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