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