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  1. /*
  2. * AC-3 Audio Decoder
  3. * This code is developed as part of Google Summer of Code 2006 Program.
  4. *
  5. * Copyright (c) 2006 Kartikey Mahendra BHATT (bhattkm at gmail dot com).
  6. * Copyright (c) 2007 Justin Ruggles
  7. *
  8. * Portions of this code are derived from liba52
  9. * http://liba52.sourceforge.net
  10. * Copyright (C) 2000-2003 Michel Lespinasse <walken@zoy.org>
  11. * Copyright (C) 1999-2000 Aaron Holtzman <aholtzma@ess.engr.uvic.ca>
  12. *
  13. * This file is part of FFmpeg.
  14. *
  15. * FFmpeg is free software; you can redistribute it and/or
  16. * modify it under the terms of the GNU General Public
  17. * License as published by the Free Software Foundation; either
  18. * version 2 of the License, or (at your option) any later version.
  19. *
  20. * FFmpeg is distributed in the hope that it will be useful,
  21. * but WITHOUT ANY WARRANTY; without even the implied warranty of
  22. * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
  23. * General Public License for more details.
  24. *
  25. * You should have received a copy of the GNU General Public
  26. * License along with FFmpeg; if not, write to the Free Software
  27. * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  28. */
  29. #include <stdio.h>
  30. #include <stddef.h>
  31. #include <math.h>
  32. #include <string.h>
  33. #include "avcodec.h"
  34. #include "ac3_parser.h"
  35. #include "bitstream.h"
  36. #include "crc.h"
  37. #include "dsputil.h"
  38. #include "random.h"
  39. /**
  40. * Table of bin locations for rematrixing bands
  41. * reference: Section 7.5.2 Rematrixing : Frequency Band Definitions
  42. */
  43. static const uint8_t rematrix_band_tab[5] = { 13, 25, 37, 61, 253 };
  44. /**
  45. * table for exponent to scale_factor mapping
  46. * scale_factors[i] = 2 ^ -i
  47. */
  48. static float scale_factors[25];
  49. /** table for grouping exponents */
  50. static uint8_t exp_ungroup_tab[128][3];
  51. /** tables for ungrouping mantissas */
  52. static float b1_mantissas[32][3];
  53. static float b2_mantissas[128][3];
  54. static float b3_mantissas[8];
  55. static float b4_mantissas[128][2];
  56. static float b5_mantissas[16];
  57. /**
  58. * Quantization table: levels for symmetric. bits for asymmetric.
  59. * reference: Table 7.18 Mapping of bap to Quantizer
  60. */
  61. static const uint8_t quantization_tab[16] = {
  62. 0, 3, 5, 7, 11, 15,
  63. 5, 6, 7, 8, 9, 10, 11, 12, 14, 16
  64. };
  65. /** dynamic range table. converts codes to scale factors. */
  66. static float dynamic_range_tab[256];
  67. /** Adjustments in dB gain */
  68. #define LEVEL_MINUS_3DB 0.7071067811865476
  69. #define LEVEL_MINUS_4POINT5DB 0.5946035575013605
  70. #define LEVEL_MINUS_6DB 0.5000000000000000
  71. #define LEVEL_MINUS_9DB 0.3535533905932738
  72. #define LEVEL_ZERO 0.0000000000000000
  73. #define LEVEL_ONE 1.0000000000000000
  74. static const float gain_levels[6] = {
  75. LEVEL_ZERO,
  76. LEVEL_ONE,
  77. LEVEL_MINUS_3DB,
  78. LEVEL_MINUS_4POINT5DB,
  79. LEVEL_MINUS_6DB,
  80. LEVEL_MINUS_9DB
  81. };
  82. /**
  83. * Table for center mix levels
  84. * reference: Section 5.4.2.4 cmixlev
  85. */
  86. static const uint8_t center_levels[4] = { 2, 3, 4, 3 };
  87. /**
  88. * Table for surround mix levels
  89. * reference: Section 5.4.2.5 surmixlev
  90. */
  91. static const uint8_t surround_levels[4] = { 2, 4, 0, 4 };
  92. /**
  93. * Table for default stereo downmixing coefficients
  94. * reference: Section 7.8.2 Downmixing Into Two Channels
  95. */
  96. static const uint8_t ac3_default_coeffs[8][5][2] = {
  97. { { 1, 0 }, { 0, 1 }, },
  98. { { 2, 2 }, },
  99. { { 1, 0 }, { 0, 1 }, },
  100. { { 1, 0 }, { 3, 3 }, { 0, 1 }, },
  101. { { 1, 0 }, { 0, 1 }, { 4, 4 }, },
  102. { { 1, 0 }, { 3, 3 }, { 0, 1 }, { 5, 5 }, },
  103. { { 1, 0 }, { 0, 1 }, { 4, 0 }, { 0, 4 }, },
  104. { { 1, 0 }, { 3, 3 }, { 0, 1 }, { 4, 0 }, { 0, 4 }, },
  105. };
  106. /* override ac3.h to include coupling channel */
  107. #undef AC3_MAX_CHANNELS
  108. #define AC3_MAX_CHANNELS 7
  109. #define CPL_CH 0
  110. #define AC3_OUTPUT_LFEON 8
  111. typedef struct {
  112. int channel_mode; ///< channel mode (acmod)
  113. int block_switch[AC3_MAX_CHANNELS]; ///< block switch flags
  114. int dither_flag[AC3_MAX_CHANNELS]; ///< dither flags
  115. int dither_all; ///< true if all channels are dithered
  116. int cpl_in_use; ///< coupling in use
  117. int channel_in_cpl[AC3_MAX_CHANNELS]; ///< channel in coupling
  118. int phase_flags_in_use; ///< phase flags in use
  119. int phase_flags[18]; ///< phase flags
  120. int cpl_band_struct[18]; ///< coupling band structure
  121. int num_rematrixing_bands; ///< number of rematrixing bands
  122. int rematrixing_flags[4]; ///< rematrixing flags
  123. int exp_strategy[AC3_MAX_CHANNELS]; ///< exponent strategies
  124. int snr_offset[AC3_MAX_CHANNELS]; ///< signal-to-noise ratio offsets
  125. int fast_gain[AC3_MAX_CHANNELS]; ///< fast gain values (signal-to-mask ratio)
  126. int dba_mode[AC3_MAX_CHANNELS]; ///< delta bit allocation mode
  127. int dba_nsegs[AC3_MAX_CHANNELS]; ///< number of delta segments
  128. uint8_t dba_offsets[AC3_MAX_CHANNELS][8]; ///< delta segment offsets
  129. uint8_t dba_lengths[AC3_MAX_CHANNELS][8]; ///< delta segment lengths
  130. uint8_t dba_values[AC3_MAX_CHANNELS][8]; ///< delta values for each segment
  131. int sample_rate; ///< sample frequency, in Hz
  132. int bit_rate; ///< stream bit rate, in bits-per-second
  133. int frame_size; ///< current frame size, in bytes
  134. int channels; ///< number of total channels
  135. int fbw_channels; ///< number of full-bandwidth channels
  136. int lfe_on; ///< lfe channel in use
  137. int lfe_ch; ///< index of LFE channel
  138. int output_mode; ///< output channel configuration
  139. int out_channels; ///< number of output channels
  140. int center_mix_level; ///< Center mix level index
  141. int surround_mix_level; ///< Surround mix level index
  142. float downmix_coeffs[AC3_MAX_CHANNELS][2]; ///< stereo downmix coefficients
  143. float dynamic_range[2]; ///< dynamic range
  144. float cpl_coords[AC3_MAX_CHANNELS][18]; ///< coupling coordinates
  145. int num_cpl_bands; ///< number of coupling bands
  146. int num_cpl_subbands; ///< number of coupling sub bands
  147. int start_freq[AC3_MAX_CHANNELS]; ///< start frequency bin
  148. int end_freq[AC3_MAX_CHANNELS]; ///< end frequency bin
  149. AC3BitAllocParameters bit_alloc_params; ///< bit allocation parameters
  150. int8_t dexps[AC3_MAX_CHANNELS][256]; ///< decoded exponents
  151. uint8_t bap[AC3_MAX_CHANNELS][256]; ///< bit allocation pointers
  152. int16_t psd[AC3_MAX_CHANNELS][256]; ///< scaled exponents
  153. int16_t band_psd[AC3_MAX_CHANNELS][50]; ///< interpolated exponents
  154. int16_t mask[AC3_MAX_CHANNELS][50]; ///< masking curve values
  155. DECLARE_ALIGNED_16(float, transform_coeffs[AC3_MAX_CHANNELS][256]); ///< transform coefficients
  156. /* For IMDCT. */
  157. MDCTContext imdct_512; ///< for 512 sample IMDCT
  158. MDCTContext imdct_256; ///< for 256 sample IMDCT
  159. DSPContext dsp; ///< for optimization
  160. float add_bias; ///< offset for float_to_int16 conversion
  161. float mul_bias; ///< scaling for float_to_int16 conversion
  162. DECLARE_ALIGNED_16(float, output[AC3_MAX_CHANNELS-1][256]); ///< output after imdct transform and windowing
  163. DECLARE_ALIGNED_16(short, int_output[AC3_MAX_CHANNELS-1][256]); ///< final 16-bit integer output
  164. DECLARE_ALIGNED_16(float, delay[AC3_MAX_CHANNELS-1][256]); ///< delay - added to the next block
  165. DECLARE_ALIGNED_16(float, tmp_imdct[256]); ///< temporary storage for imdct transform
  166. DECLARE_ALIGNED_16(float, tmp_output[512]); ///< temporary storage for output before windowing
  167. DECLARE_ALIGNED_16(float, window[256]); ///< window coefficients
  168. /* Miscellaneous. */
  169. GetBitContext gbc; ///< bitstream reader
  170. AVRandomState dith_state; ///< for dither generation
  171. AVCodecContext *avctx; ///< parent context
  172. } AC3DecodeContext;
  173. /**
  174. * Generate a Kaiser-Bessel Derived Window.
  175. */
  176. static void ac3_window_init(float *window)
  177. {
  178. int i, j;
  179. double sum = 0.0, bessel, tmp;
  180. double local_window[256];
  181. double alpha2 = (5.0 * M_PI / 256.0) * (5.0 * M_PI / 256.0);
  182. for (i = 0; i < 256; i++) {
  183. tmp = i * (256 - i) * alpha2;
  184. bessel = 1.0;
  185. for (j = 100; j > 0; j--) /* default to 100 iterations */
  186. bessel = bessel * tmp / (j * j) + 1;
  187. sum += bessel;
  188. local_window[i] = sum;
  189. }
  190. sum++;
  191. for (i = 0; i < 256; i++)
  192. window[i] = sqrt(local_window[i] / sum);
  193. }
  194. /**
  195. * Symmetrical Dequantization
  196. * reference: Section 7.3.3 Expansion of Mantissas for Symmetrical Quantization
  197. * Tables 7.19 to 7.23
  198. */
  199. static inline float
  200. symmetric_dequant(int code, int levels)
  201. {
  202. return (code - (levels >> 1)) * (2.0f / levels);
  203. }
  204. /*
  205. * Initialize tables at runtime.
  206. */
  207. static void ac3_tables_init(void)
  208. {
  209. int i;
  210. /* generate grouped mantissa tables
  211. reference: Section 7.3.5 Ungrouping of Mantissas */
  212. for(i=0; i<32; i++) {
  213. /* bap=1 mantissas */
  214. b1_mantissas[i][0] = symmetric_dequant( i / 9 , 3);
  215. b1_mantissas[i][1] = symmetric_dequant((i % 9) / 3, 3);
  216. b1_mantissas[i][2] = symmetric_dequant((i % 9) % 3, 3);
  217. }
  218. for(i=0; i<128; i++) {
  219. /* bap=2 mantissas */
  220. b2_mantissas[i][0] = symmetric_dequant( i / 25 , 5);
  221. b2_mantissas[i][1] = symmetric_dequant((i % 25) / 5, 5);
  222. b2_mantissas[i][2] = symmetric_dequant((i % 25) % 5, 5);
  223. /* bap=4 mantissas */
  224. b4_mantissas[i][0] = symmetric_dequant(i / 11, 11);
  225. b4_mantissas[i][1] = symmetric_dequant(i % 11, 11);
  226. }
  227. /* generate ungrouped mantissa tables
  228. reference: Tables 7.21 and 7.23 */
  229. for(i=0; i<7; i++) {
  230. /* bap=3 mantissas */
  231. b3_mantissas[i] = symmetric_dequant(i, 7);
  232. }
  233. for(i=0; i<15; i++) {
  234. /* bap=5 mantissas */
  235. b5_mantissas[i] = symmetric_dequant(i, 15);
  236. }
  237. /* generate dynamic range table
  238. reference: Section 7.7.1 Dynamic Range Control */
  239. for(i=0; i<256; i++) {
  240. int v = (i >> 5) - ((i >> 7) << 3) - 5;
  241. dynamic_range_tab[i] = powf(2.0f, v) * ((i & 0x1F) | 0x20);
  242. }
  243. /* generate scale factors for exponents and asymmetrical dequantization
  244. reference: Section 7.3.2 Expansion of Mantissas for Asymmetric Quantization */
  245. for (i = 0; i < 25; i++)
  246. scale_factors[i] = pow(2.0, -i);
  247. /* generate exponent tables
  248. reference: Section 7.1.3 Exponent Decoding */
  249. for(i=0; i<128; i++) {
  250. exp_ungroup_tab[i][0] = i / 25;
  251. exp_ungroup_tab[i][1] = (i % 25) / 5;
  252. exp_ungroup_tab[i][2] = (i % 25) % 5;
  253. }
  254. }
  255. /**
  256. * AVCodec initialization
  257. */
  258. static int ac3_decode_init(AVCodecContext *avctx)
  259. {
  260. AC3DecodeContext *s = avctx->priv_data;
  261. s->avctx = avctx;
  262. ac3_common_init();
  263. ac3_tables_init();
  264. ff_mdct_init(&s->imdct_256, 8, 1);
  265. ff_mdct_init(&s->imdct_512, 9, 1);
  266. ac3_window_init(s->window);
  267. dsputil_init(&s->dsp, avctx);
  268. av_init_random(0, &s->dith_state);
  269. /* set bias values for float to int16 conversion */
  270. if(s->dsp.float_to_int16 == ff_float_to_int16_c) {
  271. s->add_bias = 385.0f;
  272. s->mul_bias = 1.0f;
  273. } else {
  274. s->add_bias = 0.0f;
  275. s->mul_bias = 32767.0f;
  276. }
  277. /* allow downmixing to stereo or mono */
  278. if (avctx->channels > 0 && avctx->request_channels > 0 &&
  279. avctx->request_channels < avctx->channels &&
  280. avctx->request_channels <= 2) {
  281. avctx->channels = avctx->request_channels;
  282. }
  283. return 0;
  284. }
  285. /**
  286. * Parse the 'sync info' and 'bit stream info' from the AC-3 bitstream.
  287. * GetBitContext within AC3DecodeContext must point to
  288. * start of the synchronized ac3 bitstream.
  289. */
  290. static int ac3_parse_header(AC3DecodeContext *s)
  291. {
  292. AC3HeaderInfo hdr;
  293. GetBitContext *gbc = &s->gbc;
  294. int err, i;
  295. err = ff_ac3_parse_header(gbc->buffer, &hdr);
  296. if(err)
  297. return err;
  298. if(hdr.bitstream_id > 10)
  299. return AC3_PARSE_ERROR_BSID;
  300. /* get decoding parameters from header info */
  301. s->bit_alloc_params.sr_code = hdr.sr_code;
  302. s->channel_mode = hdr.channel_mode;
  303. s->lfe_on = hdr.lfe_on;
  304. s->bit_alloc_params.sr_shift = hdr.sr_shift;
  305. s->sample_rate = hdr.sample_rate;
  306. s->bit_rate = hdr.bit_rate;
  307. s->channels = hdr.channels;
  308. s->fbw_channels = s->channels - s->lfe_on;
  309. s->lfe_ch = s->fbw_channels + 1;
  310. s->frame_size = hdr.frame_size;
  311. /* set default output to all source channels */
  312. s->out_channels = s->channels;
  313. s->output_mode = s->channel_mode;
  314. if(s->lfe_on)
  315. s->output_mode |= AC3_OUTPUT_LFEON;
  316. /* set default mix levels */
  317. s->center_mix_level = 3; // -4.5dB
  318. s->surround_mix_level = 4; // -6.0dB
  319. /* skip over portion of header which has already been read */
  320. skip_bits(gbc, 16); // skip the sync_word
  321. skip_bits(gbc, 16); // skip crc1
  322. skip_bits(gbc, 8); // skip fscod and frmsizecod
  323. skip_bits(gbc, 11); // skip bsid, bsmod, and acmod
  324. if(s->channel_mode == AC3_CHMODE_STEREO) {
  325. skip_bits(gbc, 2); // skip dsurmod
  326. } else {
  327. if((s->channel_mode & 1) && s->channel_mode != AC3_CHMODE_MONO)
  328. s->center_mix_level = center_levels[get_bits(gbc, 2)];
  329. if(s->channel_mode & 4)
  330. s->surround_mix_level = surround_levels[get_bits(gbc, 2)];
  331. }
  332. skip_bits1(gbc); // skip lfeon
  333. /* read the rest of the bsi. read twice for dual mono mode. */
  334. i = !(s->channel_mode);
  335. do {
  336. skip_bits(gbc, 5); // skip dialog normalization
  337. if (get_bits1(gbc))
  338. skip_bits(gbc, 8); //skip compression
  339. if (get_bits1(gbc))
  340. skip_bits(gbc, 8); //skip language code
  341. if (get_bits1(gbc))
  342. skip_bits(gbc, 7); //skip audio production information
  343. } while (i--);
  344. skip_bits(gbc, 2); //skip copyright bit and original bitstream bit
  345. /* skip the timecodes (or extra bitstream information for Alternate Syntax)
  346. TODO: read & use the xbsi1 downmix levels */
  347. if (get_bits1(gbc))
  348. skip_bits(gbc, 14); //skip timecode1 / xbsi1
  349. if (get_bits1(gbc))
  350. skip_bits(gbc, 14); //skip timecode2 / xbsi2
  351. /* skip additional bitstream info */
  352. if (get_bits1(gbc)) {
  353. i = get_bits(gbc, 6);
  354. do {
  355. skip_bits(gbc, 8);
  356. } while(i--);
  357. }
  358. return 0;
  359. }
  360. /**
  361. * Set stereo downmixing coefficients based on frame header info.
  362. * reference: Section 7.8.2 Downmixing Into Two Channels
  363. */
  364. static void set_downmix_coeffs(AC3DecodeContext *s)
  365. {
  366. int i;
  367. float cmix = gain_levels[s->center_mix_level];
  368. float smix = gain_levels[s->surround_mix_level];
  369. for(i=0; i<s->fbw_channels; i++) {
  370. s->downmix_coeffs[i][0] = gain_levels[ac3_default_coeffs[s->channel_mode][i][0]];
  371. s->downmix_coeffs[i][1] = gain_levels[ac3_default_coeffs[s->channel_mode][i][1]];
  372. }
  373. if(s->channel_mode > 1 && s->channel_mode & 1) {
  374. s->downmix_coeffs[1][0] = s->downmix_coeffs[1][1] = cmix;
  375. }
  376. if(s->channel_mode == AC3_CHMODE_2F1R || s->channel_mode == AC3_CHMODE_3F1R) {
  377. int nf = s->channel_mode - 2;
  378. s->downmix_coeffs[nf][0] = s->downmix_coeffs[nf][1] = smix * LEVEL_MINUS_3DB;
  379. }
  380. if(s->channel_mode == AC3_CHMODE_2F2R || s->channel_mode == AC3_CHMODE_3F2R) {
  381. int nf = s->channel_mode - 4;
  382. s->downmix_coeffs[nf][0] = s->downmix_coeffs[nf+1][1] = smix;
  383. }
  384. }
  385. /**
  386. * Decode the grouped exponents according to exponent strategy.
  387. * reference: Section 7.1.3 Exponent Decoding
  388. */
  389. static void decode_exponents(GetBitContext *gbc, int exp_strategy, int ngrps,
  390. uint8_t absexp, int8_t *dexps)
  391. {
  392. int i, j, grp, group_size;
  393. int dexp[256];
  394. int expacc, prevexp;
  395. /* unpack groups */
  396. group_size = exp_strategy + (exp_strategy == EXP_D45);
  397. for(grp=0,i=0; grp<ngrps; grp++) {
  398. expacc = get_bits(gbc, 7);
  399. dexp[i++] = exp_ungroup_tab[expacc][0];
  400. dexp[i++] = exp_ungroup_tab[expacc][1];
  401. dexp[i++] = exp_ungroup_tab[expacc][2];
  402. }
  403. /* convert to absolute exps and expand groups */
  404. prevexp = absexp;
  405. for(i=0; i<ngrps*3; i++) {
  406. prevexp = av_clip(prevexp + dexp[i]-2, 0, 24);
  407. for(j=0; j<group_size; j++) {
  408. dexps[(i*group_size)+j] = prevexp;
  409. }
  410. }
  411. }
  412. /**
  413. * Generate transform coefficients for each coupled channel in the coupling
  414. * range using the coupling coefficients and coupling coordinates.
  415. * reference: Section 7.4.3 Coupling Coordinate Format
  416. */
  417. static void uncouple_channels(AC3DecodeContext *s)
  418. {
  419. int i, j, ch, bnd, subbnd;
  420. subbnd = -1;
  421. i = s->start_freq[CPL_CH];
  422. for(bnd=0; bnd<s->num_cpl_bands; bnd++) {
  423. do {
  424. subbnd++;
  425. for(j=0; j<12; j++) {
  426. for(ch=1; ch<=s->fbw_channels; ch++) {
  427. if(s->channel_in_cpl[ch]) {
  428. s->transform_coeffs[ch][i] = s->transform_coeffs[CPL_CH][i] * s->cpl_coords[ch][bnd] * 8.0f;
  429. if (ch == 2 && s->phase_flags[bnd])
  430. s->transform_coeffs[ch][i] = -s->transform_coeffs[ch][i];
  431. }
  432. }
  433. i++;
  434. }
  435. } while(s->cpl_band_struct[subbnd]);
  436. }
  437. }
  438. /**
  439. * Grouped mantissas for 3-level 5-level and 11-level quantization
  440. */
  441. typedef struct {
  442. float b1_mant[3];
  443. float b2_mant[3];
  444. float b4_mant[2];
  445. int b1ptr;
  446. int b2ptr;
  447. int b4ptr;
  448. } mant_groups;
  449. /**
  450. * Get the transform coefficients for a particular channel
  451. * reference: Section 7.3 Quantization and Decoding of Mantissas
  452. */
  453. static int get_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, mant_groups *m)
  454. {
  455. GetBitContext *gbc = &s->gbc;
  456. int i, gcode, tbap, start, end;
  457. uint8_t *exps;
  458. uint8_t *bap;
  459. float *coeffs;
  460. exps = s->dexps[ch_index];
  461. bap = s->bap[ch_index];
  462. coeffs = s->transform_coeffs[ch_index];
  463. start = s->start_freq[ch_index];
  464. end = s->end_freq[ch_index];
  465. for (i = start; i < end; i++) {
  466. tbap = bap[i];
  467. switch (tbap) {
  468. case 0:
  469. coeffs[i] = ((av_random(&s->dith_state) & 0xFFFF) / 65535.0f) - 0.5f;
  470. break;
  471. case 1:
  472. if(m->b1ptr > 2) {
  473. gcode = get_bits(gbc, 5);
  474. m->b1_mant[0] = b1_mantissas[gcode][0];
  475. m->b1_mant[1] = b1_mantissas[gcode][1];
  476. m->b1_mant[2] = b1_mantissas[gcode][2];
  477. m->b1ptr = 0;
  478. }
  479. coeffs[i] = m->b1_mant[m->b1ptr++];
  480. break;
  481. case 2:
  482. if(m->b2ptr > 2) {
  483. gcode = get_bits(gbc, 7);
  484. m->b2_mant[0] = b2_mantissas[gcode][0];
  485. m->b2_mant[1] = b2_mantissas[gcode][1];
  486. m->b2_mant[2] = b2_mantissas[gcode][2];
  487. m->b2ptr = 0;
  488. }
  489. coeffs[i] = m->b2_mant[m->b2ptr++];
  490. break;
  491. case 3:
  492. coeffs[i] = b3_mantissas[get_bits(gbc, 3)];
  493. break;
  494. case 4:
  495. if(m->b4ptr > 1) {
  496. gcode = get_bits(gbc, 7);
  497. m->b4_mant[0] = b4_mantissas[gcode][0];
  498. m->b4_mant[1] = b4_mantissas[gcode][1];
  499. m->b4ptr = 0;
  500. }
  501. coeffs[i] = m->b4_mant[m->b4ptr++];
  502. break;
  503. case 5:
  504. coeffs[i] = b5_mantissas[get_bits(gbc, 4)];
  505. break;
  506. default:
  507. /* asymmetric dequantization */
  508. coeffs[i] = get_sbits(gbc, quantization_tab[tbap]) * scale_factors[quantization_tab[tbap]-1];
  509. break;
  510. }
  511. coeffs[i] *= scale_factors[exps[i]];
  512. }
  513. return 0;
  514. }
  515. /**
  516. * Remove random dithering from coefficients with zero-bit mantissas
  517. * reference: Section 7.3.4 Dither for Zero Bit Mantissas (bap=0)
  518. */
  519. static void remove_dithering(AC3DecodeContext *s) {
  520. int ch, i;
  521. int end=0;
  522. float *coeffs;
  523. uint8_t *bap;
  524. for(ch=1; ch<=s->fbw_channels; ch++) {
  525. if(!s->dither_flag[ch]) {
  526. coeffs = s->transform_coeffs[ch];
  527. bap = s->bap[ch];
  528. if(s->channel_in_cpl[ch])
  529. end = s->start_freq[CPL_CH];
  530. else
  531. end = s->end_freq[ch];
  532. for(i=0; i<end; i++) {
  533. if(!bap[i])
  534. coeffs[i] = 0.0f;
  535. }
  536. if(s->channel_in_cpl[ch]) {
  537. bap = s->bap[CPL_CH];
  538. for(; i<s->end_freq[CPL_CH]; i++) {
  539. if(!bap[i])
  540. coeffs[i] = 0.0f;
  541. }
  542. }
  543. }
  544. }
  545. }
  546. /**
  547. * Get the transform coefficients.
  548. */
  549. static int get_transform_coeffs(AC3DecodeContext *s)
  550. {
  551. int ch, end;
  552. int got_cplchan = 0;
  553. mant_groups m;
  554. m.b1ptr = m.b2ptr = m.b4ptr = 3;
  555. for (ch = 1; ch <= s->channels; ch++) {
  556. /* transform coefficients for full-bandwidth channel */
  557. if (get_transform_coeffs_ch(s, ch, &m))
  558. return -1;
  559. /* tranform coefficients for coupling channel come right after the
  560. coefficients for the first coupled channel*/
  561. if (s->channel_in_cpl[ch]) {
  562. if (!got_cplchan) {
  563. if (get_transform_coeffs_ch(s, CPL_CH, &m)) {
  564. av_log(s->avctx, AV_LOG_ERROR, "error in decoupling channels\n");
  565. return -1;
  566. }
  567. uncouple_channels(s);
  568. got_cplchan = 1;
  569. }
  570. end = s->end_freq[CPL_CH];
  571. } else {
  572. end = s->end_freq[ch];
  573. }
  574. do
  575. s->transform_coeffs[ch][end] = 0;
  576. while(++end < 256);
  577. }
  578. /* if any channel doesn't use dithering, zero appropriate coefficients */
  579. if(!s->dither_all)
  580. remove_dithering(s);
  581. return 0;
  582. }
  583. /**
  584. * Stereo rematrixing.
  585. * reference: Section 7.5.4 Rematrixing : Decoding Technique
  586. */
  587. static void do_rematrixing(AC3DecodeContext *s)
  588. {
  589. int bnd, i;
  590. int end, bndend;
  591. float tmp0, tmp1;
  592. end = FFMIN(s->end_freq[1], s->end_freq[2]);
  593. for(bnd=0; bnd<s->num_rematrixing_bands; bnd++) {
  594. if(s->rematrixing_flags[bnd]) {
  595. bndend = FFMIN(end, rematrix_band_tab[bnd+1]);
  596. for(i=rematrix_band_tab[bnd]; i<bndend; i++) {
  597. tmp0 = s->transform_coeffs[1][i];
  598. tmp1 = s->transform_coeffs[2][i];
  599. s->transform_coeffs[1][i] = tmp0 + tmp1;
  600. s->transform_coeffs[2][i] = tmp0 - tmp1;
  601. }
  602. }
  603. }
  604. }
  605. /**
  606. * Perform the 256-point IMDCT
  607. */
  608. static void do_imdct_256(AC3DecodeContext *s, int chindex)
  609. {
  610. int i, k;
  611. DECLARE_ALIGNED_16(float, x[128]);
  612. FFTComplex z[2][64];
  613. float *o_ptr = s->tmp_output;
  614. for(i=0; i<2; i++) {
  615. /* de-interleave coefficients */
  616. for(k=0; k<128; k++) {
  617. x[k] = s->transform_coeffs[chindex][2*k+i];
  618. }
  619. /* run standard IMDCT */
  620. s->imdct_256.fft.imdct_calc(&s->imdct_256, o_ptr, x, s->tmp_imdct);
  621. /* reverse the post-rotation & reordering from standard IMDCT */
  622. for(k=0; k<32; k++) {
  623. z[i][32+k].re = -o_ptr[128+2*k];
  624. z[i][32+k].im = -o_ptr[2*k];
  625. z[i][31-k].re = o_ptr[2*k+1];
  626. z[i][31-k].im = o_ptr[128+2*k+1];
  627. }
  628. }
  629. /* apply AC-3 post-rotation & reordering */
  630. for(k=0; k<64; k++) {
  631. o_ptr[ 2*k ] = -z[0][ k].im;
  632. o_ptr[ 2*k+1] = z[0][63-k].re;
  633. o_ptr[128+2*k ] = -z[0][ k].re;
  634. o_ptr[128+2*k+1] = z[0][63-k].im;
  635. o_ptr[256+2*k ] = -z[1][ k].re;
  636. o_ptr[256+2*k+1] = z[1][63-k].im;
  637. o_ptr[384+2*k ] = z[1][ k].im;
  638. o_ptr[384+2*k+1] = -z[1][63-k].re;
  639. }
  640. }
  641. /**
  642. * Inverse MDCT Transform.
  643. * Convert frequency domain coefficients to time-domain audio samples.
  644. * reference: Section 7.9.4 Transformation Equations
  645. */
  646. static inline void do_imdct(AC3DecodeContext *s)
  647. {
  648. int ch;
  649. int channels;
  650. /* Don't perform the IMDCT on the LFE channel unless it's used in the output */
  651. channels = s->fbw_channels;
  652. if(s->output_mode & AC3_OUTPUT_LFEON)
  653. channels++;
  654. for (ch=1; ch<=channels; ch++) {
  655. if (s->block_switch[ch]) {
  656. do_imdct_256(s, ch);
  657. } else {
  658. s->imdct_512.fft.imdct_calc(&s->imdct_512, s->tmp_output,
  659. s->transform_coeffs[ch], s->tmp_imdct);
  660. }
  661. /* For the first half of the block, apply the window, add the delay
  662. from the previous block, and send to output */
  663. s->dsp.vector_fmul_add_add(s->output[ch-1], s->tmp_output,
  664. s->window, s->delay[ch-1], 0, 256, 1);
  665. /* For the second half of the block, apply the window and store the
  666. samples to delay, to be combined with the next block */
  667. s->dsp.vector_fmul_reverse(s->delay[ch-1], s->tmp_output+256,
  668. s->window, 256);
  669. }
  670. }
  671. /**
  672. * Downmix the output to mono or stereo.
  673. */
  674. static void ac3_downmix(AC3DecodeContext *s)
  675. {
  676. int i, j;
  677. float v0, v1, s0, s1;
  678. for(i=0; i<256; i++) {
  679. v0 = v1 = s0 = s1 = 0.0f;
  680. for(j=0; j<s->fbw_channels; j++) {
  681. v0 += s->output[j][i] * s->downmix_coeffs[j][0];
  682. v1 += s->output[j][i] * s->downmix_coeffs[j][1];
  683. s0 += s->downmix_coeffs[j][0];
  684. s1 += s->downmix_coeffs[j][1];
  685. }
  686. v0 /= s0;
  687. v1 /= s1;
  688. if(s->output_mode == AC3_CHMODE_MONO) {
  689. s->output[0][i] = (v0 + v1) * LEVEL_MINUS_3DB;
  690. } else if(s->output_mode == AC3_CHMODE_STEREO) {
  691. s->output[0][i] = v0;
  692. s->output[1][i] = v1;
  693. }
  694. }
  695. }
  696. /**
  697. * Parse an audio block from AC-3 bitstream.
  698. */
  699. static int ac3_parse_audio_block(AC3DecodeContext *s, int blk)
  700. {
  701. int fbw_channels = s->fbw_channels;
  702. int channel_mode = s->channel_mode;
  703. int i, bnd, seg, ch;
  704. GetBitContext *gbc = &s->gbc;
  705. uint8_t bit_alloc_stages[AC3_MAX_CHANNELS];
  706. memset(bit_alloc_stages, 0, AC3_MAX_CHANNELS);
  707. /* block switch flags */
  708. for (ch = 1; ch <= fbw_channels; ch++)
  709. s->block_switch[ch] = get_bits1(gbc);
  710. /* dithering flags */
  711. s->dither_all = 1;
  712. for (ch = 1; ch <= fbw_channels; ch++) {
  713. s->dither_flag[ch] = get_bits1(gbc);
  714. if(!s->dither_flag[ch])
  715. s->dither_all = 0;
  716. }
  717. /* dynamic range */
  718. i = !(s->channel_mode);
  719. do {
  720. if(get_bits1(gbc)) {
  721. s->dynamic_range[i] = ((dynamic_range_tab[get_bits(gbc, 8)]-1.0) *
  722. s->avctx->drc_scale)+1.0;
  723. } else if(blk == 0) {
  724. s->dynamic_range[i] = 1.0f;
  725. }
  726. } while(i--);
  727. /* coupling strategy */
  728. if (get_bits1(gbc)) {
  729. memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS);
  730. s->cpl_in_use = get_bits1(gbc);
  731. if (s->cpl_in_use) {
  732. /* coupling in use */
  733. int cpl_begin_freq, cpl_end_freq;
  734. /* determine which channels are coupled */
  735. for (ch = 1; ch <= fbw_channels; ch++)
  736. s->channel_in_cpl[ch] = get_bits1(gbc);
  737. /* phase flags in use */
  738. if (channel_mode == AC3_CHMODE_STEREO)
  739. s->phase_flags_in_use = get_bits1(gbc);
  740. /* coupling frequency range and band structure */
  741. cpl_begin_freq = get_bits(gbc, 4);
  742. cpl_end_freq = get_bits(gbc, 4);
  743. if (3 + cpl_end_freq - cpl_begin_freq < 0) {
  744. av_log(s->avctx, AV_LOG_ERROR, "3+cplendf = %d < cplbegf = %d\n", 3+cpl_end_freq, cpl_begin_freq);
  745. return -1;
  746. }
  747. s->num_cpl_bands = s->num_cpl_subbands = 3 + cpl_end_freq - cpl_begin_freq;
  748. s->start_freq[CPL_CH] = cpl_begin_freq * 12 + 37;
  749. s->end_freq[CPL_CH] = cpl_end_freq * 12 + 73;
  750. for (bnd = 0; bnd < s->num_cpl_subbands - 1; bnd++) {
  751. if (get_bits1(gbc)) {
  752. s->cpl_band_struct[bnd] = 1;
  753. s->num_cpl_bands--;
  754. }
  755. }
  756. s->cpl_band_struct[s->num_cpl_subbands-1] = 0;
  757. } else {
  758. /* coupling not in use */
  759. for (ch = 1; ch <= fbw_channels; ch++)
  760. s->channel_in_cpl[ch] = 0;
  761. }
  762. }
  763. /* coupling coordinates */
  764. if (s->cpl_in_use) {
  765. int cpl_coords_exist = 0;
  766. for (ch = 1; ch <= fbw_channels; ch++) {
  767. if (s->channel_in_cpl[ch]) {
  768. if (get_bits1(gbc)) {
  769. int master_cpl_coord, cpl_coord_exp, cpl_coord_mant;
  770. cpl_coords_exist = 1;
  771. master_cpl_coord = 3 * get_bits(gbc, 2);
  772. for (bnd = 0; bnd < s->num_cpl_bands; bnd++) {
  773. cpl_coord_exp = get_bits(gbc, 4);
  774. cpl_coord_mant = get_bits(gbc, 4);
  775. if (cpl_coord_exp == 15)
  776. s->cpl_coords[ch][bnd] = cpl_coord_mant / 16.0f;
  777. else
  778. s->cpl_coords[ch][bnd] = (cpl_coord_mant + 16.0f) / 32.0f;
  779. s->cpl_coords[ch][bnd] *= scale_factors[cpl_coord_exp + master_cpl_coord];
  780. }
  781. }
  782. }
  783. }
  784. /* phase flags */
  785. if (channel_mode == AC3_CHMODE_STEREO && cpl_coords_exist) {
  786. for (bnd = 0; bnd < s->num_cpl_bands; bnd++) {
  787. s->phase_flags[bnd] = s->phase_flags_in_use? get_bits1(gbc) : 0;
  788. }
  789. }
  790. }
  791. /* stereo rematrixing strategy and band structure */
  792. if (channel_mode == AC3_CHMODE_STEREO) {
  793. if (get_bits1(gbc)) {
  794. s->num_rematrixing_bands = 4;
  795. if(s->cpl_in_use && s->start_freq[CPL_CH] <= 61)
  796. s->num_rematrixing_bands -= 1 + (s->start_freq[CPL_CH] == 37);
  797. for(bnd=0; bnd<s->num_rematrixing_bands; bnd++)
  798. s->rematrixing_flags[bnd] = get_bits1(gbc);
  799. }
  800. }
  801. /* exponent strategies for each channel */
  802. s->exp_strategy[CPL_CH] = EXP_REUSE;
  803. s->exp_strategy[s->lfe_ch] = EXP_REUSE;
  804. for (ch = !s->cpl_in_use; ch <= s->channels; ch++) {
  805. if(ch == s->lfe_ch)
  806. s->exp_strategy[ch] = get_bits(gbc, 1);
  807. else
  808. s->exp_strategy[ch] = get_bits(gbc, 2);
  809. if(s->exp_strategy[ch] != EXP_REUSE)
  810. bit_alloc_stages[ch] = 3;
  811. }
  812. /* channel bandwidth */
  813. for (ch = 1; ch <= fbw_channels; ch++) {
  814. s->start_freq[ch] = 0;
  815. if (s->exp_strategy[ch] != EXP_REUSE) {
  816. int prev = s->end_freq[ch];
  817. if (s->channel_in_cpl[ch])
  818. s->end_freq[ch] = s->start_freq[CPL_CH];
  819. else {
  820. int bandwidth_code = get_bits(gbc, 6);
  821. if (bandwidth_code > 60) {
  822. av_log(s->avctx, AV_LOG_ERROR, "bandwidth code = %d > 60", bandwidth_code);
  823. return -1;
  824. }
  825. s->end_freq[ch] = bandwidth_code * 3 + 73;
  826. }
  827. if(blk > 0 && s->end_freq[ch] != prev)
  828. memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS);
  829. }
  830. }
  831. s->start_freq[s->lfe_ch] = 0;
  832. s->end_freq[s->lfe_ch] = 7;
  833. /* decode exponents for each channel */
  834. for (ch = !s->cpl_in_use; ch <= s->channels; ch++) {
  835. if (s->exp_strategy[ch] != EXP_REUSE) {
  836. int group_size, num_groups;
  837. group_size = 3 << (s->exp_strategy[ch] - 1);
  838. if(ch == CPL_CH)
  839. num_groups = (s->end_freq[ch] - s->start_freq[ch]) / group_size;
  840. else if(ch == s->lfe_ch)
  841. num_groups = 2;
  842. else
  843. num_groups = (s->end_freq[ch] + group_size - 4) / group_size;
  844. s->dexps[ch][0] = get_bits(gbc, 4) << !ch;
  845. decode_exponents(gbc, s->exp_strategy[ch], num_groups, s->dexps[ch][0],
  846. &s->dexps[ch][s->start_freq[ch]+!!ch]);
  847. if(ch != CPL_CH && ch != s->lfe_ch)
  848. skip_bits(gbc, 2); /* skip gainrng */
  849. }
  850. }
  851. /* bit allocation information */
  852. if (get_bits1(gbc)) {
  853. s->bit_alloc_params.slow_decay = ff_ac3_slow_decay_tab[get_bits(gbc, 2)] >> s->bit_alloc_params.sr_shift;
  854. s->bit_alloc_params.fast_decay = ff_ac3_fast_decay_tab[get_bits(gbc, 2)] >> s->bit_alloc_params.sr_shift;
  855. s->bit_alloc_params.slow_gain = ff_ac3_slow_gain_tab[get_bits(gbc, 2)];
  856. s->bit_alloc_params.db_per_bit = ff_ac3_db_per_bit_tab[get_bits(gbc, 2)];
  857. s->bit_alloc_params.floor = ff_ac3_floor_tab[get_bits(gbc, 3)];
  858. for(ch=!s->cpl_in_use; ch<=s->channels; ch++) {
  859. bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2);
  860. }
  861. }
  862. /* signal-to-noise ratio offsets and fast gains (signal-to-mask ratios) */
  863. if (get_bits1(gbc)) {
  864. int csnr;
  865. csnr = (get_bits(gbc, 6) - 15) << 4;
  866. for (ch = !s->cpl_in_use; ch <= s->channels; ch++) { /* snr offset and fast gain */
  867. s->snr_offset[ch] = (csnr + get_bits(gbc, 4)) << 2;
  868. s->fast_gain[ch] = ff_ac3_fast_gain_tab[get_bits(gbc, 3)];
  869. }
  870. memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS);
  871. }
  872. /* coupling leak information */
  873. if (s->cpl_in_use && get_bits1(gbc)) {
  874. s->bit_alloc_params.cpl_fast_leak = get_bits(gbc, 3);
  875. s->bit_alloc_params.cpl_slow_leak = get_bits(gbc, 3);
  876. bit_alloc_stages[CPL_CH] = FFMAX(bit_alloc_stages[CPL_CH], 2);
  877. }
  878. /* delta bit allocation information */
  879. if (get_bits1(gbc)) {
  880. /* delta bit allocation exists (strategy) */
  881. for (ch = !s->cpl_in_use; ch <= fbw_channels; ch++) {
  882. s->dba_mode[ch] = get_bits(gbc, 2);
  883. if (s->dba_mode[ch] == DBA_RESERVED) {
  884. av_log(s->avctx, AV_LOG_ERROR, "delta bit allocation strategy reserved\n");
  885. return -1;
  886. }
  887. bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2);
  888. }
  889. /* channel delta offset, len and bit allocation */
  890. for (ch = !s->cpl_in_use; ch <= fbw_channels; ch++) {
  891. if (s->dba_mode[ch] == DBA_NEW) {
  892. s->dba_nsegs[ch] = get_bits(gbc, 3);
  893. for (seg = 0; seg <= s->dba_nsegs[ch]; seg++) {
  894. s->dba_offsets[ch][seg] = get_bits(gbc, 5);
  895. s->dba_lengths[ch][seg] = get_bits(gbc, 4);
  896. s->dba_values[ch][seg] = get_bits(gbc, 3);
  897. }
  898. }
  899. }
  900. } else if(blk == 0) {
  901. for(ch=0; ch<=s->channels; ch++) {
  902. s->dba_mode[ch] = DBA_NONE;
  903. }
  904. }
  905. /* Bit allocation */
  906. for(ch=!s->cpl_in_use; ch<=s->channels; ch++) {
  907. if(bit_alloc_stages[ch] > 2) {
  908. /* Exponent mapping into PSD and PSD integration */
  909. ff_ac3_bit_alloc_calc_psd(s->dexps[ch],
  910. s->start_freq[ch], s->end_freq[ch],
  911. s->psd[ch], s->band_psd[ch]);
  912. }
  913. if(bit_alloc_stages[ch] > 1) {
  914. /* Compute excitation function, Compute masking curve, and
  915. Apply delta bit allocation */
  916. ff_ac3_bit_alloc_calc_mask(&s->bit_alloc_params, s->band_psd[ch],
  917. s->start_freq[ch], s->end_freq[ch],
  918. s->fast_gain[ch], (ch == s->lfe_ch),
  919. s->dba_mode[ch], s->dba_nsegs[ch],
  920. s->dba_offsets[ch], s->dba_lengths[ch],
  921. s->dba_values[ch], s->mask[ch]);
  922. }
  923. if(bit_alloc_stages[ch] > 0) {
  924. /* Compute bit allocation */
  925. ff_ac3_bit_alloc_calc_bap(s->mask[ch], s->psd[ch],
  926. s->start_freq[ch], s->end_freq[ch],
  927. s->snr_offset[ch],
  928. s->bit_alloc_params.floor,
  929. s->bap[ch]);
  930. }
  931. }
  932. /* unused dummy data */
  933. if (get_bits1(gbc)) {
  934. int skipl = get_bits(gbc, 9);
  935. while(skipl--)
  936. skip_bits(gbc, 8);
  937. }
  938. /* unpack the transform coefficients
  939. this also uncouples channels if coupling is in use. */
  940. if (get_transform_coeffs(s)) {
  941. av_log(s->avctx, AV_LOG_ERROR, "Error in routine get_transform_coeffs\n");
  942. return -1;
  943. }
  944. /* recover coefficients if rematrixing is in use */
  945. if(s->channel_mode == AC3_CHMODE_STEREO)
  946. do_rematrixing(s);
  947. /* apply scaling to coefficients (headroom, dynrng) */
  948. for(ch=1; ch<=s->channels; ch++) {
  949. float gain = 2.0f * s->mul_bias;
  950. if(s->channel_mode == AC3_CHMODE_DUALMONO) {
  951. gain *= s->dynamic_range[ch-1];
  952. } else {
  953. gain *= s->dynamic_range[0];
  954. }
  955. for(i=0; i<s->end_freq[ch]; i++) {
  956. s->transform_coeffs[ch][i] *= gain;
  957. }
  958. }
  959. do_imdct(s);
  960. /* downmix output if needed */
  961. if(s->channels != s->out_channels && !((s->output_mode & AC3_OUTPUT_LFEON) &&
  962. s->fbw_channels == s->out_channels)) {
  963. ac3_downmix(s);
  964. }
  965. /* convert float to 16-bit integer */
  966. for(ch=0; ch<s->out_channels; ch++) {
  967. for(i=0; i<256; i++) {
  968. s->output[ch][i] += s->add_bias;
  969. }
  970. s->dsp.float_to_int16(s->int_output[ch], s->output[ch], 256);
  971. }
  972. return 0;
  973. }
  974. /**
  975. * Decode a single AC-3 frame.
  976. */
  977. static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size, uint8_t *buf, int buf_size)
  978. {
  979. AC3DecodeContext *s = avctx->priv_data;
  980. int16_t *out_samples = (int16_t *)data;
  981. int i, blk, ch, err;
  982. /* initialize the GetBitContext with the start of valid AC-3 Frame */
  983. init_get_bits(&s->gbc, buf, buf_size * 8);
  984. /* parse the syncinfo */
  985. err = ac3_parse_header(s);
  986. if(err) {
  987. switch(err) {
  988. case AC3_PARSE_ERROR_SYNC:
  989. av_log(avctx, AV_LOG_ERROR, "frame sync error\n");
  990. break;
  991. case AC3_PARSE_ERROR_BSID:
  992. av_log(avctx, AV_LOG_ERROR, "invalid bitstream id\n");
  993. break;
  994. case AC3_PARSE_ERROR_SAMPLE_RATE:
  995. av_log(avctx, AV_LOG_ERROR, "invalid sample rate\n");
  996. break;
  997. case AC3_PARSE_ERROR_FRAME_SIZE:
  998. av_log(avctx, AV_LOG_ERROR, "invalid frame size\n");
  999. break;
  1000. default:
  1001. av_log(avctx, AV_LOG_ERROR, "invalid header\n");
  1002. break;
  1003. }
  1004. return -1;
  1005. }
  1006. /* check that reported frame size fits in input buffer */
  1007. if(s->frame_size > buf_size) {
  1008. av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
  1009. return -1;
  1010. }
  1011. /* check for crc mismatch */
  1012. if(avctx->error_resilience > 0) {
  1013. if(av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0, &buf[2], s->frame_size-2)) {
  1014. av_log(avctx, AV_LOG_ERROR, "frame CRC mismatch\n");
  1015. return -1;
  1016. }
  1017. /* TODO: error concealment */
  1018. }
  1019. avctx->sample_rate = s->sample_rate;
  1020. avctx->bit_rate = s->bit_rate;
  1021. /* channel config */
  1022. s->out_channels = s->channels;
  1023. if (avctx->request_channels > 0 && avctx->request_channels <= 2 &&
  1024. avctx->request_channels < s->channels) {
  1025. s->out_channels = avctx->request_channels;
  1026. s->output_mode = avctx->request_channels == 1 ? AC3_CHMODE_MONO : AC3_CHMODE_STEREO;
  1027. }
  1028. avctx->channels = s->out_channels;
  1029. /* set downmixing coefficients if needed */
  1030. if(s->channels != s->out_channels && !((s->output_mode & AC3_OUTPUT_LFEON) &&
  1031. s->fbw_channels == s->out_channels)) {
  1032. set_downmix_coeffs(s);
  1033. }
  1034. /* parse the audio blocks */
  1035. for (blk = 0; blk < NB_BLOCKS; blk++) {
  1036. if (ac3_parse_audio_block(s, blk)) {
  1037. av_log(avctx, AV_LOG_ERROR, "error parsing the audio block\n");
  1038. *data_size = 0;
  1039. return s->frame_size;
  1040. }
  1041. for (i = 0; i < 256; i++)
  1042. for (ch = 0; ch < s->out_channels; ch++)
  1043. *(out_samples++) = s->int_output[ch][i];
  1044. }
  1045. *data_size = NB_BLOCKS * 256 * avctx->channels * sizeof (int16_t);
  1046. return s->frame_size;
  1047. }
  1048. /**
  1049. * Uninitialize the AC-3 decoder.
  1050. */
  1051. static int ac3_decode_end(AVCodecContext *avctx)
  1052. {
  1053. AC3DecodeContext *s = avctx->priv_data;
  1054. ff_mdct_end(&s->imdct_512);
  1055. ff_mdct_end(&s->imdct_256);
  1056. return 0;
  1057. }
  1058. AVCodec ac3_decoder = {
  1059. .name = "ac3",
  1060. .type = CODEC_TYPE_AUDIO,
  1061. .id = CODEC_ID_AC3,
  1062. .priv_data_size = sizeof (AC3DecodeContext),
  1063. .init = ac3_decode_init,
  1064. .close = ac3_decode_end,
  1065. .decode = ac3_decode_frame,
  1066. };