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  1. /*
  2. * DCA compatible decoder
  3. * Copyright (C) 2004 Gildas Bazin
  4. * Copyright (C) 2004 Benjamin Zores
  5. * Copyright (C) 2006 Benjamin Larsson
  6. * Copyright (C) 2007 Konstantin Shishkov
  7. *
  8. * This file is part of FFmpeg.
  9. *
  10. * FFmpeg is free software; you can redistribute it and/or
  11. * modify it under the terms of the GNU Lesser General Public
  12. * License as published by the Free Software Foundation; either
  13. * version 2.1 of the License, or (at your option) any later version.
  14. *
  15. * FFmpeg is distributed in the hope that it will be useful,
  16. * but WITHOUT ANY WARRANTY; without even the implied warranty of
  17. * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
  18. * Lesser General Public License for more details.
  19. *
  20. * You should have received a copy of the GNU Lesser General Public
  21. * License along with FFmpeg; if not, write to the Free Software
  22. * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  23. */
  24. #include <math.h>
  25. #include <stddef.h>
  26. #include <stdio.h>
  27. #include "libavutil/channel_layout.h"
  28. #include "libavutil/common.h"
  29. #include "libavutil/float_dsp.h"
  30. #include "libavutil/internal.h"
  31. #include "libavutil/intreadwrite.h"
  32. #include "libavutil/mathematics.h"
  33. #include "libavutil/opt.h"
  34. #include "libavutil/samplefmt.h"
  35. #include "avcodec.h"
  36. #include "dca.h"
  37. #include "dcadata.h"
  38. #include "dcadsp.h"
  39. #include "dcahuff.h"
  40. #include "dca_exss.h"
  41. #include "fft.h"
  42. #include "fmtconvert.h"
  43. #include "get_bits.h"
  44. #include "internal.h"
  45. #include "mathops.h"
  46. #include "synth_filter.h"
  47. #if ARCH_ARM
  48. # include "arm/dca.h"
  49. #endif
  50. enum DCAMode {
  51. DCA_MONO = 0,
  52. DCA_CHANNEL,
  53. DCA_STEREO,
  54. DCA_STEREO_SUMDIFF,
  55. DCA_STEREO_TOTAL,
  56. DCA_3F,
  57. DCA_2F1R,
  58. DCA_3F1R,
  59. DCA_2F2R,
  60. DCA_3F2R,
  61. DCA_4F2R
  62. };
  63. enum DCAXxchSpeakerMask {
  64. DCA_XXCH_FRONT_CENTER = 0x0000001,
  65. DCA_XXCH_FRONT_LEFT = 0x0000002,
  66. DCA_XXCH_FRONT_RIGHT = 0x0000004,
  67. DCA_XXCH_SIDE_REAR_LEFT = 0x0000008,
  68. DCA_XXCH_SIDE_REAR_RIGHT = 0x0000010,
  69. DCA_XXCH_LFE1 = 0x0000020,
  70. DCA_XXCH_REAR_CENTER = 0x0000040,
  71. DCA_XXCH_SURROUND_REAR_LEFT = 0x0000080,
  72. DCA_XXCH_SURROUND_REAR_RIGHT = 0x0000100,
  73. DCA_XXCH_SIDE_SURROUND_LEFT = 0x0000200,
  74. DCA_XXCH_SIDE_SURROUND_RIGHT = 0x0000400,
  75. DCA_XXCH_FRONT_CENTER_LEFT = 0x0000800,
  76. DCA_XXCH_FRONT_CENTER_RIGHT = 0x0001000,
  77. DCA_XXCH_FRONT_HIGH_LEFT = 0x0002000,
  78. DCA_XXCH_FRONT_HIGH_CENTER = 0x0004000,
  79. DCA_XXCH_FRONT_HIGH_RIGHT = 0x0008000,
  80. DCA_XXCH_LFE2 = 0x0010000,
  81. DCA_XXCH_SIDE_FRONT_LEFT = 0x0020000,
  82. DCA_XXCH_SIDE_FRONT_RIGHT = 0x0040000,
  83. DCA_XXCH_OVERHEAD = 0x0080000,
  84. DCA_XXCH_SIDE_HIGH_LEFT = 0x0100000,
  85. DCA_XXCH_SIDE_HIGH_RIGHT = 0x0200000,
  86. DCA_XXCH_REAR_HIGH_CENTER = 0x0400000,
  87. DCA_XXCH_REAR_HIGH_LEFT = 0x0800000,
  88. DCA_XXCH_REAR_HIGH_RIGHT = 0x1000000,
  89. DCA_XXCH_REAR_LOW_CENTER = 0x2000000,
  90. DCA_XXCH_REAR_LOW_LEFT = 0x4000000,
  91. DCA_XXCH_REAR_LOW_RIGHT = 0x8000000,
  92. };
  93. static const uint32_t map_xxch_to_native[28] = {
  94. AV_CH_FRONT_CENTER,
  95. AV_CH_FRONT_LEFT,
  96. AV_CH_FRONT_RIGHT,
  97. AV_CH_SIDE_LEFT,
  98. AV_CH_SIDE_RIGHT,
  99. AV_CH_LOW_FREQUENCY,
  100. AV_CH_BACK_CENTER,
  101. AV_CH_BACK_LEFT,
  102. AV_CH_BACK_RIGHT,
  103. AV_CH_SIDE_LEFT, /* side surround left -- dup sur side L */
  104. AV_CH_SIDE_RIGHT, /* side surround right -- dup sur side R */
  105. AV_CH_FRONT_LEFT_OF_CENTER,
  106. AV_CH_FRONT_RIGHT_OF_CENTER,
  107. AV_CH_TOP_FRONT_LEFT,
  108. AV_CH_TOP_FRONT_CENTER,
  109. AV_CH_TOP_FRONT_RIGHT,
  110. AV_CH_LOW_FREQUENCY, /* lfe2 -- duplicate lfe1 position */
  111. AV_CH_FRONT_LEFT_OF_CENTER, /* side front left -- dup front cntr L */
  112. AV_CH_FRONT_RIGHT_OF_CENTER,/* side front right -- dup front cntr R */
  113. AV_CH_TOP_CENTER, /* overhead */
  114. AV_CH_TOP_FRONT_LEFT, /* side high left -- dup */
  115. AV_CH_TOP_FRONT_RIGHT, /* side high right -- dup */
  116. AV_CH_TOP_BACK_CENTER,
  117. AV_CH_TOP_BACK_LEFT,
  118. AV_CH_TOP_BACK_RIGHT,
  119. AV_CH_BACK_CENTER, /* rear low center -- dup */
  120. AV_CH_BACK_LEFT, /* rear low left -- dup */
  121. AV_CH_BACK_RIGHT /* read low right -- dup */
  122. };
  123. /* -1 are reserved or unknown */
  124. static const int dca_ext_audio_descr_mask[] = {
  125. DCA_EXT_XCH,
  126. -1,
  127. DCA_EXT_X96,
  128. DCA_EXT_XCH | DCA_EXT_X96,
  129. -1,
  130. -1,
  131. DCA_EXT_XXCH,
  132. -1,
  133. };
  134. /* Tables for mapping dts channel configurations to libavcodec multichannel api.
  135. * Some compromises have been made for special configurations. Most configurations
  136. * are never used so complete accuracy is not needed.
  137. *
  138. * L = left, R = right, C = center, S = surround, F = front, R = rear, T = total, OV = overhead.
  139. * S -> side, when both rear and back are configured move one of them to the side channel
  140. * OV -> center back
  141. * All 2 channel configurations -> AV_CH_LAYOUT_STEREO
  142. */
  143. static const uint64_t dca_core_channel_layout[] = {
  144. AV_CH_FRONT_CENTER, ///< 1, A
  145. AV_CH_LAYOUT_STEREO, ///< 2, A + B (dual mono)
  146. AV_CH_LAYOUT_STEREO, ///< 2, L + R (stereo)
  147. AV_CH_LAYOUT_STEREO, ///< 2, (L + R) + (L - R) (sum-difference)
  148. AV_CH_LAYOUT_STEREO, ///< 2, LT + RT (left and right total)
  149. AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER, ///< 3, C + L + R
  150. AV_CH_LAYOUT_STEREO | AV_CH_BACK_CENTER, ///< 3, L + R + S
  151. AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER, ///< 4, C + L + R + S
  152. AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT, ///< 4, L + R + SL + SR
  153. AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_SIDE_LEFT |
  154. AV_CH_SIDE_RIGHT, ///< 5, C + L + R + SL + SR
  155. AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |
  156. AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER, ///< 6, CL + CR + L + R + SL + SR
  157. AV_CH_LAYOUT_STEREO | AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT |
  158. AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER, ///< 6, C + L + R + LR + RR + OV
  159. AV_CH_FRONT_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |
  160. AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_BACK_CENTER |
  161. AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT, ///< 6, CF + CR + LF + RF + LR + RR
  162. AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER |
  163. AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |
  164. AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT, ///< 7, CL + C + CR + L + R + SL + SR
  165. AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |
  166. AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |
  167. AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT, ///< 8, CL + CR + L + R + SL1 + SL2 + SR1 + SR2
  168. AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER |
  169. AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |
  170. AV_CH_SIDE_LEFT | AV_CH_BACK_CENTER | AV_CH_SIDE_RIGHT, ///< 8, CL + C + CR + L + R + SL + S + SR
  171. };
  172. static const int8_t dca_lfe_index[] = {
  173. 1, 2, 2, 2, 2, 3, 2, 3, 2, 3, 2, 3, 1, 3, 2, 3
  174. };
  175. static const int8_t dca_channel_reorder_lfe[][9] = {
  176. { 0, -1, -1, -1, -1, -1, -1, -1, -1 },
  177. { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
  178. { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
  179. { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
  180. { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
  181. { 2, 0, 1, -1, -1, -1, -1, -1, -1 },
  182. { 0, 1, 3, -1, -1, -1, -1, -1, -1 },
  183. { 2, 0, 1, 4, -1, -1, -1, -1, -1 },
  184. { 0, 1, 3, 4, -1, -1, -1, -1, -1 },
  185. { 2, 0, 1, 4, 5, -1, -1, -1, -1 },
  186. { 3, 4, 0, 1, 5, 6, -1, -1, -1 },
  187. { 2, 0, 1, 4, 5, 6, -1, -1, -1 },
  188. { 0, 6, 4, 5, 2, 3, -1, -1, -1 },
  189. { 4, 2, 5, 0, 1, 6, 7, -1, -1 },
  190. { 5, 6, 0, 1, 7, 3, 8, 4, -1 },
  191. { 4, 2, 5, 0, 1, 6, 8, 7, -1 },
  192. };
  193. static const int8_t dca_channel_reorder_lfe_xch[][9] = {
  194. { 0, 2, -1, -1, -1, -1, -1, -1, -1 },
  195. { 0, 1, 3, -1, -1, -1, -1, -1, -1 },
  196. { 0, 1, 3, -1, -1, -1, -1, -1, -1 },
  197. { 0, 1, 3, -1, -1, -1, -1, -1, -1 },
  198. { 0, 1, 3, -1, -1, -1, -1, -1, -1 },
  199. { 2, 0, 1, 4, -1, -1, -1, -1, -1 },
  200. { 0, 1, 3, 4, -1, -1, -1, -1, -1 },
  201. { 2, 0, 1, 4, 5, -1, -1, -1, -1 },
  202. { 0, 1, 4, 5, 3, -1, -1, -1, -1 },
  203. { 2, 0, 1, 5, 6, 4, -1, -1, -1 },
  204. { 3, 4, 0, 1, 6, 7, 5, -1, -1 },
  205. { 2, 0, 1, 4, 5, 6, 7, -1, -1 },
  206. { 0, 6, 4, 5, 2, 3, 7, -1, -1 },
  207. { 4, 2, 5, 0, 1, 7, 8, 6, -1 },
  208. { 5, 6, 0, 1, 8, 3, 9, 4, 7 },
  209. { 4, 2, 5, 0, 1, 6, 9, 8, 7 },
  210. };
  211. static const int8_t dca_channel_reorder_nolfe[][9] = {
  212. { 0, -1, -1, -1, -1, -1, -1, -1, -1 },
  213. { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
  214. { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
  215. { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
  216. { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
  217. { 2, 0, 1, -1, -1, -1, -1, -1, -1 },
  218. { 0, 1, 2, -1, -1, -1, -1, -1, -1 },
  219. { 2, 0, 1, 3, -1, -1, -1, -1, -1 },
  220. { 0, 1, 2, 3, -1, -1, -1, -1, -1 },
  221. { 2, 0, 1, 3, 4, -1, -1, -1, -1 },
  222. { 2, 3, 0, 1, 4, 5, -1, -1, -1 },
  223. { 2, 0, 1, 3, 4, 5, -1, -1, -1 },
  224. { 0, 5, 3, 4, 1, 2, -1, -1, -1 },
  225. { 3, 2, 4, 0, 1, 5, 6, -1, -1 },
  226. { 4, 5, 0, 1, 6, 2, 7, 3, -1 },
  227. { 3, 2, 4, 0, 1, 5, 7, 6, -1 },
  228. };
  229. static const int8_t dca_channel_reorder_nolfe_xch[][9] = {
  230. { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
  231. { 0, 1, 2, -1, -1, -1, -1, -1, -1 },
  232. { 0, 1, 2, -1, -1, -1, -1, -1, -1 },
  233. { 0, 1, 2, -1, -1, -1, -1, -1, -1 },
  234. { 0, 1, 2, -1, -1, -1, -1, -1, -1 },
  235. { 2, 0, 1, 3, -1, -1, -1, -1, -1 },
  236. { 0, 1, 2, 3, -1, -1, -1, -1, -1 },
  237. { 2, 0, 1, 3, 4, -1, -1, -1, -1 },
  238. { 0, 1, 3, 4, 2, -1, -1, -1, -1 },
  239. { 2, 0, 1, 4, 5, 3, -1, -1, -1 },
  240. { 2, 3, 0, 1, 5, 6, 4, -1, -1 },
  241. { 2, 0, 1, 3, 4, 5, 6, -1, -1 },
  242. { 0, 5, 3, 4, 1, 2, 6, -1, -1 },
  243. { 3, 2, 4, 0, 1, 6, 7, 5, -1 },
  244. { 4, 5, 0, 1, 7, 2, 8, 3, 6 },
  245. { 3, 2, 4, 0, 1, 5, 8, 7, 6 },
  246. };
  247. #define DCA_DOLBY 101 /* FIXME */
  248. #define DCA_CHANNEL_BITS 6
  249. #define DCA_CHANNEL_MASK 0x3F
  250. #define DCA_LFE 0x80
  251. #define HEADER_SIZE 14
  252. #define DCA_NSYNCAUX 0x9A1105A0
  253. /** Bit allocation */
  254. typedef struct BitAlloc {
  255. int offset; ///< code values offset
  256. int maxbits[8]; ///< max bits in VLC
  257. int wrap; ///< wrap for get_vlc2()
  258. VLC vlc[8]; ///< actual codes
  259. } BitAlloc;
  260. static BitAlloc dca_bitalloc_index; ///< indexes for samples VLC select
  261. static BitAlloc dca_tmode; ///< transition mode VLCs
  262. static BitAlloc dca_scalefactor; ///< scalefactor VLCs
  263. static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs
  264. static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba,
  265. int idx)
  266. {
  267. return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) +
  268. ba->offset;
  269. }
  270. static float dca_dmix_code(unsigned code);
  271. static const uint16_t dca_vlc_offs[] = {
  272. 0, 512, 640, 768, 1282, 1794, 2436, 3080, 3770, 4454, 5364,
  273. 5372, 5380, 5388, 5392, 5396, 5412, 5420, 5428, 5460, 5492, 5508,
  274. 5572, 5604, 5668, 5796, 5860, 5892, 6412, 6668, 6796, 7308, 7564,
  275. 7820, 8076, 8620, 9132, 9388, 9910, 10166, 10680, 11196, 11726, 12240,
  276. 12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264,
  277. 18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622,
  278. };
  279. static av_cold void dca_init_vlcs(void)
  280. {
  281. static int vlcs_initialized = 0;
  282. int i, j, c = 14;
  283. static VLC_TYPE dca_table[23622][2];
  284. if (vlcs_initialized)
  285. return;
  286. dca_bitalloc_index.offset = 1;
  287. dca_bitalloc_index.wrap = 2;
  288. for (i = 0; i < 5; i++) {
  289. dca_bitalloc_index.vlc[i].table = &dca_table[dca_vlc_offs[i]];
  290. dca_bitalloc_index.vlc[i].table_allocated = dca_vlc_offs[i + 1] - dca_vlc_offs[i];
  291. init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,
  292. bitalloc_12_bits[i], 1, 1,
  293. bitalloc_12_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
  294. }
  295. dca_scalefactor.offset = -64;
  296. dca_scalefactor.wrap = 2;
  297. for (i = 0; i < 5; i++) {
  298. dca_scalefactor.vlc[i].table = &dca_table[dca_vlc_offs[i + 5]];
  299. dca_scalefactor.vlc[i].table_allocated = dca_vlc_offs[i + 6] - dca_vlc_offs[i + 5];
  300. init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
  301. scales_bits[i], 1, 1,
  302. scales_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
  303. }
  304. dca_tmode.offset = 0;
  305. dca_tmode.wrap = 1;
  306. for (i = 0; i < 4; i++) {
  307. dca_tmode.vlc[i].table = &dca_table[dca_vlc_offs[i + 10]];
  308. dca_tmode.vlc[i].table_allocated = dca_vlc_offs[i + 11] - dca_vlc_offs[i + 10];
  309. init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
  310. tmode_bits[i], 1, 1,
  311. tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
  312. }
  313. for (i = 0; i < 10; i++)
  314. for (j = 0; j < 7; j++) {
  315. if (!bitalloc_codes[i][j])
  316. break;
  317. dca_smpl_bitalloc[i + 1].offset = bitalloc_offsets[i];
  318. dca_smpl_bitalloc[i + 1].wrap = 1 + (j > 4);
  319. dca_smpl_bitalloc[i + 1].vlc[j].table = &dca_table[dca_vlc_offs[c]];
  320. dca_smpl_bitalloc[i + 1].vlc[j].table_allocated = dca_vlc_offs[c + 1] - dca_vlc_offs[c];
  321. init_vlc(&dca_smpl_bitalloc[i + 1].vlc[j], bitalloc_maxbits[i][j],
  322. bitalloc_sizes[i],
  323. bitalloc_bits[i][j], 1, 1,
  324. bitalloc_codes[i][j], 2, 2, INIT_VLC_USE_NEW_STATIC);
  325. c++;
  326. }
  327. vlcs_initialized = 1;
  328. }
  329. static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
  330. {
  331. while (len--)
  332. *dst++ = get_bits(gb, bits);
  333. }
  334. static inline int dca_xxch2index(DCAContext *s, int xxch_ch)
  335. {
  336. int i, base, mask;
  337. /* locate channel set containing the channel */
  338. for (i = -1, base = 0, mask = (s->xxch_core_spkmask & ~DCA_XXCH_LFE1);
  339. i <= s->xxch_chset && !(mask & xxch_ch); mask = s->xxch_spk_masks[++i])
  340. base += av_popcount(mask);
  341. return base + av_popcount(mask & (xxch_ch - 1));
  342. }
  343. static int dca_parse_audio_coding_header(DCAContext *s, int base_channel,
  344. int xxch)
  345. {
  346. int i, j;
  347. static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
  348. static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
  349. static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
  350. int hdr_pos = 0, hdr_size = 0;
  351. float scale_factor;
  352. int this_chans, acc_mask;
  353. int embedded_downmix;
  354. int nchans, mask[8];
  355. int coeff, ichan;
  356. /* xxch has arbitrary sized audio coding headers */
  357. if (xxch) {
  358. hdr_pos = get_bits_count(&s->gb);
  359. hdr_size = get_bits(&s->gb, 7) + 1;
  360. }
  361. nchans = get_bits(&s->gb, 3) + 1;
  362. s->total_channels = nchans + base_channel;
  363. s->prim_channels = s->total_channels;
  364. /* obtain speaker layout mask & downmix coefficients for XXCH */
  365. if (xxch) {
  366. acc_mask = s->xxch_core_spkmask;
  367. this_chans = get_bits(&s->gb, s->xxch_nbits_spk_mask - 6) << 6;
  368. s->xxch_spk_masks[s->xxch_chset] = this_chans;
  369. s->xxch_chset_nch[s->xxch_chset] = nchans;
  370. for (i = 0; i <= s->xxch_chset; i++)
  371. acc_mask |= s->xxch_spk_masks[i];
  372. /* check for downmixing information */
  373. if (get_bits1(&s->gb)) {
  374. embedded_downmix = get_bits1(&s->gb);
  375. coeff = get_bits(&s->gb, 6);
  376. if (coeff<1 || coeff>61) {
  377. av_log(s->avctx, AV_LOG_ERROR, "6bit coeff %d is out of range\n", coeff);
  378. return AVERROR_INVALIDDATA;
  379. }
  380. scale_factor = -1.0f / dca_dmix_code((coeff<<2)-3);
  381. s->xxch_dmix_sf[s->xxch_chset] = scale_factor;
  382. for (i = base_channel; i < s->prim_channels; i++) {
  383. mask[i] = get_bits(&s->gb, s->xxch_nbits_spk_mask);
  384. }
  385. for (j = base_channel; j < s->prim_channels; j++) {
  386. memset(s->xxch_dmix_coeff[j], 0, sizeof(s->xxch_dmix_coeff[0]));
  387. s->xxch_dmix_embedded |= (embedded_downmix << j);
  388. for (i = 0; i < s->xxch_nbits_spk_mask; i++) {
  389. if (mask[j] & (1 << i)) {
  390. if ((1 << i) == DCA_XXCH_LFE1) {
  391. av_log(s->avctx, AV_LOG_WARNING,
  392. "DCA-XXCH: dmix to LFE1 not supported.\n");
  393. continue;
  394. }
  395. coeff = get_bits(&s->gb, 7);
  396. ichan = dca_xxch2index(s, 1 << i);
  397. if ((coeff&63)<1 || (coeff&63)>61) {
  398. av_log(s->avctx, AV_LOG_ERROR, "7bit coeff %d is out of range\n", coeff);
  399. return AVERROR_INVALIDDATA;
  400. }
  401. s->xxch_dmix_coeff[j][ichan] = dca_dmix_code((coeff<<2)-3);
  402. }
  403. }
  404. }
  405. }
  406. }
  407. if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)
  408. s->prim_channels = DCA_PRIM_CHANNELS_MAX;
  409. for (i = base_channel; i < s->prim_channels; i++) {
  410. s->subband_activity[i] = get_bits(&s->gb, 5) + 2;
  411. if (s->subband_activity[i] > DCA_SUBBANDS)
  412. s->subband_activity[i] = DCA_SUBBANDS;
  413. }
  414. for (i = base_channel; i < s->prim_channels; i++) {
  415. s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
  416. if (s->vq_start_subband[i] > DCA_SUBBANDS)
  417. s->vq_start_subband[i] = DCA_SUBBANDS;
  418. }
  419. get_array(&s->gb, s->joint_intensity + base_channel, s->prim_channels - base_channel, 3);
  420. get_array(&s->gb, s->transient_huffman + base_channel, s->prim_channels - base_channel, 2);
  421. get_array(&s->gb, s->scalefactor_huffman + base_channel, s->prim_channels - base_channel, 3);
  422. get_array(&s->gb, s->bitalloc_huffman + base_channel, s->prim_channels - base_channel, 3);
  423. /* Get codebooks quantization indexes */
  424. if (!base_channel)
  425. memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
  426. for (j = 1; j < 11; j++)
  427. for (i = base_channel; i < s->prim_channels; i++)
  428. s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
  429. /* Get scale factor adjustment */
  430. for (j = 0; j < 11; j++)
  431. for (i = base_channel; i < s->prim_channels; i++)
  432. s->scalefactor_adj[i][j] = 1;
  433. for (j = 1; j < 11; j++)
  434. for (i = base_channel; i < s->prim_channels; i++)
  435. if (s->quant_index_huffman[i][j] < thr[j])
  436. s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
  437. if (!xxch) {
  438. if (s->crc_present) {
  439. /* Audio header CRC check */
  440. get_bits(&s->gb, 16);
  441. }
  442. } else {
  443. /* Skip to the end of the header, also ignore CRC if present */
  444. i = get_bits_count(&s->gb);
  445. if (hdr_pos + 8 * hdr_size > i)
  446. skip_bits_long(&s->gb, hdr_pos + 8 * hdr_size - i);
  447. }
  448. s->current_subframe = 0;
  449. s->current_subsubframe = 0;
  450. return 0;
  451. }
  452. static int dca_parse_frame_header(DCAContext *s)
  453. {
  454. init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
  455. /* Sync code */
  456. skip_bits_long(&s->gb, 32);
  457. /* Frame header */
  458. s->frame_type = get_bits(&s->gb, 1);
  459. s->samples_deficit = get_bits(&s->gb, 5) + 1;
  460. s->crc_present = get_bits(&s->gb, 1);
  461. s->sample_blocks = get_bits(&s->gb, 7) + 1;
  462. s->frame_size = get_bits(&s->gb, 14) + 1;
  463. if (s->frame_size < 95)
  464. return AVERROR_INVALIDDATA;
  465. s->amode = get_bits(&s->gb, 6);
  466. s->sample_rate = avpriv_dca_sample_rates[get_bits(&s->gb, 4)];
  467. if (!s->sample_rate)
  468. return AVERROR_INVALIDDATA;
  469. s->bit_rate_index = get_bits(&s->gb, 5);
  470. s->bit_rate = dca_bit_rates[s->bit_rate_index];
  471. if (!s->bit_rate)
  472. return AVERROR_INVALIDDATA;
  473. skip_bits1(&s->gb); // always 0 (reserved, cf. ETSI TS 102 114 V1.4.1)
  474. s->dynrange = get_bits(&s->gb, 1);
  475. s->timestamp = get_bits(&s->gb, 1);
  476. s->aux_data = get_bits(&s->gb, 1);
  477. s->hdcd = get_bits(&s->gb, 1);
  478. s->ext_descr = get_bits(&s->gb, 3);
  479. s->ext_coding = get_bits(&s->gb, 1);
  480. s->aspf = get_bits(&s->gb, 1);
  481. s->lfe = get_bits(&s->gb, 2);
  482. s->predictor_history = get_bits(&s->gb, 1);
  483. if (s->lfe > 2) {
  484. s->lfe = 0;
  485. av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE value: %d\n", s->lfe);
  486. return AVERROR_INVALIDDATA;
  487. }
  488. /* TODO: check CRC */
  489. if (s->crc_present)
  490. s->header_crc = get_bits(&s->gb, 16);
  491. s->multirate_inter = get_bits(&s->gb, 1);
  492. s->version = get_bits(&s->gb, 4);
  493. s->copy_history = get_bits(&s->gb, 2);
  494. s->source_pcm_res = get_bits(&s->gb, 3);
  495. s->front_sum = get_bits(&s->gb, 1);
  496. s->surround_sum = get_bits(&s->gb, 1);
  497. s->dialog_norm = get_bits(&s->gb, 4);
  498. /* FIXME: channels mixing levels */
  499. s->output = s->amode;
  500. if (s->lfe)
  501. s->output |= DCA_LFE;
  502. /* Primary audio coding header */
  503. s->subframes = get_bits(&s->gb, 4) + 1;
  504. return dca_parse_audio_coding_header(s, 0, 0);
  505. }
  506. static inline int get_scale(GetBitContext *gb, int level, int value, int log2range)
  507. {
  508. if (level < 5) {
  509. /* huffman encoded */
  510. value += get_bitalloc(gb, &dca_scalefactor, level);
  511. value = av_clip(value, 0, (1 << log2range) - 1);
  512. } else if (level < 8) {
  513. if (level + 1 > log2range) {
  514. skip_bits(gb, level + 1 - log2range);
  515. value = get_bits(gb, log2range);
  516. } else {
  517. value = get_bits(gb, level + 1);
  518. }
  519. }
  520. return value;
  521. }
  522. static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
  523. {
  524. /* Primary audio coding side information */
  525. int j, k;
  526. if (get_bits_left(&s->gb) < 0)
  527. return AVERROR_INVALIDDATA;
  528. if (!base_channel) {
  529. s->subsubframes[s->current_subframe] = get_bits(&s->gb, 2) + 1;
  530. s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3);
  531. }
  532. for (j = base_channel; j < s->prim_channels; j++) {
  533. for (k = 0; k < s->subband_activity[j]; k++)
  534. s->prediction_mode[j][k] = get_bits(&s->gb, 1);
  535. }
  536. /* Get prediction codebook */
  537. for (j = base_channel; j < s->prim_channels; j++) {
  538. for (k = 0; k < s->subband_activity[j]; k++) {
  539. if (s->prediction_mode[j][k] > 0) {
  540. /* (Prediction coefficient VQ address) */
  541. s->prediction_vq[j][k] = get_bits(&s->gb, 12);
  542. }
  543. }
  544. }
  545. /* Bit allocation index */
  546. for (j = base_channel; j < s->prim_channels; j++) {
  547. for (k = 0; k < s->vq_start_subband[j]; k++) {
  548. if (s->bitalloc_huffman[j] == 6)
  549. s->bitalloc[j][k] = get_bits(&s->gb, 5);
  550. else if (s->bitalloc_huffman[j] == 5)
  551. s->bitalloc[j][k] = get_bits(&s->gb, 4);
  552. else if (s->bitalloc_huffman[j] == 7) {
  553. av_log(s->avctx, AV_LOG_ERROR,
  554. "Invalid bit allocation index\n");
  555. return AVERROR_INVALIDDATA;
  556. } else {
  557. s->bitalloc[j][k] =
  558. get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);
  559. }
  560. if (s->bitalloc[j][k] > 26) {
  561. av_dlog(s->avctx, "bitalloc index [%i][%i] too big (%i)\n",
  562. j, k, s->bitalloc[j][k]);
  563. return AVERROR_INVALIDDATA;
  564. }
  565. }
  566. }
  567. /* Transition mode */
  568. for (j = base_channel; j < s->prim_channels; j++) {
  569. for (k = 0; k < s->subband_activity[j]; k++) {
  570. s->transition_mode[j][k] = 0;
  571. if (s->subsubframes[s->current_subframe] > 1 &&
  572. k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) {
  573. s->transition_mode[j][k] =
  574. get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]);
  575. }
  576. }
  577. }
  578. if (get_bits_left(&s->gb) < 0)
  579. return AVERROR_INVALIDDATA;
  580. for (j = base_channel; j < s->prim_channels; j++) {
  581. const uint32_t *scale_table;
  582. int scale_sum, log_size;
  583. memset(s->scale_factor[j], 0,
  584. s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);
  585. if (s->scalefactor_huffman[j] == 6) {
  586. scale_table = scale_factor_quant7;
  587. log_size = 7;
  588. } else {
  589. scale_table = scale_factor_quant6;
  590. log_size = 6;
  591. }
  592. /* When huffman coded, only the difference is encoded */
  593. scale_sum = 0;
  594. for (k = 0; k < s->subband_activity[j]; k++) {
  595. if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) {
  596. scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size);
  597. s->scale_factor[j][k][0] = scale_table[scale_sum];
  598. }
  599. if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) {
  600. /* Get second scale factor */
  601. scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size);
  602. s->scale_factor[j][k][1] = scale_table[scale_sum];
  603. }
  604. }
  605. }
  606. /* Joint subband scale factor codebook select */
  607. for (j = base_channel; j < s->prim_channels; j++) {
  608. /* Transmitted only if joint subband coding enabled */
  609. if (s->joint_intensity[j] > 0)
  610. s->joint_huff[j] = get_bits(&s->gb, 3);
  611. }
  612. if (get_bits_left(&s->gb) < 0)
  613. return AVERROR_INVALIDDATA;
  614. /* Scale factors for joint subband coding */
  615. for (j = base_channel; j < s->prim_channels; j++) {
  616. int source_channel;
  617. /* Transmitted only if joint subband coding enabled */
  618. if (s->joint_intensity[j] > 0) {
  619. int scale = 0;
  620. source_channel = s->joint_intensity[j] - 1;
  621. /* When huffman coded, only the difference is encoded
  622. * (is this valid as well for joint scales ???) */
  623. for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) {
  624. scale = get_scale(&s->gb, s->joint_huff[j], 64 /* bias */, 7);
  625. s->joint_scale_factor[j][k] = scale; /*joint_scale_table[scale]; */
  626. }
  627. if (!(s->debug_flag & 0x02)) {
  628. av_log(s->avctx, AV_LOG_DEBUG,
  629. "Joint stereo coding not supported\n");
  630. s->debug_flag |= 0x02;
  631. }
  632. }
  633. }
  634. /* Dynamic range coefficient */
  635. if (!base_channel && s->dynrange)
  636. s->dynrange_coef = get_bits(&s->gb, 8);
  637. /* Side information CRC check word */
  638. if (s->crc_present) {
  639. get_bits(&s->gb, 16);
  640. }
  641. /*
  642. * Primary audio data arrays
  643. */
  644. /* VQ encoded high frequency subbands */
  645. for (j = base_channel; j < s->prim_channels; j++)
  646. for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
  647. /* 1 vector -> 32 samples */
  648. s->high_freq_vq[j][k] = get_bits(&s->gb, 10);
  649. /* Low frequency effect data */
  650. if (!base_channel && s->lfe) {
  651. int quant7;
  652. /* LFE samples */
  653. int lfe_samples = 2 * s->lfe * (4 + block_index);
  654. int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);
  655. float lfe_scale;
  656. for (j = lfe_samples; j < lfe_end_sample; j++) {
  657. /* Signed 8 bits int */
  658. s->lfe_data[j] = get_sbits(&s->gb, 8);
  659. }
  660. /* Scale factor index */
  661. quant7 = get_bits(&s->gb, 8);
  662. if (quant7 > 127) {
  663. avpriv_request_sample(s->avctx, "LFEScaleIndex larger than 127");
  664. return AVERROR_INVALIDDATA;
  665. }
  666. s->lfe_scale_factor = scale_factor_quant7[quant7];
  667. /* Quantization step size * scale factor */
  668. lfe_scale = 0.035 * s->lfe_scale_factor;
  669. for (j = lfe_samples; j < lfe_end_sample; j++)
  670. s->lfe_data[j] *= lfe_scale;
  671. }
  672. return 0;
  673. }
  674. static void qmf_32_subbands(DCAContext *s, int chans,
  675. float samples_in[32][8], float *samples_out,
  676. float scale)
  677. {
  678. const float *prCoeff;
  679. int sb_act = s->subband_activity[chans];
  680. scale *= sqrt(1 / 8.0);
  681. /* Select filter */
  682. if (!s->multirate_inter) /* Non-perfect reconstruction */
  683. prCoeff = fir_32bands_nonperfect;
  684. else /* Perfect reconstruction */
  685. prCoeff = fir_32bands_perfect;
  686. s->dcadsp.qmf_32_subbands(samples_in, sb_act, &s->synth, &s->imdct,
  687. s->subband_fir_hist[chans],
  688. &s->hist_index[chans],
  689. s->subband_fir_noidea[chans], prCoeff,
  690. samples_out, s->raXin, scale);
  691. }
  692. static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
  693. int num_deci_sample, float *samples_in,
  694. float *samples_out)
  695. {
  696. /* samples_in: An array holding decimated samples.
  697. * Samples in current subframe starts from samples_in[0],
  698. * while samples_in[-1], samples_in[-2], ..., stores samples
  699. * from last subframe as history.
  700. *
  701. * samples_out: An array holding interpolated samples
  702. */
  703. int idx;
  704. const float *prCoeff;
  705. int deciindex;
  706. /* Select decimation filter */
  707. if (decimation_select == 1) {
  708. idx = 1;
  709. prCoeff = lfe_fir_128;
  710. } else {
  711. idx = 0;
  712. prCoeff = lfe_fir_64;
  713. }
  714. /* Interpolation */
  715. for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
  716. s->dcadsp.lfe_fir[idx](samples_out, samples_in, prCoeff);
  717. samples_in++;
  718. samples_out += 2 * 32 * (1 + idx);
  719. }
  720. }
  721. /* downmixing routines */
  722. #define MIX_REAR1(samples, s1, rs, coef) \
  723. samples[0][i] += samples[s1][i] * coef[rs][0]; \
  724. samples[1][i] += samples[s1][i] * coef[rs][1];
  725. #define MIX_REAR2(samples, s1, s2, rs, coef) \
  726. samples[0][i] += samples[s1][i] * coef[rs][0] + samples[s2][i] * coef[rs + 1][0]; \
  727. samples[1][i] += samples[s1][i] * coef[rs][1] + samples[s2][i] * coef[rs + 1][1];
  728. #define MIX_FRONT3(samples, coef) \
  729. t = samples[c][i]; \
  730. u = samples[l][i]; \
  731. v = samples[r][i]; \
  732. samples[0][i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \
  733. samples[1][i] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];
  734. #define DOWNMIX_TO_STEREO(op1, op2) \
  735. for (i = 0; i < 256; i++) { \
  736. op1 \
  737. op2 \
  738. }
  739. static void dca_downmix(float **samples, int srcfmt, int lfe_present,
  740. float coef[DCA_PRIM_CHANNELS_MAX + 1][2],
  741. const int8_t *channel_mapping)
  742. {
  743. int c, l, r, sl, sr, s;
  744. int i;
  745. float t, u, v;
  746. switch (srcfmt) {
  747. case DCA_MONO:
  748. case DCA_4F2R:
  749. av_log(NULL, AV_LOG_ERROR, "Not implemented!\n");
  750. break;
  751. case DCA_CHANNEL:
  752. case DCA_STEREO:
  753. case DCA_STEREO_TOTAL:
  754. case DCA_STEREO_SUMDIFF:
  755. break;
  756. case DCA_3F:
  757. c = channel_mapping[0];
  758. l = channel_mapping[1];
  759. r = channel_mapping[2];
  760. DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );
  761. break;
  762. case DCA_2F1R:
  763. s = channel_mapping[2];
  764. DOWNMIX_TO_STEREO(MIX_REAR1(samples, s, 2, coef), );
  765. break;
  766. case DCA_3F1R:
  767. c = channel_mapping[0];
  768. l = channel_mapping[1];
  769. r = channel_mapping[2];
  770. s = channel_mapping[3];
  771. DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
  772. MIX_REAR1(samples, s, 3, coef));
  773. break;
  774. case DCA_2F2R:
  775. sl = channel_mapping[2];
  776. sr = channel_mapping[3];
  777. DOWNMIX_TO_STEREO(MIX_REAR2(samples, sl, sr, 2, coef), );
  778. break;
  779. case DCA_3F2R:
  780. c = channel_mapping[0];
  781. l = channel_mapping[1];
  782. r = channel_mapping[2];
  783. sl = channel_mapping[3];
  784. sr = channel_mapping[4];
  785. DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
  786. MIX_REAR2(samples, sl, sr, 3, coef));
  787. break;
  788. }
  789. if (lfe_present) {
  790. int lf_buf = dca_lfe_index[srcfmt];
  791. int lf_idx = dca_channels[srcfmt];
  792. for (i = 0; i < 256; i++) {
  793. samples[0][i] += samples[lf_buf][i] * coef[lf_idx][0];
  794. samples[1][i] += samples[lf_buf][i] * coef[lf_idx][1];
  795. }
  796. }
  797. }
  798. #ifndef decode_blockcodes
  799. /* Very compact version of the block code decoder that does not use table
  800. * look-up but is slightly slower */
  801. static int decode_blockcode(int code, int levels, int32_t *values)
  802. {
  803. int i;
  804. int offset = (levels - 1) >> 1;
  805. for (i = 0; i < 4; i++) {
  806. int div = FASTDIV(code, levels);
  807. values[i] = code - offset - div * levels;
  808. code = div;
  809. }
  810. return code;
  811. }
  812. static int decode_blockcodes(int code1, int code2, int levels, int32_t *values)
  813. {
  814. return decode_blockcode(code1, levels, values) |
  815. decode_blockcode(code2, levels, values + 4);
  816. }
  817. #endif
  818. static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
  819. static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
  820. static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
  821. {
  822. int k, l;
  823. int subsubframe = s->current_subsubframe;
  824. const float *quant_step_table;
  825. /* FIXME */
  826. float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
  827. LOCAL_ALIGNED_16(int32_t, block, [8 * DCA_SUBBANDS]);
  828. /*
  829. * Audio data
  830. */
  831. /* Select quantization step size table */
  832. if (s->bit_rate_index == 0x1f)
  833. quant_step_table = lossless_quant_d;
  834. else
  835. quant_step_table = lossy_quant_d;
  836. for (k = base_channel; k < s->prim_channels; k++) {
  837. float rscale[DCA_SUBBANDS];
  838. if (get_bits_left(&s->gb) < 0)
  839. return AVERROR_INVALIDDATA;
  840. for (l = 0; l < s->vq_start_subband[k]; l++) {
  841. int m;
  842. /* Select the mid-tread linear quantizer */
  843. int abits = s->bitalloc[k][l];
  844. float quant_step_size = quant_step_table[abits];
  845. /*
  846. * Determine quantization index code book and its type
  847. */
  848. /* Select quantization index code book */
  849. int sel = s->quant_index_huffman[k][abits];
  850. /*
  851. * Extract bits from the bit stream
  852. */
  853. if (!abits) {
  854. rscale[l] = 0;
  855. memset(block + 8 * l, 0, 8 * sizeof(block[0]));
  856. } else {
  857. /* Deal with transients */
  858. int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];
  859. rscale[l] = quant_step_size * s->scale_factor[k][l][sfi] *
  860. s->scalefactor_adj[k][sel];
  861. if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {
  862. if (abits <= 7) {
  863. /* Block code */
  864. int block_code1, block_code2, size, levels, err;
  865. size = abits_sizes[abits - 1];
  866. levels = abits_levels[abits - 1];
  867. block_code1 = get_bits(&s->gb, size);
  868. block_code2 = get_bits(&s->gb, size);
  869. err = decode_blockcodes(block_code1, block_code2,
  870. levels, block + 8 * l);
  871. if (err) {
  872. av_log(s->avctx, AV_LOG_ERROR,
  873. "ERROR: block code look-up failed\n");
  874. return AVERROR_INVALIDDATA;
  875. }
  876. } else {
  877. /* no coding */
  878. for (m = 0; m < 8; m++)
  879. block[8 * l + m] = get_sbits(&s->gb, abits - 3);
  880. }
  881. } else {
  882. /* Huffman coded */
  883. for (m = 0; m < 8; m++)
  884. block[8 * l + m] = get_bitalloc(&s->gb,
  885. &dca_smpl_bitalloc[abits], sel);
  886. }
  887. }
  888. }
  889. s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[k][0],
  890. block, rscale, 8 * s->vq_start_subband[k]);
  891. for (l = 0; l < s->vq_start_subband[k]; l++) {
  892. int m;
  893. /*
  894. * Inverse ADPCM if in prediction mode
  895. */
  896. if (s->prediction_mode[k][l]) {
  897. int n;
  898. if (s->predictor_history)
  899. subband_samples[k][l][0] += (adpcm_vb[s->prediction_vq[k][l]][0] *
  900. s->subband_samples_hist[k][l][3] +
  901. adpcm_vb[s->prediction_vq[k][l]][1] *
  902. s->subband_samples_hist[k][l][2] +
  903. adpcm_vb[s->prediction_vq[k][l]][2] *
  904. s->subband_samples_hist[k][l][1] +
  905. adpcm_vb[s->prediction_vq[k][l]][3] *
  906. s->subband_samples_hist[k][l][0]) *
  907. (1.0f / 8192);
  908. for (m = 1; m < 8; m++) {
  909. float sum = adpcm_vb[s->prediction_vq[k][l]][0] *
  910. subband_samples[k][l][m - 1];
  911. for (n = 2; n <= 4; n++)
  912. if (m >= n)
  913. sum += adpcm_vb[s->prediction_vq[k][l]][n - 1] *
  914. subband_samples[k][l][m - n];
  915. else if (s->predictor_history)
  916. sum += adpcm_vb[s->prediction_vq[k][l]][n - 1] *
  917. s->subband_samples_hist[k][l][m - n + 4];
  918. subband_samples[k][l][m] += sum * (1.0f / 8192);
  919. }
  920. }
  921. }
  922. /*
  923. * Decode VQ encoded high frequencies
  924. */
  925. if (s->subband_activity[k] > s->vq_start_subband[k]) {
  926. if (!(s->debug_flag & 0x01)) {
  927. av_log(s->avctx, AV_LOG_DEBUG,
  928. "Stream with high frequencies VQ coding\n");
  929. s->debug_flag |= 0x01;
  930. }
  931. s->dcadsp.decode_hf(subband_samples[k], s->high_freq_vq[k],
  932. high_freq_vq, subsubframe * 8,
  933. s->scale_factor[k], s->vq_start_subband[k],
  934. s->subband_activity[k]);
  935. }
  936. }
  937. /* Check for DSYNC after subsubframe */
  938. if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {
  939. if (get_bits(&s->gb, 16) != 0xFFFF) {
  940. av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
  941. return AVERROR_INVALIDDATA;
  942. }
  943. }
  944. /* Backup predictor history for adpcm */
  945. for (k = base_channel; k < s->prim_channels; k++)
  946. for (l = 0; l < s->vq_start_subband[k]; l++)
  947. AV_COPY128(s->subband_samples_hist[k][l], &subband_samples[k][l][4]);
  948. return 0;
  949. }
  950. static int dca_filter_channels(DCAContext *s, int block_index)
  951. {
  952. float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
  953. int k;
  954. /* 32 subbands QMF */
  955. for (k = 0; k < s->prim_channels; k++) {
  956. if (s->channel_order_tab[k] >= 0)
  957. qmf_32_subbands(s, k, subband_samples[k],
  958. s->samples_chanptr[s->channel_order_tab[k]],
  959. M_SQRT1_2 / 32768.0);
  960. }
  961. /* Generate LFE samples for this subsubframe FIXME!!! */
  962. if (s->lfe) {
  963. lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,
  964. s->lfe_data + 2 * s->lfe * (block_index + 4),
  965. s->samples_chanptr[s->lfe_index]);
  966. /* Outputs 20bits pcm samples */
  967. }
  968. /* Downmixing to Stereo */
  969. if (s->prim_channels + !!s->lfe > 2 &&
  970. s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
  971. dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef,
  972. s->channel_order_tab);
  973. }
  974. return 0;
  975. }
  976. static int dca_subframe_footer(DCAContext *s, int base_channel)
  977. {
  978. int in, out, aux_data_count, aux_data_end, reserved;
  979. uint32_t nsyncaux;
  980. /*
  981. * Unpack optional information
  982. */
  983. /* presumably optional information only appears in the core? */
  984. if (!base_channel) {
  985. if (s->timestamp)
  986. skip_bits_long(&s->gb, 32);
  987. if (s->aux_data) {
  988. aux_data_count = get_bits(&s->gb, 6);
  989. // align (32-bit)
  990. skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
  991. aux_data_end = 8 * aux_data_count + get_bits_count(&s->gb);
  992. if ((nsyncaux = get_bits_long(&s->gb, 32)) != DCA_NSYNCAUX) {
  993. av_log(s->avctx, AV_LOG_ERROR, "nSYNCAUX mismatch %#"PRIx32"\n",
  994. nsyncaux);
  995. return AVERROR_INVALIDDATA;
  996. }
  997. if (get_bits1(&s->gb)) { // bAUXTimeStampFlag
  998. avpriv_request_sample(s->avctx,
  999. "Auxiliary Decode Time Stamp Flag");
  1000. // align (4-bit)
  1001. skip_bits(&s->gb, (-get_bits_count(&s->gb)) & 4);
  1002. // 44 bits: nMSByte (8), nMarker (4), nLSByte (28), nMarker (4)
  1003. skip_bits_long(&s->gb, 44);
  1004. }
  1005. if ((s->core_downmix = get_bits1(&s->gb))) {
  1006. int am = get_bits(&s->gb, 3);
  1007. switch (am) {
  1008. case 0:
  1009. s->core_downmix_amode = DCA_MONO;
  1010. break;
  1011. case 1:
  1012. s->core_downmix_amode = DCA_STEREO;
  1013. break;
  1014. case 2:
  1015. s->core_downmix_amode = DCA_STEREO_TOTAL;
  1016. break;
  1017. case 3:
  1018. s->core_downmix_amode = DCA_3F;
  1019. break;
  1020. case 4:
  1021. s->core_downmix_amode = DCA_2F1R;
  1022. break;
  1023. case 5:
  1024. s->core_downmix_amode = DCA_2F2R;
  1025. break;
  1026. case 6:
  1027. s->core_downmix_amode = DCA_3F1R;
  1028. break;
  1029. default:
  1030. av_log(s->avctx, AV_LOG_ERROR,
  1031. "Invalid mode %d for embedded downmix coefficients\n",
  1032. am);
  1033. return AVERROR_INVALIDDATA;
  1034. }
  1035. for (out = 0; out < dca_channels[s->core_downmix_amode]; out++) {
  1036. for (in = 0; in < s->prim_channels + !!s->lfe; in++) {
  1037. uint16_t tmp = get_bits(&s->gb, 9);
  1038. if ((tmp & 0xFF) > 241) {
  1039. av_log(s->avctx, AV_LOG_ERROR,
  1040. "Invalid downmix coefficient code %"PRIu16"\n",
  1041. tmp);
  1042. return AVERROR_INVALIDDATA;
  1043. }
  1044. s->core_downmix_codes[in][out] = tmp;
  1045. }
  1046. }
  1047. }
  1048. align_get_bits(&s->gb); // byte align
  1049. skip_bits(&s->gb, 16); // nAUXCRC16
  1050. // additional data (reserved, cf. ETSI TS 102 114 V1.4.1)
  1051. if ((reserved = (aux_data_end - get_bits_count(&s->gb))) < 0) {
  1052. av_log(s->avctx, AV_LOG_ERROR,
  1053. "Overread auxiliary data by %d bits\n", -reserved);
  1054. return AVERROR_INVALIDDATA;
  1055. } else if (reserved) {
  1056. avpriv_request_sample(s->avctx,
  1057. "Core auxiliary data reserved content");
  1058. skip_bits_long(&s->gb, reserved);
  1059. }
  1060. }
  1061. if (s->crc_present && s->dynrange)
  1062. get_bits(&s->gb, 16);
  1063. }
  1064. return 0;
  1065. }
  1066. /**
  1067. * Decode a dca frame block
  1068. *
  1069. * @param s pointer to the DCAContext
  1070. */
  1071. static int dca_decode_block(DCAContext *s, int base_channel, int block_index)
  1072. {
  1073. int ret;
  1074. /* Sanity check */
  1075. if (s->current_subframe >= s->subframes) {
  1076. av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
  1077. s->current_subframe, s->subframes);
  1078. return AVERROR_INVALIDDATA;
  1079. }
  1080. if (!s->current_subsubframe) {
  1081. /* Read subframe header */
  1082. if ((ret = dca_subframe_header(s, base_channel, block_index)))
  1083. return ret;
  1084. }
  1085. /* Read subsubframe */
  1086. if ((ret = dca_subsubframe(s, base_channel, block_index)))
  1087. return ret;
  1088. /* Update state */
  1089. s->current_subsubframe++;
  1090. if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {
  1091. s->current_subsubframe = 0;
  1092. s->current_subframe++;
  1093. }
  1094. if (s->current_subframe >= s->subframes) {
  1095. /* Read subframe footer */
  1096. if ((ret = dca_subframe_footer(s, base_channel)))
  1097. return ret;
  1098. }
  1099. return 0;
  1100. }
  1101. int ff_dca_xbr_parse_frame(DCAContext *s)
  1102. {
  1103. int scale_table_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS][2];
  1104. int active_bands[DCA_CHSETS_MAX][DCA_CHSET_CHANS_MAX];
  1105. int abits_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS];
  1106. int anctemp[DCA_CHSET_CHANS_MAX];
  1107. int chset_fsize[DCA_CHSETS_MAX];
  1108. int n_xbr_ch[DCA_CHSETS_MAX];
  1109. int hdr_size, num_chsets, xbr_tmode, hdr_pos;
  1110. int i, j, k, l, chset, chan_base;
  1111. av_log(s->avctx, AV_LOG_DEBUG, "DTS-XBR: decoding XBR extension\n");
  1112. /* get bit position of sync header */
  1113. hdr_pos = get_bits_count(&s->gb) - 32;
  1114. hdr_size = get_bits(&s->gb, 6) + 1;
  1115. num_chsets = get_bits(&s->gb, 2) + 1;
  1116. for(i = 0; i < num_chsets; i++)
  1117. chset_fsize[i] = get_bits(&s->gb, 14) + 1;
  1118. xbr_tmode = get_bits1(&s->gb);
  1119. for(i = 0; i < num_chsets; i++) {
  1120. n_xbr_ch[i] = get_bits(&s->gb, 3) + 1;
  1121. k = get_bits(&s->gb, 2) + 5;
  1122. for(j = 0; j < n_xbr_ch[i]; j++)
  1123. active_bands[i][j] = get_bits(&s->gb, k) + 1;
  1124. }
  1125. /* skip to the end of the header */
  1126. i = get_bits_count(&s->gb);
  1127. if(hdr_pos + hdr_size * 8 > i)
  1128. skip_bits_long(&s->gb, hdr_pos + hdr_size * 8 - i);
  1129. /* loop over the channel data sets */
  1130. /* only decode as many channels as we've decoded base data for */
  1131. for(chset = 0, chan_base = 0;
  1132. chset < num_chsets && chan_base + n_xbr_ch[chset] <= s->prim_channels;
  1133. chan_base += n_xbr_ch[chset++]) {
  1134. int start_posn = get_bits_count(&s->gb);
  1135. int subsubframe = 0;
  1136. int subframe = 0;
  1137. /* loop over subframes */
  1138. for (k = 0; k < (s->sample_blocks / 8); k++) {
  1139. /* parse header if we're on first subsubframe of a block */
  1140. if(subsubframe == 0) {
  1141. /* Parse subframe header */
  1142. for(i = 0; i < n_xbr_ch[chset]; i++) {
  1143. anctemp[i] = get_bits(&s->gb, 2) + 2;
  1144. }
  1145. for(i = 0; i < n_xbr_ch[chset]; i++) {
  1146. get_array(&s->gb, abits_high[i], active_bands[chset][i], anctemp[i]);
  1147. }
  1148. for(i = 0; i < n_xbr_ch[chset]; i++) {
  1149. anctemp[i] = get_bits(&s->gb, 3);
  1150. if(anctemp[i] < 1) {
  1151. av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: SYNC ERROR\n");
  1152. return AVERROR_INVALIDDATA;
  1153. }
  1154. }
  1155. /* generate scale factors */
  1156. for(i = 0; i < n_xbr_ch[chset]; i++) {
  1157. const uint32_t *scale_table;
  1158. int nbits;
  1159. if (s->scalefactor_huffman[chan_base+i] == 6) {
  1160. scale_table = scale_factor_quant7;
  1161. } else {
  1162. scale_table = scale_factor_quant6;
  1163. }
  1164. nbits = anctemp[i];
  1165. for(j = 0; j < active_bands[chset][i]; j++) {
  1166. if(abits_high[i][j] > 0) {
  1167. scale_table_high[i][j][0] =
  1168. scale_table[get_bits(&s->gb, nbits)];
  1169. if(xbr_tmode && s->transition_mode[i][j]) {
  1170. scale_table_high[i][j][1] =
  1171. scale_table[get_bits(&s->gb, nbits)];
  1172. }
  1173. }
  1174. }
  1175. }
  1176. }
  1177. /* decode audio array for this block */
  1178. for(i = 0; i < n_xbr_ch[chset]; i++) {
  1179. for(j = 0; j < active_bands[chset][i]; j++) {
  1180. const int xbr_abits = abits_high[i][j];
  1181. const float quant_step_size = lossless_quant_d[xbr_abits];
  1182. const int sfi = xbr_tmode && s->transition_mode[i][j] && subsubframe >= s->transition_mode[i][j];
  1183. const float rscale = quant_step_size * scale_table_high[i][j][sfi];
  1184. float *subband_samples = s->subband_samples[k][chan_base+i][j];
  1185. int block[8];
  1186. if(xbr_abits <= 0)
  1187. continue;
  1188. if(xbr_abits > 7) {
  1189. get_array(&s->gb, block, 8, xbr_abits - 3);
  1190. } else {
  1191. int block_code1, block_code2, size, levels, err;
  1192. size = abits_sizes[xbr_abits - 1];
  1193. levels = abits_levels[xbr_abits - 1];
  1194. block_code1 = get_bits(&s->gb, size);
  1195. block_code2 = get_bits(&s->gb, size);
  1196. err = decode_blockcodes(block_code1, block_code2,
  1197. levels, block);
  1198. if (err) {
  1199. av_log(s->avctx, AV_LOG_ERROR,
  1200. "ERROR: DTS-XBR: block code look-up failed\n");
  1201. return AVERROR_INVALIDDATA;
  1202. }
  1203. }
  1204. /* scale & sum into subband */
  1205. for(l = 0; l < 8; l++)
  1206. subband_samples[l] += (float)block[l] * rscale;
  1207. }
  1208. }
  1209. /* check DSYNC marker */
  1210. if(s->aspf || subsubframe == s->subsubframes[subframe] - 1) {
  1211. if(get_bits(&s->gb, 16) != 0xffff) {
  1212. av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: Didn't get subframe DSYNC\n");
  1213. return AVERROR_INVALIDDATA;
  1214. }
  1215. }
  1216. /* advance sub-sub-frame index */
  1217. if(++subsubframe >= s->subsubframes[subframe]) {
  1218. subsubframe = 0;
  1219. subframe++;
  1220. }
  1221. }
  1222. /* skip to next channel set */
  1223. i = get_bits_count(&s->gb);
  1224. if(start_posn + chset_fsize[chset] * 8 != i) {
  1225. j = start_posn + chset_fsize[chset] * 8 - i;
  1226. if(j < 0 || j >= 8)
  1227. av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: end of channel set,"
  1228. " skipping further than expected (%d bits)\n", j);
  1229. skip_bits_long(&s->gb, j);
  1230. }
  1231. }
  1232. return 0;
  1233. }
  1234. /* parse initial header for XXCH and dump details */
  1235. int ff_dca_xxch_decode_frame(DCAContext *s)
  1236. {
  1237. int hdr_size, spkmsk_bits, num_chsets, core_spk, hdr_pos;
  1238. int i, chset, base_channel, chstart, fsize[8];
  1239. /* assume header word has already been parsed */
  1240. hdr_pos = get_bits_count(&s->gb) - 32;
  1241. hdr_size = get_bits(&s->gb, 6) + 1;
  1242. /*chhdr_crc =*/ skip_bits1(&s->gb);
  1243. spkmsk_bits = get_bits(&s->gb, 5) + 1;
  1244. num_chsets = get_bits(&s->gb, 2) + 1;
  1245. for (i = 0; i < num_chsets; i++)
  1246. fsize[i] = get_bits(&s->gb, 14) + 1;
  1247. core_spk = get_bits(&s->gb, spkmsk_bits);
  1248. s->xxch_core_spkmask = core_spk;
  1249. s->xxch_nbits_spk_mask = spkmsk_bits;
  1250. s->xxch_dmix_embedded = 0;
  1251. /* skip to the end of the header */
  1252. i = get_bits_count(&s->gb);
  1253. if (hdr_pos + hdr_size * 8 > i)
  1254. skip_bits_long(&s->gb, hdr_pos + hdr_size * 8 - i);
  1255. for (chset = 0; chset < num_chsets; chset++) {
  1256. chstart = get_bits_count(&s->gb);
  1257. base_channel = s->prim_channels;
  1258. s->xxch_chset = chset;
  1259. /* XXCH and Core headers differ, see 6.4.2 "XXCH Channel Set Header" vs.
  1260. 5.3.2 "Primary Audio Coding Header", DTS Spec 1.3.1 */
  1261. dca_parse_audio_coding_header(s, base_channel, 1);
  1262. /* decode channel data */
  1263. for (i = 0; i < (s->sample_blocks / 8); i++) {
  1264. if (dca_decode_block(s, base_channel, i)) {
  1265. av_log(s->avctx, AV_LOG_ERROR,
  1266. "Error decoding DTS-XXCH extension\n");
  1267. continue;
  1268. }
  1269. }
  1270. /* skip to end of this section */
  1271. i = get_bits_count(&s->gb);
  1272. if (chstart + fsize[chset] * 8 > i)
  1273. skip_bits_long(&s->gb, chstart + fsize[chset] * 8 - i);
  1274. }
  1275. s->xxch_chset = num_chsets;
  1276. return 0;
  1277. }
  1278. static float dca_dmix_code(unsigned code)
  1279. {
  1280. int sign = (code >> 8) - 1;
  1281. code &= 0xff;
  1282. return ((dca_dmixtable[code] ^ sign) - sign) * (1.0 / (1 << 15));
  1283. }
  1284. /**
  1285. * Main frame decoding function
  1286. * FIXME add arguments
  1287. */
  1288. static int dca_decode_frame(AVCodecContext *avctx, void *data,
  1289. int *got_frame_ptr, AVPacket *avpkt)
  1290. {
  1291. AVFrame *frame = data;
  1292. const uint8_t *buf = avpkt->data;
  1293. int buf_size = avpkt->size;
  1294. int channel_mask;
  1295. int channel_layout;
  1296. int lfe_samples;
  1297. int num_core_channels = 0;
  1298. int i, ret;
  1299. float **samples_flt;
  1300. float *src_chan;
  1301. float *dst_chan;
  1302. DCAContext *s = avctx->priv_data;
  1303. int core_ss_end;
  1304. int channels, full_channels;
  1305. float scale;
  1306. int achan;
  1307. int chset;
  1308. int mask;
  1309. int lavc;
  1310. int posn;
  1311. int j, k;
  1312. int endch;
  1313. s->xch_present = 0;
  1314. s->dca_buffer_size = avpriv_dca_convert_bitstream(buf, buf_size, s->dca_buffer,
  1315. DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);
  1316. if (s->dca_buffer_size == AVERROR_INVALIDDATA) {
  1317. av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
  1318. return AVERROR_INVALIDDATA;
  1319. }
  1320. if ((ret = dca_parse_frame_header(s)) < 0) {
  1321. // seems like the frame is corrupt, try with the next one
  1322. return ret;
  1323. }
  1324. // set AVCodec values with parsed data
  1325. avctx->sample_rate = s->sample_rate;
  1326. avctx->bit_rate = s->bit_rate;
  1327. s->profile = FF_PROFILE_DTS;
  1328. for (i = 0; i < (s->sample_blocks / 8); i++) {
  1329. if ((ret = dca_decode_block(s, 0, i))) {
  1330. av_log(avctx, AV_LOG_ERROR, "error decoding block\n");
  1331. return ret;
  1332. }
  1333. }
  1334. /* record number of core channels incase less than max channels are requested */
  1335. num_core_channels = s->prim_channels;
  1336. if (s->prim_channels + !!s->lfe > 2 &&
  1337. avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
  1338. /* Stereo downmix coefficients
  1339. *
  1340. * The decoder can only downmix to 2-channel, so we need to ensure
  1341. * embedded downmix coefficients are actually targeting 2-channel.
  1342. */
  1343. if (s->core_downmix && (s->core_downmix_amode == DCA_STEREO ||
  1344. s->core_downmix_amode == DCA_STEREO_TOTAL)) {
  1345. for (i = 0; i < num_core_channels + !!s->lfe; i++) {
  1346. /* Range checked earlier */
  1347. s->downmix_coef[i][0] = dca_dmix_code(s->core_downmix_codes[i][0]);
  1348. s->downmix_coef[i][1] = dca_dmix_code(s->core_downmix_codes[i][1]);
  1349. }
  1350. s->output = s->core_downmix_amode;
  1351. } else {
  1352. int am = s->amode & DCA_CHANNEL_MASK;
  1353. if (am >= FF_ARRAY_ELEMS(dca_default_coeffs)) {
  1354. av_log(s->avctx, AV_LOG_ERROR,
  1355. "Invalid channel mode %d\n", am);
  1356. return AVERROR_INVALIDDATA;
  1357. }
  1358. if (num_core_channels + !!s->lfe >
  1359. FF_ARRAY_ELEMS(dca_default_coeffs[0])) {
  1360. avpriv_request_sample(s->avctx, "Downmixing %d channels",
  1361. s->prim_channels + !!s->lfe);
  1362. return AVERROR_PATCHWELCOME;
  1363. }
  1364. for (i = 0; i < num_core_channels + !!s->lfe; i++) {
  1365. s->downmix_coef[i][0] = dca_default_coeffs[am][i][0];
  1366. s->downmix_coef[i][1] = dca_default_coeffs[am][i][1];
  1367. }
  1368. }
  1369. av_dlog(s->avctx, "Stereo downmix coeffs:\n");
  1370. for (i = 0; i < num_core_channels + !!s->lfe; i++) {
  1371. av_dlog(s->avctx, "L, input channel %d = %f\n", i,
  1372. s->downmix_coef[i][0]);
  1373. av_dlog(s->avctx, "R, input channel %d = %f\n", i,
  1374. s->downmix_coef[i][1]);
  1375. }
  1376. av_dlog(s->avctx, "\n");
  1377. }
  1378. if (s->ext_coding)
  1379. s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];
  1380. else
  1381. s->core_ext_mask = 0;
  1382. core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;
  1383. /* only scan for extensions if ext_descr was unknown or indicated a
  1384. * supported XCh extension */
  1385. if (s->core_ext_mask < 0 || s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH)) {
  1386. /* if ext_descr was unknown, clear s->core_ext_mask so that the
  1387. * extensions scan can fill it up */
  1388. s->core_ext_mask = FFMAX(s->core_ext_mask, 0);
  1389. /* extensions start at 32-bit boundaries into bitstream */
  1390. skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
  1391. while (core_ss_end - get_bits_count(&s->gb) >= 32) {
  1392. uint32_t bits = get_bits_long(&s->gb, 32);
  1393. switch (bits) {
  1394. case 0x5a5a5a5a: {
  1395. int ext_amode, xch_fsize;
  1396. s->xch_base_channel = s->prim_channels;
  1397. /* validate sync word using XCHFSIZE field */
  1398. xch_fsize = show_bits(&s->gb, 10);
  1399. if ((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&
  1400. (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))
  1401. continue;
  1402. /* skip length-to-end-of-frame field for the moment */
  1403. skip_bits(&s->gb, 10);
  1404. s->core_ext_mask |= DCA_EXT_XCH;
  1405. /* extension amode(number of channels in extension) should be 1 */
  1406. /* AFAIK XCh is not used for more channels */
  1407. if ((ext_amode = get_bits(&s->gb, 4)) != 1) {
  1408. av_log(avctx, AV_LOG_ERROR,
  1409. "XCh extension amode %d not supported!\n",
  1410. ext_amode);
  1411. continue;
  1412. }
  1413. if (s->xch_base_channel < 2) {
  1414. avpriv_request_sample(avctx, "XCh with fewer than 2 base channels");
  1415. continue;
  1416. }
  1417. /* much like core primary audio coding header */
  1418. dca_parse_audio_coding_header(s, s->xch_base_channel, 0);
  1419. for (i = 0; i < (s->sample_blocks / 8); i++)
  1420. if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {
  1421. av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");
  1422. continue;
  1423. }
  1424. s->xch_present = 1;
  1425. break;
  1426. }
  1427. case 0x47004a03:
  1428. /* XXCh: extended channels */
  1429. /* usually found either in core or HD part in DTS-HD HRA streams,
  1430. * but not in DTS-ES which contains XCh extensions instead */
  1431. s->core_ext_mask |= DCA_EXT_XXCH;
  1432. ff_dca_xxch_decode_frame(s);
  1433. break;
  1434. case 0x1d95f262: {
  1435. int fsize96 = show_bits(&s->gb, 12) + 1;
  1436. if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)
  1437. continue;
  1438. av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n",
  1439. get_bits_count(&s->gb));
  1440. skip_bits(&s->gb, 12);
  1441. av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);
  1442. av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));
  1443. s->core_ext_mask |= DCA_EXT_X96;
  1444. break;
  1445. }
  1446. }
  1447. skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
  1448. }
  1449. } else {
  1450. /* no supported extensions, skip the rest of the core substream */
  1451. skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));
  1452. }
  1453. if (s->core_ext_mask & DCA_EXT_X96)
  1454. s->profile = FF_PROFILE_DTS_96_24;
  1455. else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))
  1456. s->profile = FF_PROFILE_DTS_ES;
  1457. /* check for ExSS (HD part) */
  1458. if (s->dca_buffer_size - s->frame_size > 32 &&
  1459. get_bits_long(&s->gb, 32) == DCA_HD_MARKER)
  1460. ff_dca_exss_parse_header(s);
  1461. avctx->profile = s->profile;
  1462. full_channels = channels = s->prim_channels + !!s->lfe;
  1463. /* If we have XXCH then the channel layout is managed differently */
  1464. /* note that XLL will also have another way to do things */
  1465. if (!(s->core_ext_mask & DCA_EXT_XXCH)
  1466. || (s->core_ext_mask & DCA_EXT_XXCH && avctx->request_channels > 0
  1467. && avctx->request_channels
  1468. < num_core_channels + !!s->lfe + s->xxch_chset_nch[0]))
  1469. { /* xxx should also do MA extensions */
  1470. if (s->amode < 16) {
  1471. avctx->channel_layout = dca_core_channel_layout[s->amode];
  1472. if (s->prim_channels + !!s->lfe > 2 &&
  1473. avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
  1474. /*
  1475. * Neither the core's auxiliary data nor our default tables contain
  1476. * downmix coefficients for the additional channel coded in the XCh
  1477. * extension, so when we're doing a Stereo downmix, don't decode it.
  1478. */
  1479. s->xch_disable = 1;
  1480. }
  1481. #if FF_API_REQUEST_CHANNELS
  1482. FF_DISABLE_DEPRECATION_WARNINGS
  1483. if (s->xch_present && !s->xch_disable &&
  1484. (!avctx->request_channels ||
  1485. avctx->request_channels > num_core_channels + !!s->lfe)) {
  1486. FF_ENABLE_DEPRECATION_WARNINGS
  1487. #else
  1488. if (s->xch_present && !s->xch_disable) {
  1489. #endif
  1490. if (avctx->channel_layout & AV_CH_BACK_CENTER) {
  1491. avpriv_request_sample(avctx, "XCh with Back center channel");
  1492. return AVERROR_INVALIDDATA;
  1493. }
  1494. avctx->channel_layout |= AV_CH_BACK_CENTER;
  1495. if (s->lfe) {
  1496. avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
  1497. s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode];
  1498. } else {
  1499. s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];
  1500. }
  1501. if (s->channel_order_tab[s->xch_base_channel] < 0)
  1502. return AVERROR_INVALIDDATA;
  1503. } else {
  1504. channels = num_core_channels + !!s->lfe;
  1505. s->xch_present = 0; /* disable further xch processing */
  1506. if (s->lfe) {
  1507. avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
  1508. s->channel_order_tab = dca_channel_reorder_lfe[s->amode];
  1509. } else
  1510. s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];
  1511. }
  1512. if (channels > !!s->lfe &&
  1513. s->channel_order_tab[channels - 1 - !!s->lfe] < 0)
  1514. return AVERROR_INVALIDDATA;
  1515. if (av_get_channel_layout_nb_channels(avctx->channel_layout) != channels) {
  1516. av_log(avctx, AV_LOG_ERROR, "Number of channels %d mismatches layout %d\n", channels, av_get_channel_layout_nb_channels(avctx->channel_layout));
  1517. return AVERROR_INVALIDDATA;
  1518. }
  1519. if (num_core_channels + !!s->lfe > 2 &&
  1520. avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
  1521. channels = 2;
  1522. s->output = s->prim_channels == 2 ? s->amode : DCA_STEREO;
  1523. avctx->channel_layout = AV_CH_LAYOUT_STEREO;
  1524. }
  1525. else if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) {
  1526. static const int8_t dca_channel_order_native[9] = { 0, 1, 2, 3, 4, 5, 6, 7, 8 };
  1527. s->channel_order_tab = dca_channel_order_native;
  1528. }
  1529. s->lfe_index = dca_lfe_index[s->amode];
  1530. } else {
  1531. av_log(avctx, AV_LOG_ERROR,
  1532. "Non standard configuration %d !\n", s->amode);
  1533. return AVERROR_INVALIDDATA;
  1534. }
  1535. s->xxch_dmix_embedded = 0;
  1536. } else {
  1537. /* we only get here if an XXCH channel set can be added to the mix */
  1538. channel_mask = s->xxch_core_spkmask;
  1539. if (avctx->request_channels > 0
  1540. && avctx->request_channels < s->prim_channels) {
  1541. channels = num_core_channels + !!s->lfe;
  1542. for (i = 0; i < s->xxch_chset && channels + s->xxch_chset_nch[i]
  1543. <= avctx->request_channels; i++) {
  1544. channels += s->xxch_chset_nch[i];
  1545. channel_mask |= s->xxch_spk_masks[i];
  1546. }
  1547. } else {
  1548. channels = s->prim_channels + !!s->lfe;
  1549. for (i = 0; i < s->xxch_chset; i++) {
  1550. channel_mask |= s->xxch_spk_masks[i];
  1551. }
  1552. }
  1553. /* Given the DTS spec'ed channel mask, generate an avcodec version */
  1554. channel_layout = 0;
  1555. for (i = 0; i < s->xxch_nbits_spk_mask; ++i) {
  1556. if (channel_mask & (1 << i)) {
  1557. channel_layout |= map_xxch_to_native[i];
  1558. }
  1559. }
  1560. /* make sure that we have managed to get equivalent dts/avcodec channel
  1561. * masks in some sense -- unfortunately some channels could overlap */
  1562. if (av_popcount(channel_mask) != av_popcount(channel_layout)) {
  1563. av_log(avctx, AV_LOG_DEBUG,
  1564. "DTS-XXCH: Inconsistent avcodec/dts channel layouts\n");
  1565. return AVERROR_INVALIDDATA;
  1566. }
  1567. avctx->channel_layout = channel_layout;
  1568. if (!(avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE)) {
  1569. /* Estimate DTS --> avcodec ordering table */
  1570. for (chset = -1, j = 0; chset < s->xxch_chset; ++chset) {
  1571. mask = chset >= 0 ? s->xxch_spk_masks[chset]
  1572. : s->xxch_core_spkmask;
  1573. for (i = 0; i < s->xxch_nbits_spk_mask; i++) {
  1574. if (mask & ~(DCA_XXCH_LFE1 | DCA_XXCH_LFE2) & (1 << i)) {
  1575. lavc = map_xxch_to_native[i];
  1576. posn = av_popcount(channel_layout & (lavc - 1));
  1577. s->xxch_order_tab[j++] = posn;
  1578. }
  1579. }
  1580. }
  1581. s->lfe_index = av_popcount(channel_layout & (AV_CH_LOW_FREQUENCY-1));
  1582. } else { /* native ordering */
  1583. for (i = 0; i < channels; i++)
  1584. s->xxch_order_tab[i] = i;
  1585. s->lfe_index = channels - 1;
  1586. }
  1587. s->channel_order_tab = s->xxch_order_tab;
  1588. }
  1589. if (avctx->channels != channels) {
  1590. if (avctx->channels)
  1591. av_log(avctx, AV_LOG_INFO, "Number of channels changed in DCA decoder (%d -> %d)\n", avctx->channels, channels);
  1592. avctx->channels = channels;
  1593. }
  1594. /* get output buffer */
  1595. frame->nb_samples = 256 * (s->sample_blocks / 8);
  1596. if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
  1597. return ret;
  1598. samples_flt = (float **) frame->extended_data;
  1599. /* allocate buffer for extra channels if downmixing */
  1600. if (avctx->channels < full_channels) {
  1601. ret = av_samples_get_buffer_size(NULL, full_channels - channels,
  1602. frame->nb_samples,
  1603. avctx->sample_fmt, 0);
  1604. if (ret < 0)
  1605. return ret;
  1606. av_fast_malloc(&s->extra_channels_buffer,
  1607. &s->extra_channels_buffer_size, ret);
  1608. if (!s->extra_channels_buffer)
  1609. return AVERROR(ENOMEM);
  1610. ret = av_samples_fill_arrays((uint8_t **) s->extra_channels, NULL,
  1611. s->extra_channels_buffer,
  1612. full_channels - channels,
  1613. frame->nb_samples, avctx->sample_fmt, 0);
  1614. if (ret < 0)
  1615. return ret;
  1616. }
  1617. /* filter to get final output */
  1618. for (i = 0; i < (s->sample_blocks / 8); i++) {
  1619. int ch;
  1620. for (ch = 0; ch < channels; ch++)
  1621. s->samples_chanptr[ch] = samples_flt[ch] + i * 256;
  1622. for (; ch < full_channels; ch++)
  1623. s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * 256;
  1624. dca_filter_channels(s, i);
  1625. /* If this was marked as a DTS-ES stream we need to subtract back- */
  1626. /* channel from SL & SR to remove matrixed back-channel signal */
  1627. if ((s->source_pcm_res & 1) && s->xch_present) {
  1628. float *back_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel]];
  1629. float *lt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 2]];
  1630. float *rt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 1]];
  1631. s->fdsp->vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);
  1632. s->fdsp->vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);
  1633. }
  1634. /* If stream contains XXCH, we might need to undo an embedded downmix */
  1635. if (s->xxch_dmix_embedded) {
  1636. /* Loop over channel sets in turn */
  1637. ch = num_core_channels;
  1638. for (chset = 0; chset < s->xxch_chset; chset++) {
  1639. endch = ch + s->xxch_chset_nch[chset];
  1640. mask = s->xxch_dmix_embedded;
  1641. /* undo downmix */
  1642. for (j = ch; j < endch; j++) {
  1643. if (mask & (1 << j)) { /* this channel has been mixed-out */
  1644. src_chan = s->samples_chanptr[s->channel_order_tab[j]];
  1645. for (k = 0; k < endch; k++) {
  1646. achan = s->channel_order_tab[k];
  1647. scale = s->xxch_dmix_coeff[j][k];
  1648. if (scale != 0.0) {
  1649. dst_chan = s->samples_chanptr[achan];
  1650. s->fdsp->vector_fmac_scalar(dst_chan, src_chan,
  1651. -scale, 256);
  1652. }
  1653. }
  1654. }
  1655. }
  1656. /* if a downmix has been embedded then undo the pre-scaling */
  1657. if ((mask & (1 << ch)) && s->xxch_dmix_sf[chset] != 1.0f) {
  1658. scale = s->xxch_dmix_sf[chset];
  1659. for (j = 0; j < ch; j++) {
  1660. src_chan = s->samples_chanptr[s->channel_order_tab[j]];
  1661. for (k = 0; k < 256; k++)
  1662. src_chan[k] *= scale;
  1663. }
  1664. /* LFE channel is always part of core, scale if it exists */
  1665. if (s->lfe) {
  1666. src_chan = s->samples_chanptr[s->lfe_index];
  1667. for (k = 0; k < 256; k++)
  1668. src_chan[k] *= scale;
  1669. }
  1670. }
  1671. ch = endch;
  1672. }
  1673. }
  1674. }
  1675. /* update lfe history */
  1676. lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);
  1677. for (i = 0; i < 2 * s->lfe * 4; i++)
  1678. s->lfe_data[i] = s->lfe_data[i + lfe_samples];
  1679. /* AVMatrixEncoding
  1680. *
  1681. * DCA_STEREO_TOTAL (Lt/Rt) is equivalent to Dolby Surround */
  1682. ret = ff_side_data_update_matrix_encoding(frame,
  1683. (s->output & ~DCA_LFE) == DCA_STEREO_TOTAL ?
  1684. AV_MATRIX_ENCODING_DOLBY : AV_MATRIX_ENCODING_NONE);
  1685. if (ret < 0)
  1686. return ret;
  1687. *got_frame_ptr = 1;
  1688. return buf_size;
  1689. }
  1690. /**
  1691. * DCA initialization
  1692. *
  1693. * @param avctx pointer to the AVCodecContext
  1694. */
  1695. static av_cold int dca_decode_init(AVCodecContext *avctx)
  1696. {
  1697. DCAContext *s = avctx->priv_data;
  1698. s->avctx = avctx;
  1699. dca_init_vlcs();
  1700. s->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);
  1701. if (!s->fdsp)
  1702. return AVERROR(ENOMEM);
  1703. ff_mdct_init(&s->imdct, 6, 1, 1.0);
  1704. ff_synth_filter_init(&s->synth);
  1705. ff_dcadsp_init(&s->dcadsp);
  1706. ff_fmt_convert_init(&s->fmt_conv, avctx);
  1707. avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
  1708. /* allow downmixing to stereo */
  1709. #if FF_API_REQUEST_CHANNELS
  1710. FF_DISABLE_DEPRECATION_WARNINGS
  1711. if (avctx->request_channels == 2)
  1712. avctx->request_channel_layout = AV_CH_LAYOUT_STEREO;
  1713. FF_ENABLE_DEPRECATION_WARNINGS
  1714. #endif
  1715. if (avctx->channels > 2 &&
  1716. avctx->request_channel_layout == AV_CH_LAYOUT_STEREO)
  1717. avctx->channels = 2;
  1718. return 0;
  1719. }
  1720. static av_cold int dca_decode_end(AVCodecContext *avctx)
  1721. {
  1722. DCAContext *s = avctx->priv_data;
  1723. ff_mdct_end(&s->imdct);
  1724. av_freep(&s->extra_channels_buffer);
  1725. av_freep(&s->fdsp);
  1726. return 0;
  1727. }
  1728. static const AVProfile profiles[] = {
  1729. { FF_PROFILE_DTS, "DTS" },
  1730. { FF_PROFILE_DTS_ES, "DTS-ES" },
  1731. { FF_PROFILE_DTS_96_24, "DTS 96/24" },
  1732. { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },
  1733. { FF_PROFILE_DTS_HD_MA, "DTS-HD MA" },
  1734. { FF_PROFILE_UNKNOWN },
  1735. };
  1736. static const AVOption options[] = {
  1737. { "disable_xch", "disable decoding of the XCh extension", offsetof(DCAContext, xch_disable), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },
  1738. { NULL },
  1739. };
  1740. static const AVClass dca_decoder_class = {
  1741. .class_name = "DCA decoder",
  1742. .item_name = av_default_item_name,
  1743. .option = options,
  1744. .version = LIBAVUTIL_VERSION_INT,
  1745. .category = AV_CLASS_CATEGORY_DECODER,
  1746. };
  1747. AVCodec ff_dca_decoder = {
  1748. .name = "dca",
  1749. .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
  1750. .type = AVMEDIA_TYPE_AUDIO,
  1751. .id = AV_CODEC_ID_DTS,
  1752. .priv_data_size = sizeof(DCAContext),
  1753. .init = dca_decode_init,
  1754. .decode = dca_decode_frame,
  1755. .close = dca_decode_end,
  1756. .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
  1757. .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
  1758. AV_SAMPLE_FMT_NONE },
  1759. .profiles = NULL_IF_CONFIG_SMALL(profiles),
  1760. .priv_class = &dca_decoder_class,
  1761. };