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

1158 lines
40KB

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