHR(1.93185165257813657349/4) //0.5 / cos(pi*(15)/36)
  403. /* 12 points IMDCT. We compute it "by hand" by factorizing obvious
  404. cases. */
  405. static void imdct12(INTFLOAT *out, SUINTFLOAT *in)
  406. {
  407. SUINTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
  408. in0 = in[0*3];
  409. in1 = in[1*3] + in[0*3];
  410. in2 = in[2*3] + in[1*3];
  411. in3 = in[3*3] + in[2*3];
  412. in4 = in[4*3] + in[3*3];
  413. in5 = in[5*3] + in[4*3];
  414. in5 += in3;
  415. in3 += in1;
  416. in2 = MULH3(in2, C3, 2);
  417. in3 = MULH3(in3, C3, 4);
  418. t1 = in0 - in4;
  419. t2 = MULH3(in1 - in5, C4, 2);
  420. out[ 7] =
  421. out[10] = t1 + t2;
  422. out[ 1] =
  423. out[ 4] = t1 - t2;
  424. in0 += SHR(in4, 1);
  425. in4 = in0 + in2;
  426. in5 += 2*in1;
  427. in1 = MULH3(in5 + in3, C5, 1);
  428. out[ 8] =
  429. out[ 9] = in4 + in1;
  430. out[ 2] =
  431. out[ 3] = in4 - in1;
  432. in0 -= in2;
  433. in5 = MULH3(in5 - in3, C6, 2);
  434. out[ 0] =
  435. out[ 5] = in0 - in5;
  436. out[ 6] =
  437. out[11] = in0 + in5;
  438. }
  439. static int handle_crc(MPADecodeContext *s, int sec_len)
  440. {
  441. if (s->error_protection && (s->err_recognition & AV_EF_CRCCHECK)) {
  442. const uint8_t *buf = s->gb.buffer - HEADER_SIZE;
  443. int sec_byte_len = sec_len >> 3;
  444. int sec_rem_bits = sec_len & 7;
  445. const AVCRC *crc_tab = av_crc_get_table(AV_CRC_16_ANSI);
  446. uint8_t tmp_buf[4];
  447. uint32_t crc_val = av_crc(crc_tab, UINT16_MAX, &buf[2], 2);
  448. crc_val = av_crc(crc_tab, crc_val, &buf[6], sec_byte_len);
  449. AV_WB32(tmp_buf,
  450. ((buf[6 + sec_byte_len] & (0xFF00>>sec_rem_bits))<<24) +
  451. ((s->crc<<16) >> sec_rem_bits));
  452. crc_val = av_crc(crc_tab, crc_val, tmp_buf, 3);
  453. if (crc_val) {
  454. av_log(s->avctx, AV_LOG_ERROR, "CRC mismatch %X!\n", crc_val);
  455. if (s->err_recognition & AV_EF_EXPLODE)
  456. return AVERROR_INVALIDDATA;
  457. }
  458. }
  459. return 0;
  460. }
  461. /* return the number of decoded frames */
  462. static int mp_decode_layer1(MPADecodeContext *s)
  463. {
  464. int bound, i, v, n, ch, j, mant;
  465. uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
  466. uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
  467. int ret;
  468. ret = handle_crc(s, (s->nb_channels == 1) ? 8*16 : 8*32);
  469. if (ret < 0)
  470. return ret;
  471. if (s->mode == MPA_JSTEREO)
  472. bound = (s->mode_ext + 1) * 4;
  473. else
  474. bound = SBLIMIT;
  475. /* allocation bits */
  476. for (i = 0; i < bound; i++) {
  477. for (ch = 0; ch < s->nb_channels; ch++) {
  478. allocation[ch][i] = get_bits(&s->gb, 4);
  479. }
  480. }
  481. for (i = bound; i < SBLIMIT; i++)
  482. allocation[0][i] = get_bits(&s->gb, 4);
  483. /* scale factors */
  484. for (i = 0; i < bound; i++) {
  485. for (ch = 0; ch < s->nb_channels; ch++) {
  486. if (allocation[ch][i])
  487. scale_factors[ch][i] = get_bits(&s->gb, 6);
  488. }
  489. }
  490. for (i = bound; i < SBLIMIT; i++) {
  491. if (allocation[0][i]) {
  492. scale_factors[0][i] = get_bits(&s->gb, 6);
  493. scale_factors[1][i] = get_bits(&s->gb, 6);
  494. }
  495. }
  496. /* compute samples */
  497. for (j = 0; j < 12; j++) {
  498. for (i = 0; i < bound; i++) {
  499. for (ch = 0; ch < s->nb_channels; ch++) {
  500. n = allocation[ch][i];
  501. if (n) {
  502. mant = get_bits(&s->gb, n + 1);
  503. v = l1_unscale(n, mant, scale_factors[ch][i]);
  504. } else {
  505. v = 0;
  506. }
  507. s->sb_samples[ch][j][i] = v;
  508. }
  509. }
  510. for (i = bound; i < SBLIMIT; i++) {
  511. n = allocation[0][i];
  512. if (n) {
  513. mant = get_bits(&s->gb, n + 1);
  514. v = l1_unscale(n, mant, scale_factors[0][i]);
  515. s->sb_samples[0][j][i] = v;
  516. v = l1_unscale(n, mant, scale_factors[1][i]);
  517. s->sb_samples[1][j][i] = v;
  518. } else {
  519. s->sb_samples[0][j][i] = 0;
  520. s->sb_samples[1][j][i] = 0;
  521. }
  522. }
  523. }
  524. return 12;
  525. }
  526. static int mp_decode_layer2(MPADecodeContext *s)
  527. {
  528. int sblimit; /* number of used subbands */
  529. const unsigned char *alloc_table;
  530. int table, bit_alloc_bits, i, j, ch, bound, v;
  531. unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
  532. unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
  533. unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
  534. int scale, qindex, bits, steps, k, l, m, b;
  535. int ret;
  536. /* select decoding table */
  537. table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
  538. s->sample_rate, s->lsf);
  539. sblimit = ff_mpa_sblimit_table[table];
  540. alloc_table = ff_mpa_alloc_tables[table];
  541. if (s->mode == MPA_JSTEREO)
  542. bound = (s->mode_ext + 1) * 4;
  543. else
  544. bound = sblimit;
  545. ff_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
  546. /* sanity check */
  547. if (bound > sblimit)
  548. bound = sblimit;
  549. /* parse bit allocation */
  550. j = 0;
  551. for (i = 0; i < bound; i++) {
  552. bit_alloc_bits = alloc_table[j];
  553. for (ch = 0; ch < s->nb_channels; ch++)
  554. bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
  555. j += 1 << bit_alloc_bits;
  556. }
  557. for (i = bound; i < sblimit; i++) {
  558. bit_alloc_bits = alloc_table[j];
  559. v = get_bits(&s->gb, bit_alloc_bits);
  560. bit_alloc[0][i] = v;
  561. bit_alloc[1][i] = v;
  562. j += 1 << bit_alloc_bits;
  563. }
  564. /* scale codes */
  565. for (i = 0; i < sblimit; i++) {
  566. for (ch = 0; ch < s->nb_channels; ch++) {
  567. if (bit_alloc[ch][i])
  568. scale_code[ch][i] = get_bits(&s->gb, 2);
  569. }
  570. }
  571. ret = handle_crc(s, get_bits_count(&s->gb) - 16);
  572. if (ret < 0)
  573. return ret;
  574. /* scale factors */
  575. for (i = 0; i < sblimit; i++) {
  576. for (ch = 0; ch < s->nb_channels; ch++) {
  577. if (bit_alloc[ch][i]) {
  578. sf = scale_factors[ch][i];
  579. switch (scale_code[ch][i]) {
  580. default:
  581. case 0:
  582. sf[0] = get_bits(&s->gb, 6);
  583. sf[1] = get_bits(&s->gb, 6);
  584. sf[2] = get_bits(&s->gb, 6);
  585. break;
  586. case 2:
  587. sf[0] = get_bits(&s->gb, 6);
  588. sf[1] = sf[0];
  589. sf[2] = sf[0];
  590. break;
  591. case 1:
  592. sf[0] = get_bits(&s->gb, 6);
  593. sf[2] = get_bits(&s->gb, 6);
  594. sf[1] = sf[0];
  595. break;
  596. case 3:
  597. sf[0] = get_bits(&s->gb, 6);
  598. sf[2] = get_bits(&s->gb, 6);
  599. sf[1] = sf[2];
  600. break;
  601. }
  602. }
  603. }
  604. }
  605. /* samples */
  606. for (k = 0; k < 3; k++) {
  607. for (l = 0; l < 12; l += 3) {
  608. j = 0;
  609. for (i = 0; i < bound; i++) {
  610. bit_alloc_bits = alloc_table[j];
  611. for (ch = 0; ch < s->nb_channels; ch++) {
  612. b = bit_alloc[ch][i];
  613. if (b) {
  614. scale = scale_factors[ch][i][k];
  615. qindex = alloc_table[j+b];
  616. bits = ff_mpa_quant_bits[qindex];
  617. if (bits < 0) {
  618. int v2;
  619. /* 3 values at the same time */
  620. v = get_bits(&s->gb, -bits);
  621. v2 = division_tabs[qindex][v];
  622. steps = ff_mpa_quant_steps[qindex];
  623. s->sb_samples[ch][k * 12 + l + 0][i] =
  624. l2_unscale_group(steps, v2 & 15, scale);
  625. s->sb_samples[ch][k * 12 + l + 1][i] =
  626. l2_unscale_group(steps, (v2 >> 4) & 15, scale);
  627. s->sb_samples[ch][k * 12 + l + 2][i] =
  628. l2_unscale_group(steps, v2 >> 8 , scale);
  629. } else {
  630. for (m = 0; m < 3; m++) {
  631. v = get_bits(&s->gb, bits);
  632. v = l1_unscale(bits - 1, v, scale);
  633. s->sb_samples[ch][k * 12 + l + m][i] = v;
  634. }
  635. }
  636. } else {
  637. s->sb_samples[ch][k * 12 + l + 0][i] = 0;
  638. s->sb_samples[ch][k * 12 + l + 1][i] = 0;
  639. s->sb_samples[ch][k * 12 + l + 2][i] = 0;
  640. }
  641. }
  642. /* next subband in alloc table */
  643. j += 1 << bit_alloc_bits;
  644. }
  645. /* XXX: find a way to avoid this duplication of code */
  646. for (i = bound; i < sblimit; i++) {
  647. bit_alloc_bits = alloc_table[j];
  648. b = bit_alloc[0][i];
  649. if (b) {
  650. int mant, scale0, scale1;
  651. scale0 = scale_factors[0][i][k];
  652. scale1 = scale_factors[1][i][k];
  653. qindex = alloc_table[j+b];
  654. bits = ff_mpa_quant_bits[qindex];
  655. if (bits < 0) {
  656. /* 3 values at the same time */
  657. v = get_bits(&s->gb, -bits);
  658. steps = ff_mpa_quant_steps[qindex];
  659. mant = v % steps;
  660. v = v / steps;
  661. s->sb_samples[0][k * 12 + l + 0][i] =
  662. l2_unscale_group(steps, mant, scale0);
  663. s->sb_samples[1][k * 12 + l + 0][i] =
  664. l2_unscale_group(steps, mant, scale1);
  665. mant = v % steps;
  666. v = v / steps;
  667. s->sb_samples[0][k * 12 + l + 1][i] =
  668. l2_unscale_group(steps, mant, scale0);
  669. s->sb_samples[1][k * 12 + l + 1][i] =
  670. l2_unscale_group(steps, mant, scale1);
  671. s->sb_samples[0][k * 12 + l + 2][i] =
  672. l2_unscale_group(steps, v, scale0);
  673. s->sb_samples[1][k * 12 + l + 2][i] =
  674. l2_unscale_group(steps, v, scale1);
  675. } else {
  676. for (m = 0; m < 3; m++) {
  677. mant = get_bits(&s->gb, bits);
  678. s->sb_samples[0][k * 12 + l + m][i] =
  679. l1_unscale(bits - 1, mant, scale0);
  680. s->sb_samples[1][k * 12 + l + m][i] =
  681. l1_unscale(bits - 1, mant, scale1);
  682. }
  683. }
  684. } else {
  685. s->sb_samples[0][k * 12 + l + 0][i] = 0;
  686. s->sb_samples[0][k * 12 + l + 1][i] = 0;
  687. s->sb_samples[0][k * 12 + l + 2][i] = 0;
  688. s->sb_samples[1][k * 12 + l + 0][i] = 0;
  689. s->sb_samples[1][k * 12 + l + 1][i] = 0;
  690. s->sb_samples[1][k * 12 + l + 2][i] = 0;
  691. }
  692. /* next subband in alloc table */
  693. j += 1 << bit_alloc_bits;
  694. }
  695. /* fill remaining samples to zero */
  696. for (i = sblimit; i < SBLIMIT; i++) {
  697. for (ch = 0; ch < s->nb_channels; ch++) {
  698. s->sb_samples[ch][k * 12 + l + 0][i] = 0;
  699. s->sb_samples[ch][k * 12 + l + 1][i] = 0;
  700. s->sb_samples[ch][k * 12 + l + 2][i] = 0;
  701. }
  702. }
  703. }
  704. }
  705. return 3 * 12;
  706. }
  707. #define SPLIT(dst,sf,n) \
  708. if (n == 3) { \
  709. int m = (sf * 171) >> 9; \
  710. dst = sf - 3 * m; \
  711. sf = m; \
  712. } else if (n == 4) { \
  713. dst = sf & 3; \
  714. sf >>= 2; \
  715. } else if (n == 5) { \
  716. int m = (sf * 205) >> 10; \
  717. dst = sf - 5 * m; \
  718. sf = m; \
  719. } else if (n == 6) { \
  720. int m = (sf * 171) >> 10; \
  721. dst = sf - 6 * m; \
  722. sf = m; \
  723. } else { \
  724. dst = 0; \
  725. }
  726. static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2,
  727. int n3)
  728. {
  729. SPLIT(slen[3], sf, n3)
  730. SPLIT(slen[2], sf, n2)
  731. SPLIT(slen[1], sf, n1)
  732. slen[0] = sf;
  733. }
  734. static void exponents_from_scale_factors(MPADecodeContext *s, GranuleDef *g,
  735. int16_t *exponents)
  736. {
  737. const uint8_t *bstab, *pretab;
  738. int len, i, j, k, l, v0, shift, gain, gains[3];
  739. int16_t *exp_ptr;
  740. exp_ptr = exponents;
  741. gain = g->global_gain - 210;
  742. shift = g->scalefac_scale + 1;
  743. bstab = band_size_long[s->sample_rate_index];
  744. pretab = mpa_pretab[g->preflag];
  745. for (i = 0; i < g->long_end; i++) {
  746. v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
  747. len = bstab[i];
  748. for (j = len; j > 0; j--)
  749. *exp_ptr++ = v0;
  750. }
  751. if (g->short_start < 13) {
  752. bstab = band_size_short[s->sample_rate_index];
  753. gains[0] = gain - (g->subblock_gain[0] << 3);
  754. gains[1] = gain - (g->subblock_gain[1] << 3);
  755. gains[2] = gain - (g->subblock_gain[2] << 3);
  756. k = g->long_end;
  757. for (i = g->short_start; i < 13; i++) {
  758. len = bstab[i];
  759. for (l = 0; l < 3; l++) {
  760. v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
  761. for (j = len; j > 0; j--)
  762. *exp_ptr++ = v0;
  763. }
  764. }
  765. }
  766. }
  767. static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos,
  768. int *end_pos2)
  769. {
  770. if (s->in_gb.buffer && *pos >= s->gb.size_in_bits - s->extrasize * 8) {
  771. s->gb = s->in_gb;
  772. s->in_gb.buffer = NULL;
  773. s->extrasize = 0;
  774. av_assert2((get_bits_count(&s->gb) & 7) == 0);
  775. skip_bits_long(&s->gb, *pos - *end_pos);
  776. *end_pos2 =
  777. *end_pos = *end_pos2 + get_bits_count(&s->gb) - *pos;
  778. *pos = get_bits_count(&s->gb);
  779. }
  780. }
  781. /* Following is an optimized code for
  782. INTFLOAT v = *src
  783. if(get_bits1(&s->gb))
  784. v = -v;
  785. *dst = v;
  786. */
  787. #if USE_FLOATS
  788. #define READ_FLIP_SIGN(dst,src) \
  789. v = AV_RN32A(src) ^ (get_bits1(&s->gb) << 31); \
  790. AV_WN32A(dst, v);
  791. #else
  792. #define READ_FLIP_SIGN(dst,src) \
  793. v = -get_bits1(&s->gb); \
  794. *(dst) = (*(src) ^ v) - v;
  795. #endif
  796. static int huffman_decode(MPADecodeContext *s, GranuleDef *g,
  797. int16_t *exponents, int end_pos2)
  798. {
  799. int s_index;
  800. int i;
  801. int last_pos, bits_left;
  802. VLC *vlc;
  803. int end_pos = FFMIN(end_pos2, s->gb.size_in_bits - s->extrasize * 8);
  804. /* low frequencies (called big values) */
  805. s_index = 0;
  806. for (i = 0; i < 3; i++) {
  807. int j, k, l, linbits;
  808. j = g->region_size[i];
  809. if (j == 0)
  810. continue;
  811. /* select vlc table */
  812. k = g->table_select[i];
  813. l = mpa_huff_data[k][0];
  814. linbits = mpa_huff_data[k][1];
  815. vlc = &huff_vlc[l];
  816. if (!l) {
  817. memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * 2 * j);
  818. s_index += 2 * j;
  819. continue;
  820. }
  821. /* read huffcode and compute each couple */
  822. for (; j > 0; j--) {
  823. int exponent, x, y;
  824. int v;
  825. int pos = get_bits_count(&s->gb);
  826. if (pos >= end_pos){
  827. switch_buffer(s, &pos, &end_pos, &end_pos2);
  828. if (pos >= end_pos)
  829. break;
  830. }
  831. y = get_vlc2(&s->gb, vlc->table, 7, 3);
  832. if (!y) {
  833. g->sb_hybrid[s_index ] =
  834. g->sb_hybrid[s_index+1] = 0;
  835. s_index += 2;
  836. continue;
  837. }
  838. exponent= exponents[s_index];
  839. ff_dlog(s->avctx, "region=%d n=%d y=%d exp=%d\n",
  840. i, g->region_size[i] - j, y, exponent);
  841. if (y & 16) {
  842. x = y >> 5;
  843. y = y & 0x0f;
  844. if (x < 15) {
  845. READ_FLIP_SIGN(g->sb_hybrid + s_index, RENAME(expval_table)[exponent] + x)
  846. } else {
  847. x += get_bitsz(&s->gb, linbits);
  848. v = l3_unscale(x, exponent);
  849. if (get_bits1(&s->gb))
  850. v = -v;
  851. g->sb_hybrid[s_index] = v;
  852. }
  853. if (y < 15) {
  854. READ_FLIP_SIGN(g->sb_hybrid + s_index + 1, RENAME(expval_table)[exponent] + y)
  855. } else {
  856. y += get_bitsz(&s->gb, linbits);
  857. v = l3_unscale(y, exponent);
  858. if (get_bits1(&s->gb))
  859. v = -v;
  860. g->sb_hybrid[s_index+1] = v;
  861. }
  862. } else {
  863. x = y >> 5;
  864. y = y & 0x0f;
  865. x += y;
  866. if (x < 15) {
  867. READ_FLIP_SIGN(g->sb_hybrid + s_index + !!y, RENAME(expval_table)[exponent] + x)
  868. } else {
  869. x += get_bitsz(&s->gb, linbits);
  870. v = l3_unscale(x, exponent);
  871. if (get_bits1(&s->gb))
  872. v = -v;
  873. g->sb_hybrid[s_index+!!y] = v;
  874. }
  875. g->sb_hybrid[s_index + !y] = 0;
  876. }
  877. s_index += 2;
  878. }
  879. }
  880. /* high frequencies */
  881. vlc = &huff_quad_vlc[g->count1table_select];
  882. last_pos = 0;
  883. while (s_index <= 572) {
  884. int pos, code;
  885. pos = get_bits_count(&s->gb);
  886. if (pos >= end_pos) {
  887. if (pos > end_pos2 && last_pos) {
  888. /* some encoders generate an incorrect size for this
  889. part. We must go back into the data */
  890. s_index -= 4;
  891. skip_bits_long(&s->gb, last_pos - pos);
  892. av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
  893. if(s->err_recognition & (AV_EF_BITSTREAM|AV_EF_COMPLIANT))
  894. s_index=0;
  895. break;
  896. }
  897. switch_buffer(s, &pos, &end_pos, &end_pos2);
  898. if (pos >= end_pos)
  899. break;
  900. }
  901. last_pos = pos;
  902. code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
  903. ff_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
  904. g->sb_hybrid[s_index+0] =
  905. g->sb_hybrid[s_index+1] =
  906. g->sb_hybrid[s_index+2] =
  907. g->sb_hybrid[s_index+3] = 0;
  908. while (code) {
  909. static const int idxtab[16] = { 3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0 };
  910. int v;
  911. int pos = s_index + idxtab[code];
  912. code ^= 8 >> idxtab[code];
  913. READ_FLIP_SIGN(g->sb_hybrid + pos, RENAME(exp_table)+exponents[pos])
  914. }
  915. s_index += 4;
  916. }
  917. /* skip extension bits */
  918. bits_left = end_pos2 - get_bits_count(&s->gb);
  919. if (bits_left < 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_COMPLIANT))) {
  920. av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
  921. s_index=0;
  922. } else if (bits_left > 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_AGGRESSIVE))) {
  923. av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
  924. s_index = 0;
  925. }
  926. memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * (576 - s_index));
  927. skip_bits_long(&s->gb, bits_left);
  928. i = get_bits_count(&s->gb);
  929. switch_buffer(s, &i, &end_pos, &end_pos2);
  930. return 0;
  931. }
  932. /* Reorder short blocks from bitstream order to interleaved order. It
  933. would be faster to do it in parsing, but the code would be far more
  934. complicated */
  935. static void reorder_block(MPADecodeContext *s, GranuleDef *g)
  936. {
  937. int i, j, len;
  938. INTFLOAT *ptr, *dst, *ptr1;
  939. INTFLOAT tmp[576];
  940. if (g->block_type != 2)
  941. return;
  942. if (g->switch_point) {
  943. if (s->sample_rate_index != 8)
  944. ptr = g->sb_hybrid + 36;
  945. else
  946. ptr = g->sb_hybrid + 72;
  947. } else {
  948. ptr = g->sb_hybrid;
  949. }
  950. for (i = g->short_start; i < 13; i++) {
  951. len = band_size_short[s->sample_rate_index][i];
  952. ptr1 = ptr;
  953. dst = tmp;
  954. for (j = len; j > 0; j--) {
  955. *dst++ = ptr[0*len];
  956. *dst++ = ptr[1*len];
  957. *dst++ = ptr[2*len];
  958. ptr++;
  959. }
  960. ptr += 2 * len;
  961. memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
  962. }
  963. }
  964. #define ISQRT2 FIXR(0.70710678118654752440)
  965. static void compute_stereo(MPADecodeContext *s, GranuleDef *g0, GranuleDef *g1)
  966. {
  967. int i, j, k, l;
  968. int sf_max, sf, len, non_zero_found;
  969. INTFLOAT (*is_tab)[16], *tab0, *tab1, v1, v2;
  970. SUINTFLOAT tmp0, tmp1;
  971. int non_zero_found_short[3];
  972. /* intensity stereo */
  973. if (s->mode_ext & MODE_EXT_I_STEREO) {
  974. if (!s->lsf) {
  975. is_tab = is_table;
  976. sf_max = 7;
  977. } else {
  978. is_tab = is_table_lsf[g1->scalefac_compress & 1];
  979. sf_max = 16;
  980. }
  981. tab0 = g0->sb_hybrid + 576;
  982. tab1 = g1->sb_hybrid + 576;
  983. non_zero_found_short[0] = 0;
  984. non_zero_found_short[1] = 0;
  985. non_zero_found_short[2] = 0;
  986. k = (13 - g1->short_start) * 3 + g1->long_end - 3;
  987. for (i = 12; i >= g1->short_start; i--) {
  988. /* for last band, use previous scale factor */
  989. if (i != 11)
  990. k -= 3;
  991. len = band_size_short[s->sample_rate_index][i];
  992. for (l = 2; l >= 0; l--) {
  993. tab0 -= len;
  994. tab1 -= len;
  995. if (!non_zero_found_short[l]) {
  996. /* test if non zero band. if so, stop doing i-stereo */
  997. for (j = 0; j < len; j++) {
  998. if (tab1[j] != 0) {
  999. non_zero_found_short[l] = 1;
  1000. goto found1;
  1001. }
  1002. }
  1003. sf = g1->scale_factors[k + l];
  1004. if (sf >= sf_max)
  1005. goto found1;
  1006. v1 = is_tab[0][sf];
  1007. v2 = is_tab[1][sf];
  1008. for (j = 0; j < len; j++) {
  1009. tmp0 = tab0[j];
  1010. tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
  1011. tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
  1012. }
  1013. } else {
  1014. found1:
  1015. if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1016. /* lower part of the spectrum : do ms stereo
  1017. if enabled */
  1018. for (j = 0; j < len; j++) {
  1019. tmp0 = tab0[j];
  1020. tmp1 = tab1[j];
  1021. tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
  1022. tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
  1023. }
  1024. }
  1025. }
  1026. }
  1027. }
  1028. non_zero_found = non_zero_found_short[0] |
  1029. non_zero_found_short[1] |
  1030. non_zero_found_short[2];
  1031. for (i = g1->long_end - 1;i >= 0;i--) {
  1032. len = band_size_long[s->sample_rate_index][i];
  1033. tab0 -= len;
  1034. tab1 -= len;
  1035. /* test if non zero band. if so, stop doing i-stereo */
  1036. if (!non_zero_found) {
  1037. for (j = 0; j < len; j++) {
  1038. if (tab1[j] != 0) {
  1039. non_zero_found = 1;
  1040. goto found2;
  1041. }
  1042. }
  1043. /* for last band, use previous scale factor */
  1044. k = (i == 21) ? 20 : i;
  1045. sf = g1->scale_factors[k];
  1046. if (sf >= sf_max)
  1047. goto found2;
  1048. v1 = is_tab[0][sf];
  1049. v2 = is_tab[1][sf];
  1050. for (j = 0; j < len; j++) {
  1051. tmp0 = tab0[j];
  1052. tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
  1053. tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
  1054. }
  1055. } else {
  1056. found2:
  1057. if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1058. /* lower part of the spectrum : do ms stereo
  1059. if enabled */
  1060. for (j = 0; j < len; j++) {
  1061. tmp0 = tab0[j];
  1062. tmp1 = tab1[j];
  1063. tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
  1064. tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
  1065. }
  1066. }
  1067. }
  1068. }
  1069. } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
  1070. /* ms stereo ONLY */
  1071. /* NOTE: the 1/sqrt(2) normalization factor is included in the
  1072. global gain */
  1073. #if USE_FLOATS
  1074. s->fdsp->butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);
  1075. #else
  1076. tab0 = g0->sb_hybrid;
  1077. tab1 = g1->sb_hybrid;
  1078. for (i = 0; i < 576; i++) {
  1079. tmp0 = tab0[i];
  1080. tmp1 = tab1[i];
  1081. tab0[i] = tmp0 + tmp1;
  1082. tab1[i] = tmp0 - tmp1;
  1083. }
  1084. #endif
  1085. }
  1086. }
  1087. #if USE_FLOATS
  1088. #if HAVE_MIPSFPU
  1089. # include "mips/compute_antialias_float.h"
  1090. #endif /* HAVE_MIPSFPU */
  1091. #else
  1092. #if HAVE_MIPSDSP
  1093. # include "mips/compute_antialias_fixed.h"
  1094. #endif /* HAVE_MIPSDSP */
  1095. #endif /* USE_FLOATS */
  1096. #ifndef compute_antialias
  1097. #if USE_FLOATS
  1098. #define AA(j) do { \
  1099. float tmp0 = ptr[-1-j]; \
  1100. float tmp1 = ptr[ j]; \
  1101. ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1]; \
  1102. ptr[ j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0]; \
  1103. } while (0)
  1104. #else
  1105. #define AA(j) do { \
  1106. SUINT tmp0 = ptr[-1-j]; \
  1107. SUINT tmp1 = ptr[ j]; \
  1108. SUINT tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]); \
  1109. ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2])); \
  1110. ptr[ j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3])); \
  1111. } while (0)
  1112. #endif
  1113. static void compute_antialias(MPADecodeContext *s, GranuleDef *g)
  1114. {
  1115. INTFLOAT *ptr;
  1116. int n, i;
  1117. /* we antialias only "long" bands */
  1118. if (g->block_type == 2) {
  1119. if (!g->switch_point)
  1120. return;
  1121. /* XXX: check this for 8000Hz case */
  1122. n = 1;
  1123. } else {
  1124. n = SBLIMIT - 1;
  1125. }
  1126. ptr = g->sb_hybrid + 18;
  1127. for (i = n; i > 0; i--) {
  1128. AA(0);
  1129. AA(1);
  1130. AA(2);
  1131. AA(3);
  1132. AA(4);
  1133. AA(5);
  1134. AA(6);
  1135. AA(7);
  1136. ptr += 18;
  1137. }
  1138. }
  1139. #endif /* compute_antialias */
  1140. static void compute_imdct(MPADecodeContext *s, GranuleDef *g,
  1141. INTFLOAT *sb_samples, INTFLOAT *mdct_buf)
  1142. {
  1143. INTFLOAT *win, *out_ptr, *ptr, *buf, *ptr1;
  1144. INTFLOAT out2[12];
  1145. int i, j, mdct_long_end, sblimit;
  1146. /* find last non zero block */
  1147. ptr = g->sb_hybrid + 576;
  1148. ptr1 = g->sb_hybrid + 2 * 18;
  1149. while (ptr >= ptr1) {
  1150. int32_t *p;
  1151. ptr -= 6;
  1152. p = (int32_t*)ptr;
  1153. if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
  1154. break;
  1155. }
  1156. sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
  1157. if (g->block_type == 2) {
  1158. /* XXX: check for 8000 Hz */
  1159. if (g->switch_point)
  1160. mdct_long_end = 2;
  1161. else
  1162. mdct_long_end = 0;
  1163. } else {
  1164. mdct_long_end = sblimit;
  1165. }
  1166. s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid,
  1167. mdct_long_end, g->switch_point,
  1168. g->block_type);
  1169. buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3);
  1170. ptr = g->sb_hybrid + 18 * mdct_long_end;
  1171. for (j = mdct_long_end; j < sblimit; j++) {
  1172. /* select frequency inversion */
  1173. win = RENAME(ff_mdct_win)[2 + (4 & -(j & 1))];
  1174. out_ptr = sb_samples + j;
  1175. for (i = 0; i < 6; i++) {
  1176. *out_ptr = buf[4*i];
  1177. out_ptr += SBLIMIT;
  1178. }
  1179. imdct12(out2, ptr + 0);
  1180. for (i = 0; i < 6; i++) {
  1181. *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*1)];
  1182. buf[4*(i + 6*2)] = MULH3(out2[i + 6], win[i + 6], 1);
  1183. out_ptr += SBLIMIT;
  1184. }
  1185. imdct12(out2, ptr + 1);
  1186. for (i = 0; i < 6; i++) {
  1187. *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*2)];
  1188. buf[4*(i + 6*0)] = MULH3(out2[i + 6], win[i + 6], 1);
  1189. out_ptr += SBLIMIT;
  1190. }
  1191. imdct12(out2, ptr + 2);
  1192. for (i = 0; i < 6; i++) {
  1193. buf[4*(i + 6*0)] = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*0)];
  1194. buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1);
  1195. buf[4*(i + 6*2)] = 0;
  1196. }
  1197. ptr += 18;
  1198. buf += (j&3) != 3 ? 1 : (4*18-3);
  1199. }
  1200. /* zero bands */
  1201. for (j = sblimit; j < SBLIMIT; j++) {
  1202. /* overlap */
  1203. out_ptr = sb_samples + j;
  1204. for (i = 0; i < 18; i++) {
  1205. *out_ptr = buf[4*i];
  1206. buf[4*i] = 0;
  1207. out_ptr += SBLIMIT;
  1208. }
  1209. buf += (j&3) != 3 ? 1 : (4*18-3);
  1210. }
  1211. }
  1212. /* main layer3 decoding function */
  1213. static int mp_decode_layer3(MPADecodeContext *s)
  1214. {
  1215. int nb_granules, main_data_begin;
  1216. int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
  1217. GranuleDef *g;
  1218. int16_t exponents[576]; //FIXME try INTFLOAT
  1219. int ret;
  1220. /* read side info */
  1221. if (s->lsf) {
  1222. ret = handle_crc(s, ((s->nb_channels == 1) ? 8*9 : 8*17));
  1223. main_data_begin = get_bits(&s->gb, 8);
  1224. skip_bits(&s->gb, s->nb_channels);
  1225. nb_granules = 1;
  1226. } else {
  1227. ret = handle_crc(s, ((s->nb_channels == 1) ? 8*17 : 8*32));
  1228. main_data_begin = get_bits(&s->gb, 9);
  1229. if (s->nb_channels == 2)
  1230. skip_bits(&s->gb, 3);
  1231. else
  1232. skip_bits(&s->gb, 5);
  1233. nb_granules = 2;
  1234. for (ch = 0; ch < s->nb_channels; ch++) {
  1235. s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
  1236. s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
  1237. }
  1238. }
  1239. if (ret < 0)
  1240. return ret;
  1241. for (gr = 0; gr < nb_granules; gr++) {
  1242. for (ch = 0; ch < s->nb_channels; ch++) {
  1243. ff_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
  1244. g = &s->granules[ch][gr];
  1245. g->part2_3_length = get_bits(&s->gb, 12);
  1246. g->big_values = get_bits(&s->gb, 9);
  1247. if (g->big_values > 288) {
  1248. av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
  1249. return AVERROR_INVALIDDATA;
  1250. }
  1251. g->global_gain = get_bits(&s->gb, 8);
  1252. /* if MS stereo only is selected, we precompute the
  1253. 1/sqrt(2) renormalization factor */
  1254. if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
  1255. MODE_EXT_MS_STEREO)
  1256. g->global_gain -= 2;
  1257. if (s->lsf)
  1258. g->scalefac_compress = get_bits(&s->gb, 9);
  1259. else
  1260. g->scalefac_compress = get_bits(&s->gb, 4);
  1261. blocksplit_flag = get_bits1(&s->gb);
  1262. if (blocksplit_flag) {
  1263. g->block_type = get_bits(&s->gb, 2);
  1264. if (g->block_type == 0) {
  1265. av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
  1266. return AVERROR_INVALIDDATA;
  1267. }
  1268. g->switch_point = get_bits1(&s->gb);
  1269. for (i = 0; i < 2; i++)
  1270. g->table_select[i] = get_bits(&s->gb, 5);
  1271. for (i = 0; i < 3; i++)
  1272. g->subblock_gain[i] = get_bits(&s->gb, 3);
  1273. init_short_region(s, g);
  1274. } else {
  1275. int region_address1, region_address2;
  1276. g->block_type = 0;
  1277. g->switch_point = 0;
  1278. for (i = 0; i < 3; i++)
  1279. g->table_select[i] = get_bits(&s->gb, 5);
  1280. /* compute huffman coded region sizes */
  1281. region_address1 = get_bits(&s->gb, 4);
  1282. region_address2 = get_bits(&s->gb, 3);
  1283. ff_dlog(s->avctx, "region1=%d region2=%d\n",
  1284. region_address1, region_address2);
  1285. init_long_region(s, g, region_address1, region_address2);
  1286. }
  1287. region_offset2size(g);
  1288. compute_band_indexes(s, g);
  1289. g->preflag = 0;
  1290. if (!s->lsf)
  1291. g->preflag = get_bits1(&s->gb);
  1292. g->scalefac_scale = get_bits1(&s->gb);
  1293. g->count1table_select = get_bits1(&s->gb);
  1294. ff_dlog(s->avctx, "block_type=%d switch_point=%d\n",
  1295. g->block_type, g->switch_point);
  1296. }
  1297. }
  1298. if (!s->adu_mode) {
  1299. int skip;
  1300. const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
  1301. s->extrasize = av_clip((get_bits_left(&s->gb) >> 3) - s->extrasize, 0,
  1302. FFMAX(0, LAST_BUF_SIZE - s->last_buf_size));
  1303. av_assert1((get_bits_count(&s->gb) & 7) == 0);
  1304. /* now we get bits from the main_data_begin offset */
  1305. ff_dlog(s->avctx, "seekback:%d, lastbuf:%d\n",
  1306. main_data_begin, s->last_buf_size);
  1307. memcpy(s->last_buf + s->last_buf_size, ptr, s->extrasize);
  1308. s->in_gb = s->gb;
  1309. init_get_bits(&s->gb, s->last_buf, (s->last_buf_size + s->extrasize) * 8);
  1310. s->last_buf_size <<= 3;
  1311. for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) {
  1312. for (ch = 0; ch < s->nb_channels; ch++) {
  1313. g = &s->granules[ch][gr];
  1314. s->last_buf_size += g->part2_3_length;
  1315. memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
  1316. compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
  1317. }
  1318. }
  1319. skip = s->last_buf_size - 8 * main_data_begin;
  1320. if (skip >= s->gb.size_in_bits - s->extrasize * 8 && s->in_gb.buffer) {
  1321. skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits + s->extrasize * 8);
  1322. s->gb = s->in_gb;
  1323. s->in_gb.buffer = NULL;
  1324. s->extrasize = 0;
  1325. } else {
  1326. skip_bits_long(&s->gb, skip);
  1327. }
  1328. } else {
  1329. gr = 0;
  1330. s->extrasize = 0;
  1331. }
  1332. for (; gr < nb_granules; gr++) {
  1333. for (ch = 0; ch < s->nb_channels; ch++) {
  1334. g = &s->granules[ch][gr];
  1335. bits_pos = get_bits_count(&s->gb);
  1336. if (!s->lsf) {
  1337. uint8_t *sc;
  1338. int slen, slen1, slen2;
  1339. /* MPEG-1 scale factors */
  1340. slen1 = slen_table[0][g->scalefac_compress];
  1341. slen2 = slen_table[1][g->scalefac_compress];
  1342. ff_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
  1343. if (g->block_type == 2) {
  1344. n = g->switch_point ? 17 : 18;
  1345. j = 0;
  1346. if (slen1) {
  1347. for (i = 0; i < n; i++)
  1348. g->scale_factors[j++] = get_bits(&s->gb, slen1);
  1349. } else {
  1350. for (i = 0; i < n; i++)
  1351. g->scale_factors[j++] = 0;
  1352. }
  1353. if (slen2) {
  1354. for (i = 0; i < 18; i++)
  1355. g->scale_factors[j++] = get_bits(&s->gb, slen2);
  1356. for (i = 0; i < 3; i++)
  1357. g->scale_factors[j++] = 0;
  1358. } else {
  1359. for (i = 0; i < 21; i++)
  1360. g->scale_factors[j++] = 0;
  1361. }
  1362. } else {
  1363. sc = s->granules[ch][0].scale_factors;
  1364. j = 0;
  1365. for (k = 0; k < 4; k++) {
  1366. n = k == 0 ? 6 : 5;
  1367. if ((g->scfsi & (0x8 >> k)) == 0) {
  1368. slen = (k < 2) ? slen1 : slen2;
  1369. if (slen) {
  1370. for (i = 0; i < n; i++)
  1371. g->scale_factors[j++] = get_bits(&s->gb, slen);
  1372. } else {
  1373. for (i = 0; i < n; i++)
  1374. g->scale_factors[j++] = 0;
  1375. }
  1376. } else {
  1377. /* simply copy from last granule */
  1378. for (i = 0; i < n; i++) {
  1379. g->scale_factors[j] = sc[j];
  1380. j++;
  1381. }
  1382. }
  1383. }
  1384. g->scale_factors[j++] = 0;
  1385. }
  1386. } else {
  1387. int tindex, tindex2, slen[4], sl, sf;
  1388. /* LSF scale factors */
  1389. if (g->block_type == 2)
  1390. tindex = g->switch_point ? 2 : 1;
  1391. else
  1392. tindex = 0;
  1393. sf = g->scalefac_compress;
  1394. if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
  1395. /* intensity stereo case */
  1396. sf >>= 1;
  1397. if (sf < 180) {
  1398. lsf_sf_expand(slen, sf, 6, 6, 0);
  1399. tindex2 = 3;
  1400. } else if (sf < 244) {
  1401. lsf_sf_expand(slen, sf - 180, 4, 4, 0);
  1402. tindex2 = 4;
  1403. } else {
  1404. lsf_sf_expand(slen, sf - 244, 3, 0, 0);
  1405. tindex2 = 5;
  1406. }
  1407. } else {
  1408. /* normal case */
  1409. if (sf < 400) {
  1410. lsf_sf_expand(slen, sf, 5, 4, 4);
  1411. tindex2 = 0;
  1412. } else if (sf < 500) {
  1413. lsf_sf_expand(slen, sf - 400, 5, 4, 0);
  1414. tindex2 = 1;
  1415. } else {
  1416. lsf_sf_expand(slen, sf - 500, 3, 0, 0);
  1417. tindex2 = 2;
  1418. g->preflag = 1;
  1419. }
  1420. }
  1421. j = 0;
  1422. for (k = 0; k < 4; k++) {
  1423. n = lsf_nsf_table[tindex2][tindex][k];
  1424. sl = slen[k];
  1425. if (sl) {
  1426. for (i = 0; i < n; i++)
  1427. g->scale_factors[j++] = get_bits(&s->gb, sl);
  1428. } else {
  1429. for (i = 0; i < n; i++)
  1430. g->scale_factors[j++] = 0;
  1431. }
  1432. }
  1433. /* XXX: should compute exact size */
  1434. for (; j < 40; j++)
  1435. g->scale_factors[j] = 0;
  1436. }
  1437. exponents_from_scale_factors(s, g, exponents);
  1438. /* read Huffman coded residue */
  1439. huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
  1440. } /* ch */
  1441. if (s->mode == MPA_JSTEREO)
  1442. compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
  1443. for (ch = 0; ch < s->nb_channels; ch++) {
  1444. g = &s->granules[ch][gr];
  1445. reorder_block(s, g);
  1446. compute_antialias(s, g);
  1447. compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
  1448. }
  1449. } /* gr */
  1450. if (get_bits_count(&s->gb) < 0)
  1451. skip_bits_long(&s->gb, -get_bits_count(&s->gb));
  1452. return nb_granules * 18;
  1453. }
  1454. static int mp_decode_frame(MPADecodeContext *s, OUT_INT **samples,
  1455. const uint8_t *buf, int buf_size)
  1456. {
  1457. int i, nb_frames, ch, ret;
  1458. OUT_INT *samples_ptr;
  1459. init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);
  1460. if (s->error_protection)
  1461. s->crc = get_bits(&s->gb, 16);
  1462. switch(s->layer) {
  1463. case 1:
  1464. s->avctx->frame_size = 384;
  1465. nb_frames = mp_decode_layer1(s);
  1466. break;
  1467. case 2:
  1468. s->avctx->frame_size = 1152;
  1469. nb_frames = mp_decode_layer2(s);
  1470. break;
  1471. case 3:
  1472. s->avctx->frame_size = s->lsf ? 576 : 1152;
  1473. default:
  1474. nb_frames = mp_decode_layer3(s);
  1475. s->last_buf_size=0;
  1476. if (s->in_gb.buffer) {
  1477. align_get_bits(&s->gb);
  1478. i = (get_bits_left(&s->gb) >> 3) - s->extrasize;
  1479. if (i >= 0 && i <= BACKSTEP_SIZE) {
  1480. memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);
  1481. s->last_buf_size=i;
  1482. } else
  1483. av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
  1484. s->gb = s->in_gb;
  1485. s->in_gb.buffer = NULL;
  1486. s->extrasize = 0;
  1487. }
  1488. align_get_bits(&s->gb);
  1489. av_assert1((get_bits_count(&s->gb) & 7) == 0);
  1490. i = (get_bits_left(&s->gb) >> 3) - s->extrasize;
  1491. if (i < 0 || i > BACKSTEP_SIZE || nb_frames < 0) {
  1492. if (i < 0)
  1493. av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
  1494. i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
  1495. }
  1496. av_assert1(i <= buf_size - HEADER_SIZE && i >= 0);
  1497. memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
  1498. s->last_buf_size += i;
  1499. }
  1500. if(nb_frames < 0)
  1501. return nb_frames;
  1502. /* get output buffer */
  1503. if (!samples) {
  1504. av_assert0(s->frame);
  1505. s->frame->nb_samples = s->avctx->frame_size;
  1506. if ((ret = ff_get_buffer(s->avctx, s->frame, 0)) < 0)
  1507. return ret;
  1508. samples = (OUT_INT **)s->frame->extended_data;
  1509. }
  1510. /* apply the synthesis filter */
  1511. for (ch = 0; ch < s->nb_channels; ch++) {
  1512. int sample_stride;
  1513. if (s->avctx->sample_fmt == OUT_FMT_P) {
  1514. samples_ptr = samples[ch];
  1515. sample_stride = 1;
  1516. } else {
  1517. samples_ptr = samples[0] + ch;
  1518. sample_stride = s->nb_channels;
  1519. }
  1520. for (i = 0; i < nb_frames; i++) {
  1521. RENAME(ff_mpa_synth_filter)(&s->mpadsp, s->synth_buf[ch],
  1522. &(s->synth_buf_offset[ch]),
  1523. RENAME(ff_mpa_synth_window),
  1524. &s->dither_state, samples_ptr,
  1525. sample_stride, s->sb_samples[ch][i]);
  1526. samples_ptr += 32 * sample_stride;
  1527. }
  1528. }
  1529. return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
  1530. }
  1531. static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,
  1532. AVPacket *avpkt)
  1533. {
  1534. const uint8_t *buf = avpkt->data;
  1535. int buf_size = avpkt->size;
  1536. MPADecodeContext *s = avctx->priv_data;
  1537. uint32_t header;
  1538. int ret;
  1539. int skipped = 0;
  1540. while(buf_size && !*buf){
  1541. buf++;
  1542. buf_size--;
  1543. skipped++;
  1544. }
  1545. if (buf_size < HEADER_SIZE)
  1546. return AVERROR_INVALIDDATA;
  1547. header = AV_RB32(buf);
  1548. if (header>>8 == AV_RB32("TAG")>>8) {
  1549. av_log(avctx, AV_LOG_DEBUG, "discarding ID3 tag\n");
  1550. return buf_size + skipped;
  1551. }
  1552. ret = avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);
  1553. if (ret < 0) {
  1554. av_log(avctx, AV_LOG_ERROR, "Header missing\n");
  1555. return AVERROR_INVALIDDATA;
  1556. } else if (ret == 1) {
  1557. /* free format: prepare to compute frame size */
  1558. s->frame_size = -1;
  1559. return AVERROR_INVALIDDATA;
  1560. }
  1561. /* update codec info */
  1562. avctx->channels = s->nb_channels;
  1563. avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
  1564. if (!avctx->bit_rate)
  1565. avctx->bit_rate = s->bit_rate;
  1566. if (s->frame_size <= 0) {
  1567. av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
  1568. return AVERROR_INVALIDDATA;
  1569. } else if (s->frame_size < buf_size) {
  1570. av_log(avctx, AV_LOG_DEBUG, "incorrect frame size - multiple frames in buffer?\n");
  1571. buf_size= s->frame_size;
  1572. }
  1573. s->frame = data;
  1574. ret = mp_decode_frame(s, NULL, buf, buf_size);
  1575. if (ret >= 0) {
  1576. s->frame->nb_samples = avctx->frame_size;
  1577. *got_frame_ptr = 1;
  1578. avctx->sample_rate = s->sample_rate;
  1579. //FIXME maybe move the other codec info stuff from above here too
  1580. } else {
  1581. av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
  1582. /* Only return an error if the bad frame makes up the whole packet or
  1583. * the error is related to buffer management.
  1584. * If there is more data in the packet, just consume the bad frame
  1585. * instead of returning an error, which would discard the whole
  1586. * packet. */
  1587. *got_frame_ptr = 0;
  1588. if (buf_size == avpkt->size || ret != AVERROR_INVALIDDATA)
  1589. return ret;
  1590. }
  1591. s->frame_size = 0;
  1592. return buf_size + skipped;
  1593. }
  1594. static void mp_flush(MPADecodeContext *ctx)
  1595. {
  1596. memset(ctx->synth_buf, 0, sizeof(ctx->synth_buf));
  1597. memset(ctx->mdct_buf, 0, sizeof(ctx->mdct_buf));
  1598. ctx->last_buf_size = 0;
  1599. ctx->dither_state = 0;
  1600. }
  1601. static void flush(AVCodecContext *avctx)
  1602. {
  1603. mp_flush(avctx->priv_data);
  1604. }
  1605. #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
  1606. static int decode_frame_adu(AVCodecContext *avctx, void *data,
  1607. int *got_frame_ptr, AVPacket *avpkt)
  1608. {
  1609. const uint8_t *buf = avpkt->data;
  1610. int buf_size = avpkt->size;
  1611. MPADecodeContext *s = avctx->priv_data;
  1612. uint32_t header;
  1613. int len, ret;
  1614. int av_unused out_size;
  1615. len = buf_size;
  1616. // Discard too short frames
  1617. if (buf_size < HEADER_SIZE) {
  1618. av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
  1619. return AVERROR_INVALIDDATA;
  1620. }
  1621. if (len > MPA_MAX_CODED_FRAME_SIZE)
  1622. len = MPA_MAX_CODED_FRAME_SIZE;
  1623. // Get header and restore sync word
  1624. header = AV_RB32(buf) | 0xffe00000;
  1625. ret = avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);
  1626. if (ret < 0) {
  1627. av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n");
  1628. return ret;
  1629. }
  1630. /* update codec info */
  1631. avctx->sample_rate = s->sample_rate;
  1632. avctx->channels = s->nb_channels;
  1633. avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
  1634. if (!avctx->bit_rate)
  1635. avctx->bit_rate = s->bit_rate;
  1636. s->frame_size = len;
  1637. s->frame = data;
  1638. ret = mp_decode_frame(s, NULL, buf, buf_size);
  1639. if (ret < 0) {
  1640. av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
  1641. return ret;
  1642. }
  1643. *got_frame_ptr = 1;
  1644. return buf_size;
  1645. }
  1646. #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
  1647. #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
  1648. /**
  1649. * Context for MP3On4 decoder
  1650. */
  1651. typedef struct MP3On4DecodeContext {
  1652. int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
  1653. int syncword; ///< syncword patch
  1654. const uint8_t *coff; ///< channel offsets in output buffer
  1655. MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
  1656. } MP3On4DecodeContext;
  1657. #include "mpeg4audio.h"
  1658. /* Next 3 arrays are indexed by channel config number (passed via codecdata) */
  1659. /* number of mp3 decoder instances */
  1660. static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 };
  1661. /* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */
  1662. static const uint8_t chan_offset[8][5] = {
  1663. { 0 },
  1664. { 0 }, // C
  1665. { 0 }, // FLR
  1666. { 2, 0 }, // C FLR
  1667. { 2, 0, 3 }, // C FLR BS
  1668. { 2, 0, 3 }, // C FLR BLRS
  1669. { 2, 0, 4, 3 }, // C FLR BLRS LFE
  1670. { 2, 0, 6, 4, 3 }, // C FLR BLRS BLR LFE
  1671. };
  1672. /* mp3on4 channel layouts */
  1673. static const int16_t chan_layout[8] = {
  1674. 0,
  1675. AV_CH_LAYOUT_MONO,
  1676. AV_CH_LAYOUT_STEREO,
  1677. AV_CH_LAYOUT_SURROUND,
  1678. AV_CH_LAYOUT_4POINT0,
  1679. AV_CH_LAYOUT_5POINT0,
  1680. AV_CH_LAYOUT_5POINT1,
  1681. AV_CH_LAYOUT_7POINT1
  1682. };
  1683. static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
  1684. {
  1685. MP3On4DecodeContext *s = avctx->priv_data;
  1686. int i;
  1687. if (s->mp3decctx[0])
  1688. av_freep(&s->mp3decctx[0]->fdsp);
  1689. for (i = 0; i < s->frames; i++)
  1690. av_freep(&s->mp3decctx[i]);
  1691. return 0;
  1692. }
  1693. static av_cold int decode_init_mp3on4(AVCodecContext * avctx)
  1694. {
  1695. MP3On4DecodeContext *s = avctx->priv_data;
  1696. MPEG4AudioConfig cfg;
  1697. int i;
  1698. if ((avctx->extradata_size < 2) || !avctx->extradata) {
  1699. av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
  1700. return AVERROR_INVALIDDATA;
  1701. }
  1702. avpriv_mpeg4audio_get_config2(&cfg, avctx->extradata,
  1703. avctx->extradata_size, 1, avctx);
  1704. if (!cfg.chan_config || cfg.chan_config > 7) {
  1705. av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
  1706. return AVERROR_INVALIDDATA;
  1707. }
  1708. s->frames = mp3Frames[cfg.chan_config];
  1709. s->coff = chan_offset[cfg.chan_config];
  1710. avctx->channels = ff_mpeg4audio_channels[cfg.chan_config];
  1711. avctx->channel_layout = chan_layout[cfg.chan_config];
  1712. if (cfg.sample_rate < 16000)
  1713. s->syncword = 0xffe00000;
  1714. else
  1715. s->syncword = 0xfff00000;
  1716. /* Init the first mp3 decoder in standard way, so that all tables get builded
  1717. * We replace avctx->priv_data with the context of the first decoder so that
  1718. * decode_init() does not have to be changed.
  1719. * Other decoders will be initialized here copying data from the first context
  1720. */
  1721. // Allocate zeroed memory for the first decoder context
  1722. s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
  1723. if (!s->mp3decctx[0])
  1724. goto alloc_fail;
  1725. // Put decoder context in place to make init_decode() happy
  1726. avctx->priv_data = s->mp3decctx[0];
  1727. decode_init(avctx);
  1728. // Restore mp3on4 context pointer
  1729. avctx->priv_data = s;
  1730. s->mp3decctx[0]->adu_mode = 1; // Set adu mode
  1731. /* Create a separate codec/context for each frame (first is already ok).
  1732. * Each frame is 1 or 2 channels - up to 5 frames allowed
  1733. */
  1734. for (i = 1; i < s->frames; i++) {
  1735. s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
  1736. if (!s->mp3decctx[i])
  1737. goto alloc_fail;
  1738. s->mp3decctx[i]->adu_mode = 1;
  1739. s->mp3decctx[i]->avctx = avctx;
  1740. s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;
  1741. s->mp3decctx[i]->fdsp = s->mp3decctx[0]->fdsp;
  1742. }
  1743. return 0;
  1744. alloc_fail:
  1745. decode_close_mp3on4(avctx);
  1746. return AVERROR(ENOMEM);
  1747. }
  1748. static void flush_mp3on4(AVCodecContext *avctx)
  1749. {
  1750. int i;
  1751. MP3On4DecodeContext *s = avctx->priv_data;
  1752. for (i = 0; i < s->frames; i++)
  1753. mp_flush(s->mp3decctx[i]);
  1754. }
  1755. static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
  1756. int *got_frame_ptr, AVPacket *avpkt)
  1757. {
  1758. AVFrame *frame = data;
  1759. const uint8_t *buf = avpkt->data;
  1760. int buf_size = avpkt->size;
  1761. MP3On4DecodeContext *s = avctx->priv_data;
  1762. MPADecodeContext *m;
  1763. int fsize, len = buf_size, out_size = 0;
  1764. uint32_t header;
  1765. OUT_INT **out_samples;
  1766. OUT_INT *outptr[2];
  1767. int fr, ch, ret;
  1768. /* get output buffer */
  1769. frame->nb_samples = MPA_FRAME_SIZE;
  1770. if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
  1771. return ret;
  1772. out_samples = (OUT_INT **)frame->extended_data;
  1773. // Discard too short frames
  1774. if (buf_size < HEADER_SIZE)
  1775. return AVERROR_INVALIDDATA;
  1776. avctx->bit_rate = 0;
  1777. ch = 0;
  1778. for (fr = 0; fr < s->frames; fr++) {
  1779. fsize = AV_RB16(buf) >> 4;
  1780. fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
  1781. m = s->mp3decctx[fr];
  1782. av_assert1(m);
  1783. if (fsize < HEADER_SIZE) {
  1784. av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n");
  1785. return AVERROR_INVALIDDATA;
  1786. }
  1787. header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
  1788. ret = avpriv_mpegaudio_decode_header((MPADecodeHeader *)m, header);
  1789. if (ret < 0) {
  1790. av_log(avctx, AV_LOG_ERROR, "Bad header, discard block\n");
  1791. return AVERROR_INVALIDDATA;
  1792. }
  1793. if (ch + m->nb_channels > avctx->channels ||
  1794. s->coff[fr] + m->nb_channels > avctx->channels) {
  1795. av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "
  1796. "channel count\n");
  1797. return AVERROR_INVALIDDATA;
  1798. }
  1799. ch += m->nb_channels;
  1800. outptr[0] = out_samples[s->coff[fr]];
  1801. if (m->nb_channels > 1)
  1802. outptr[1] = out_samples[s->coff[fr] + 1];
  1803. if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0) {
  1804. av_log(avctx, AV_LOG_ERROR, "failed to decode channel %d\n", ch);
  1805. memset(outptr[0], 0, MPA_FRAME_SIZE*sizeof(OUT_INT));
  1806. if (m->nb_channels > 1)
  1807. memset(outptr[1], 0, MPA_FRAME_SIZE*sizeof(OUT_INT));
  1808. ret = m->nb_channels * MPA_FRAME_SIZE*sizeof(OUT_INT);
  1809. }
  1810. out_size += ret;
  1811. buf += fsize;
  1812. len -= fsize;
  1813. avctx->bit_rate += m->bit_rate;
  1814. }
  1815. if (ch != avctx->channels) {
  1816. av_log(avctx, AV_LOG_ERROR, "failed to decode all channels\n");
  1817. return AVERROR_INVALIDDATA;
  1818. }
  1819. /* update codec info */
  1820. avctx->sample_rate = s->mp3decctx[0]->sample_rate;
  1821. frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));
  1822. *got_frame_ptr = 1;
  1823. return buf_size;
  1824. }
  1825. #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */