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FFmpeg/libavcodec/dcadec.c

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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 Libav.

  9.  *

  10.  * Libav 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.  * Libav 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 Libav; if not, write to the Free Software

  22.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

  23.  */

  24. 

  25. #include <math.h>

  26. #include <stddef.h>

  27. #include <stdio.h>

  28. 

  29. #include "libavutil/channel_layout.h"

  30. #include "libavutil/common.h"

  31. #include "libavutil/float_dsp.h"

  32. #include "libavutil/internal.h"

  33. #include "libavutil/intreadwrite.h"

  34. #include "libavutil/mathematics.h"

  35. #include "libavutil/opt.h"

  36. #include "libavutil/samplefmt.h"

  37. 

  38. #include "avcodec.h"

  39. #include "dca.h"

  40. #include "dcadata.h"

  41. #include "dcadsp.h"

  42. #include "dcahuff.h"

  43. #include "dca_exss.h"

  44. #include "fft.h"

  45. #include "fmtconvert.h"

  46. #include "get_bits.h"

  47. #include "internal.h"

  48. #include "mathops.h"

  49. #include "put_bits.h"

  50. #include "synth_filter.h"

  51. 

  52. #if ARCH_ARM

  53. #   include "arm/dca.h"

  54. #endif

  55. 

  56. //#define TRACE

  57. 

  58. enum DCAMode {

  59.     DCA_MONO = 0,

  60.     DCA_CHANNEL,

  61.     DCA_STEREO,

  62.     DCA_STEREO_SUMDIFF,

  63.     DCA_STEREO_TOTAL,

  64.     DCA_3F,

  65.     DCA_2F1R,

  66.     DCA_3F1R,

  67.     DCA_2F2R,

  68.     DCA_3F2R,

  69.     DCA_4F2R

  70. };

  71. 

  72. /* -1 are reserved or unknown */

  73. static const int dca_ext_audio_descr_mask[] = {

  74.     DCA_EXT_XCH,

  75.     -1,

  76.     DCA_EXT_X96,

  77.     DCA_EXT_XCH | DCA_EXT_X96,

  78.     -1,

  79.     -1,

  80.     DCA_EXT_XXCH,

  81.     -1,

  82. };

  83. 

  84. /* Tables for mapping dts channel configurations to libavcodec multichannel api.

  85.  * Some compromises have been made for special configurations. Most configurations

  86.  * are never used so complete accuracy is not needed.

  87.  *

  88.  * L = left, R = right, C = center, S = surround, F = front, R = rear, T = total, OV = overhead.

  89.  * S  -> side, when both rear and back are configured move one of them to the side channel

  90.  * OV -> center back

  91.  * All 2 channel configurations -> AV_CH_LAYOUT_STEREO

  92.  */

  93. static const uint64_t dca_core_channel_layout[] = {

  94.     AV_CH_FRONT_CENTER,                                                     ///< 1, A

  95.     AV_CH_LAYOUT_STEREO,                                                    ///< 2, A + B (dual mono)

  96.     AV_CH_LAYOUT_STEREO,                                                    ///< 2, L + R (stereo)

  97.     AV_CH_LAYOUT_STEREO,                                                    ///< 2, (L + R) + (L - R) (sum-difference)

  98.     AV_CH_LAYOUT_STEREO,                                                    ///< 2, LT + RT (left and right total)

  99.     AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER,                               ///< 3, C + L + R

  100.     AV_CH_LAYOUT_STEREO | AV_CH_BACK_CENTER,                                ///< 3, L + R + S

  101.     AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER,           ///< 4, C + L + R + S

  102.     AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT,               ///< 4, L + R + SL + SR

  103. 

  104.     AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_SIDE_LEFT |

  105.     AV_CH_SIDE_RIGHT,                                                       ///< 5, C + L + R + SL + SR

  106. 

  107.     AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |

  108.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER,               ///< 6, CL + CR + L + R + SL + SR

  109. 

  110.     AV_CH_LAYOUT_STEREO | AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT |

  111.     AV_CH_FRONT_CENTER  | AV_CH_BACK_CENTER,                                ///< 6, C + L + R + LR + RR + OV

  112. 

  113.     AV_CH_FRONT_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |

  114.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_BACK_CENTER   |

  115.     AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT,                                     ///< 6, CF + CR + LF + RF + LR + RR

  116. 

  117.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER   |

  118.     AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |

  119.     AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT,                                     ///< 7, CL + C + CR + L + R + SL + SR

  120. 

  121.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |

  122.     AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |

  123.     AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT,                                     ///< 8, CL + CR + L + R + SL1 + SL2 + SR1 + SR2

  124. 

  125.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER   |

  126.     AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |

  127.     AV_CH_SIDE_LEFT | AV_CH_BACK_CENTER | AV_CH_SIDE_RIGHT,                 ///< 8, CL + C + CR + L + R + SL + S + SR

  128. };

  129. 

  130. static const int8_t dca_lfe_index[] = {

  131.     1, 2, 2, 2, 2, 3, 2, 3, 2, 3, 2, 3, 1, 3, 2, 3

  132. };

  133. 

  134. static const int8_t dca_channel_reorder_lfe[][9] = {

  135.     { 0, -1, -1, -1, -1, -1, -1, -1, -1 },

  136.     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },

  137.     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },

  138.     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },

  139.     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },

  140.     { 2,  0,  1, -1, -1, -1, -1, -1, -1 },

  141.     { 0,  1,  3, -1, -1, -1, -1, -1, -1 },

  142.     { 2,  0,  1,  4, -1, -1, -1, -1, -1 },

  143.     { 0,  1,  3,  4, -1, -1, -1, -1, -1 },

  144.     { 2,  0,  1,  4,  5, -1, -1, -1, -1 },

  145.     { 3,  4,  0,  1,  5,  6, -1, -1, -1 },

  146.     { 2,  0,  1,  4,  5,  6, -1, -1, -1 },

  147.     { 0,  6,  4,  5,  2,  3, -1, -1, -1 },

  148.     { 4,  2,  5,  0,  1,  6,  7, -1, -1 },

  149.     { 5,  6,  0,  1,  7,  3,  8,  4, -1 },

  150.     { 4,  2,  5,  0,  1,  6,  8,  7, -1 },

  151. };

  152. 

  153. static const int8_t dca_channel_reorder_lfe_xch[][9] = {

  154.     { 0,  2, -1, -1, -1, -1, -1, -1, -1 },

  155.     { 0,  1,  3, -1, -1, -1, -1, -1, -1 },

  156.     { 0,  1,  3, -1, -1, -1, -1, -1, -1 },

  157.     { 0,  1,  3, -1, -1, -1, -1, -1, -1 },

  158.     { 0,  1,  3, -1, -1, -1, -1, -1, -1 },

  159.     { 2,  0,  1,  4, -1, -1, -1, -1, -1 },

  160.     { 0,  1,  3,  4, -1, -1, -1, -1, -1 },

  161.     { 2,  0,  1,  4,  5, -1, -1, -1, -1 },

  162.     { 0,  1,  4,  5,  3, -1, -1, -1, -1 },

  163.     { 2,  0,  1,  5,  6,  4, -1, -1, -1 },

  164.     { 3,  4,  0,  1,  6,  7,  5, -1, -1 },

  165.     { 2,  0,  1,  4,  5,  6,  7, -1, -1 },

  166.     { 0,  6,  4,  5,  2,  3,  7, -1, -1 },

  167.     { 4,  2,  5,  0,  1,  7,  8,  6, -1 },

  168.     { 5,  6,  0,  1,  8,  3,  9,  4,  7 },

  169.     { 4,  2,  5,  0,  1,  6,  9,  8,  7 },

  170. };

  171. 

  172. static const int8_t dca_channel_reorder_nolfe[][9] = {

  173.     { 0, -1, -1, -1, -1, -1, -1, -1, -1 },

  174.     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },

  175.     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },

  176.     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },

  177.     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },

  178.     { 2,  0,  1, -1, -1, -1, -1, -1, -1 },

  179.     { 0,  1,  2, -1, -1, -1, -1, -1, -1 },

  180.     { 2,  0,  1,  3, -1, -1, -1, -1, -1 },

  181.     { 0,  1,  2,  3, -1, -1, -1, -1, -1 },

  182.     { 2,  0,  1,  3,  4, -1, -1, -1, -1 },

  183.     { 2,  3,  0,  1,  4,  5, -1, -1, -1 },

  184.     { 2,  0,  1,  3,  4,  5, -1, -1, -1 },

  185.     { 0,  5,  3,  4,  1,  2, -1, -1, -1 },

  186.     { 3,  2,  4,  0,  1,  5,  6, -1, -1 },

  187.     { 4,  5,  0,  1,  6,  2,  7,  3, -1 },

  188.     { 3,  2,  4,  0,  1,  5,  7,  6, -1 },

  189. };

  190. 

  191. static const int8_t dca_channel_reorder_nolfe_xch[][9] = {

  192.     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },

  193.     { 0,  1,  2, -1, -1, -1, -1, -1, -1 },

  194.     { 0,  1,  2, -1, -1, -1, -1, -1, -1 },

  195.     { 0,  1,  2, -1, -1, -1, -1, -1, -1 },

  196.     { 0,  1,  2, -1, -1, -1, -1, -1, -1 },

  197.     { 2,  0,  1,  3, -1, -1, -1, -1, -1 },

  198.     { 0,  1,  2,  3, -1, -1, -1, -1, -1 },

  199.     { 2,  0,  1,  3,  4, -1, -1, -1, -1 },

  200.     { 0,  1,  3,  4,  2, -1, -1, -1, -1 },

  201.     { 2,  0,  1,  4,  5,  3, -1, -1, -1 },

  202.     { 2,  3,  0,  1,  5,  6,  4, -1, -1 },

  203.     { 2,  0,  1,  3,  4,  5,  6, -1, -1 },

  204.     { 0,  5,  3,  4,  1,  2,  6, -1, -1 },

  205.     { 3,  2,  4,  0,  1,  6,  7,  5, -1 },

  206.     { 4,  5,  0,  1,  7,  2,  8,  3,  6 },

  207.     { 3,  2,  4,  0,  1,  5,  8,  7,  6 },

  208. };

  209. 

  210. #define DCA_DOLBY                  101           /* FIXME */

  211. 

  212. #define DCA_CHANNEL_BITS             6

  213. #define DCA_CHANNEL_MASK          0x3F

  214. 

  215. #define DCA_LFE                   0x80

  216. 

  217. #define HEADER_SIZE                 14

  218. 

  219. #define DCA_NSYNCAUX        0x9A1105A0

  220. 

  221. /** Bit allocation */

  222. typedef struct BitAlloc {

  223.     int offset;                 ///< code values offset

  224.     int maxbits[8];             ///< max bits in VLC

  225.     int wrap;                   ///< wrap for get_vlc2()

  226.     VLC vlc[8];                 ///< actual codes

  227. } BitAlloc;

  228. 

  229. static BitAlloc dca_bitalloc_index;    ///< indexes for samples VLC select

  230. static BitAlloc dca_tmode;             ///< transition mode VLCs

  231. static BitAlloc dca_scalefactor;       ///< scalefactor VLCs

  232. static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs

  233. 

  234. static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba,

  235.                                          int idx)

  236. {

  237.     return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) +

  238.            ba->offset;

  239. }

  240. 

  241. static const uint16_t dca_vlc_offs[] = {

  242.         0,   512,   640,   768,  1282,  1794,  2436,  3080,  3770,  4454,  5364,

  243.      5372,  5380,  5388,  5392,  5396,  5412,  5420,  5428,  5460,  5492,  5508,

  244.      5572,  5604,  5668,  5796,  5860,  5892,  6412,  6668,  6796,  7308,  7564,

  245.      7820,  8076,  8620,  9132,  9388,  9910, 10166, 10680, 11196, 11726, 12240,

  246.     12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264,

  247.     18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622,

  248. };

  249. 

  250. static av_cold void dca_init_vlcs(void)

  251. {

  252.     static int vlcs_initialized = 0;

  253.     int i, j, c = 14;

  254.     static VLC_TYPE dca_table[23622][2];

  255. 

  256.     if (vlcs_initialized)

  257.         return;

  258. 

  259.     dca_bitalloc_index.offset = 1;

  260.     dca_bitalloc_index.wrap   = 2;

  261.     for (i = 0; i < 5; i++) {

  262.         dca_bitalloc_index.vlc[i].table           = &dca_table[dca_vlc_offs[i]];

  263.         dca_bitalloc_index.vlc[i].table_allocated = dca_vlc_offs[i + 1] - dca_vlc_offs[i];

  264.         init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,

  265.                  bitalloc_12_bits[i], 1, 1,

  266.                  bitalloc_12_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);

  267.     }

  268.     dca_scalefactor.offset = -64;

  269.     dca_scalefactor.wrap   = 2;

  270.     for (i = 0; i < 5; i++) {

  271.         dca_scalefactor.vlc[i].table           = &dca_table[dca_vlc_offs[i + 5]];

  272.         dca_scalefactor.vlc[i].table_allocated = dca_vlc_offs[i + 6] - dca_vlc_offs[i + 5];

  273.         init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,

  274.                  scales_bits[i], 1, 1,

  275.                  scales_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);

  276.     }

  277.     dca_tmode.offset = 0;

  278.     dca_tmode.wrap   = 1;

  279.     for (i = 0; i < 4; i++) {

  280.         dca_tmode.vlc[i].table           = &dca_table[dca_vlc_offs[i + 10]];

  281.         dca_tmode.vlc[i].table_allocated = dca_vlc_offs[i + 11] - dca_vlc_offs[i + 10];

  282.         init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,

  283.                  tmode_bits[i], 1, 1,

  284.                  tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);

  285.     }

  286. 

  287.     for (i = 0; i < 10; i++)

  288.         for (j = 0; j < 7; j++) {

  289.             if (!bitalloc_codes[i][j])

  290.                 break;

  291.             dca_smpl_bitalloc[i + 1].offset                 = bitalloc_offsets[i];

  292.             dca_smpl_bitalloc[i + 1].wrap                   = 1 + (j > 4);

  293.             dca_smpl_bitalloc[i + 1].vlc[j].table           = &dca_table[dca_vlc_offs[c]];

  294.             dca_smpl_bitalloc[i + 1].vlc[j].table_allocated = dca_vlc_offs[c + 1] - dca_vlc_offs[c];

  295. 

  296.             init_vlc(&dca_smpl_bitalloc[i + 1].vlc[j], bitalloc_maxbits[i][j],

  297.                      bitalloc_sizes[i],

  298.                      bitalloc_bits[i][j], 1, 1,

  299.                      bitalloc_codes[i][j], 2, 2, INIT_VLC_USE_NEW_STATIC);

  300.             c++;

  301.         }

  302.     vlcs_initialized = 1;

  303. }

  304. 

  305. static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)

  306. {

  307.     while (len--)

  308.         *dst++ = get_bits(gb, bits);

  309. }

  310. 

  311. static int dca_parse_audio_coding_header(DCAContext *s, int base_channel)

  312. {

  313.     int i, j;

  314.     static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };

  315.     static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };

  316.     static const int thr[11]    = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };

  317. 

  318.     s->total_channels = get_bits(&s->gb, 3) + 1 + base_channel;

  319.     s->prim_channels  = s->total_channels;

  320. 

  321.     if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)

  322.         s->prim_channels = DCA_PRIM_CHANNELS_MAX;

  323. 

  324.     for (i = base_channel; i < s->prim_channels; i++) {

  325.         s->subband_activity[i] = get_bits(&s->gb, 5) + 2;

  326.         if (s->subband_activity[i] > DCA_SUBBANDS)

  327.             s->subband_activity[i] = DCA_SUBBANDS;

  328.     }

  329.     for (i = base_channel; i < s->prim_channels; i++) {

  330.         s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;

  331.         if (s->vq_start_subband[i] > DCA_SUBBANDS)

  332.             s->vq_start_subband[i] = DCA_SUBBANDS;

  333.     }

  334.     get_array(&s->gb, s->joint_intensity + base_channel,     s->prim_channels - base_channel, 3);

  335.     get_array(&s->gb, s->transient_huffman + base_channel,   s->prim_channels - base_channel, 2);

  336.     get_array(&s->gb, s->scalefactor_huffman + base_channel, s->prim_channels - base_channel, 3);

  337.     get_array(&s->gb, s->bitalloc_huffman + base_channel,    s->prim_channels - base_channel, 3);

  338. 

  339.     /* Get codebooks quantization indexes */

  340.     if (!base_channel)

  341.         memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));

  342.     for (j = 1; j < 11; j++)

  343.         for (i = base_channel; i < s->prim_channels; i++)

  344.             s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);

  345. 

  346.     /* Get scale factor adjustment */

  347.     for (j = 0; j < 11; j++)

  348.         for (i = base_channel; i < s->prim_channels; i++)

  349.             s->scalefactor_adj[i][j] = 1;

  350. 

  351.     for (j = 1; j < 11; j++)

  352.         for (i = base_channel; i < s->prim_channels; i++)

  353.             if (s->quant_index_huffman[i][j] < thr[j])

  354.                 s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];

  355. 

  356.     if (s->crc_present) {

  357.         /* Audio header CRC check */

  358.         get_bits(&s->gb, 16);

  359.     }

  360. 

  361.     s->current_subframe    = 0;

  362.     s->current_subsubframe = 0;

  363. 

  364. #ifdef TRACE

  365.     av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes);

  366.     av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels);

  367.     for (i = base_channel; i < s->prim_channels; i++) {

  368.         av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n",

  369.                s->subband_activity[i]);

  370.         av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n",

  371.                s->vq_start_subband[i]);

  372.         av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n",

  373.                s->joint_intensity[i]);

  374.         av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n",

  375.                s->transient_huffman[i]);

  376.         av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n",

  377.                s->scalefactor_huffman[i]);

  378.         av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n",

  379.                s->bitalloc_huffman[i]);

  380.         av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:");

  381.         for (j = 0; j < 11; j++)

  382.             av_log(s->avctx, AV_LOG_DEBUG, " %i", s->quant_index_huffman[i][j]);

  383.         av_log(s->avctx, AV_LOG_DEBUG, "\n");

  384.         av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");

  385.         for (j = 0; j < 11; j++)

  386.             av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]);

  387.         av_log(s->avctx, AV_LOG_DEBUG, "\n");

  388.     }

  389. #endif

  390. 

  391.     return 0;

  392. }

  393. 

  394. static int dca_parse_frame_header(DCAContext *s)

  395. {

  396.     init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);

  397. 

  398.     /* Sync code */

  399.     skip_bits_long(&s->gb, 32);

  400. 

  401.     /* Frame header */

  402.     s->frame_type        = get_bits(&s->gb, 1);

  403.     s->samples_deficit   = get_bits(&s->gb, 5) + 1;

  404.     s->crc_present       = get_bits(&s->gb, 1);

  405.     s->sample_blocks     = get_bits(&s->gb, 7) + 1;

  406.     s->frame_size        = get_bits(&s->gb, 14) + 1;

  407.     if (s->frame_size < 95)

  408.         return AVERROR_INVALIDDATA;

  409.     s->amode             = get_bits(&s->gb, 6);

  410.     s->sample_rate       = avpriv_dca_sample_rates[get_bits(&s->gb, 4)];

  411.     if (!s->sample_rate)

  412.         return AVERROR_INVALIDDATA;

  413.     s->bit_rate_index    = get_bits(&s->gb, 5);

  414.     s->bit_rate          = dca_bit_rates[s->bit_rate_index];

  415.     if (!s->bit_rate)

  416.         return AVERROR_INVALIDDATA;

  417. 

  418.     skip_bits1(&s->gb); // always 0 (reserved, cf. ETSI TS 102 114 V1.4.1)

  419.     s->dynrange          = get_bits(&s->gb, 1);

  420.     s->timestamp         = get_bits(&s->gb, 1);

  421.     s->aux_data          = get_bits(&s->gb, 1);

  422.     s->hdcd              = get_bits(&s->gb, 1);

  423.     s->ext_descr         = get_bits(&s->gb, 3);

  424.     s->ext_coding        = get_bits(&s->gb, 1);

  425.     s->aspf              = get_bits(&s->gb, 1);

  426.     s->lfe               = get_bits(&s->gb, 2);

  427.     s->predictor_history = get_bits(&s->gb, 1);

  428. 

  429.     if (s->lfe > 2) {

  430.         av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE value: %d\n", s->lfe);

  431.         return AVERROR_INVALIDDATA;

  432.     }

  433. 

  434.     /* TODO: check CRC */

  435.     if (s->crc_present)

  436.         s->header_crc    = get_bits(&s->gb, 16);

  437. 

  438.     s->multirate_inter   = get_bits(&s->gb, 1);

  439.     s->version           = get_bits(&s->gb, 4);

  440.     s->copy_history      = get_bits(&s->gb, 2);

  441.     s->source_pcm_res    = get_bits(&s->gb, 3);

  442.     s->front_sum         = get_bits(&s->gb, 1);

  443.     s->surround_sum      = get_bits(&s->gb, 1);

  444.     s->dialog_norm       = get_bits(&s->gb, 4);

  445. 

  446.     /* FIXME: channels mixing levels */

  447.     s->output = s->amode;

  448.     if (s->lfe)

  449.         s->output |= DCA_LFE;

  450. 

  451. #ifdef TRACE

  452.     av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type);

  453.     av_log(s->avctx, AV_LOG_DEBUG, "samples deficit: %i\n", s->samples_deficit);

  454.     av_log(s->avctx, AV_LOG_DEBUG, "crc present: %i\n", s->crc_present);

  455.     av_log(s->avctx, AV_LOG_DEBUG, "sample blocks: %i (%i samples)\n",

  456.            s->sample_blocks, s->sample_blocks * 32);

  457.     av_log(s->avctx, AV_LOG_DEBUG, "frame size: %i bytes\n", s->frame_size);

  458.     av_log(s->avctx, AV_LOG_DEBUG, "amode: %i (%i channels)\n",

  459.            s->amode, dca_channels[s->amode]);

  460.     av_log(s->avctx, AV_LOG_DEBUG, "sample rate: %i Hz\n",

  461.            s->sample_rate);

  462.     av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i bits/s\n",

  463.            s->bit_rate);

  464.     av_log(s->avctx, AV_LOG_DEBUG, "dynrange: %i\n", s->dynrange);

  465.     av_log(s->avctx, AV_LOG_DEBUG, "timestamp: %i\n", s->timestamp);

  466.     av_log(s->avctx, AV_LOG_DEBUG, "aux_data: %i\n", s->aux_data);

  467.     av_log(s->avctx, AV_LOG_DEBUG, "hdcd: %i\n", s->hdcd);

  468.     av_log(s->avctx, AV_LOG_DEBUG, "ext descr: %i\n", s->ext_descr);

  469.     av_log(s->avctx, AV_LOG_DEBUG, "ext coding: %i\n", s->ext_coding);

  470.     av_log(s->avctx, AV_LOG_DEBUG, "aspf: %i\n", s->aspf);

  471.     av_log(s->avctx, AV_LOG_DEBUG, "lfe: %i\n", s->lfe);

  472.     av_log(s->avctx, AV_LOG_DEBUG, "predictor history: %i\n",

  473.            s->predictor_history);

  474.     av_log(s->avctx, AV_LOG_DEBUG, "header crc: %i\n", s->header_crc);

  475.     av_log(s->avctx, AV_LOG_DEBUG, "multirate inter: %i\n",

  476.            s->multirate_inter);

  477.     av_log(s->avctx, AV_LOG_DEBUG, "version number: %i\n", s->version);

  478.     av_log(s->avctx, AV_LOG_DEBUG, "copy history: %i\n", s->copy_history);

  479.     av_log(s->avctx, AV_LOG_DEBUG,

  480.            "source pcm resolution: %i (%i bits/sample)\n",

  481.            s->source_pcm_res, dca_bits_per_sample[s->source_pcm_res]);

  482.     av_log(s->avctx, AV_LOG_DEBUG, "front sum: %i\n", s->front_sum);

  483.     av_log(s->avctx, AV_LOG_DEBUG, "surround sum: %i\n", s->surround_sum);

  484.     av_log(s->avctx, AV_LOG_DEBUG, "dialog norm: %i\n", s->dialog_norm);

  485.     av_log(s->avctx, AV_LOG_DEBUG, "\n");

  486. #endif

  487. 

  488.     /* Primary audio coding header */

  489.     s->subframes = get_bits(&s->gb, 4) + 1;

  490. 

  491.     return dca_parse_audio_coding_header(s, 0);

  492. }

  493. 

  494. static inline int get_scale(GetBitContext *gb, int level, int value, int log2range)

  495. {

  496.     if (level < 5) {

  497.         /* huffman encoded */

  498.         value += get_bitalloc(gb, &dca_scalefactor, level);

  499.         value  = av_clip(value, 0, (1 << log2range) - 1);

  500.     } else if (level < 8) {

  501.         if (level + 1 > log2range) {

  502.             skip_bits(gb, level + 1 - log2range);

  503.             value = get_bits(gb, log2range);

  504.         } else {

  505.             value = get_bits(gb, level + 1);

  506.         }

  507.     }

  508.     return value;

  509. }

  510. 

  511. static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)

  512. {

  513.     /* Primary audio coding side information */

  514.     int j, k;

  515. 

  516.     if (get_bits_left(&s->gb) < 0)

  517.         return AVERROR_INVALIDDATA;

  518. 

  519.     if (!base_channel) {

  520.         s->subsubframes[s->current_subframe]    = get_bits(&s->gb, 2) + 1;

  521.         s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3);

  522.     }

  523. 

  524.     for (j = base_channel; j < s->prim_channels; j++) {

  525.         for (k = 0; k < s->subband_activity[j]; k++)

  526.             s->prediction_mode[j][k] = get_bits(&s->gb, 1);

  527.     }

  528. 

  529.     /* Get prediction codebook */

  530.     for (j = base_channel; j < s->prim_channels; j++) {

  531.         for (k = 0; k < s->subband_activity[j]; k++) {

  532.             if (s->prediction_mode[j][k] > 0) {

  533.                 /* (Prediction coefficient VQ address) */

  534.                 s->prediction_vq[j][k] = get_bits(&s->gb, 12);

  535.             }

  536.         }

  537.     }

  538. 

  539.     /* Bit allocation index */

  540.     for (j = base_channel; j < s->prim_channels; j++) {

  541.         for (k = 0; k < s->vq_start_subband[j]; k++) {

  542.             if (s->bitalloc_huffman[j] == 6)

  543.                 s->bitalloc[j][k] = get_bits(&s->gb, 5);

  544.             else if (s->bitalloc_huffman[j] == 5)

  545.                 s->bitalloc[j][k] = get_bits(&s->gb, 4);

  546.             else if (s->bitalloc_huffman[j] == 7) {

  547.                 av_log(s->avctx, AV_LOG_ERROR,

  548.                        "Invalid bit allocation index\n");

  549.                 return AVERROR_INVALIDDATA;

  550.             } else {

  551.                 s->bitalloc[j][k] =

  552.                     get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);

  553.             }

  554. 

  555.             if (s->bitalloc[j][k] > 26) {

  556.                 av_dlog(s->avctx, "bitalloc index [%i][%i] too big (%i)\n",

  557.                         j, k, s->bitalloc[j][k]);

  558.                 return AVERROR_INVALIDDATA;

  559.             }

  560.         }

  561.     }

  562. 

  563.     /* Transition mode */

  564.     for (j = base_channel; j < s->prim_channels; j++) {

  565.         for (k = 0; k < s->subband_activity[j]; k++) {

  566.             s->transition_mode[j][k] = 0;

  567.             if (s->subsubframes[s->current_subframe] > 1 &&

  568.                 k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) {

  569.                 s->transition_mode[j][k] =

  570.                     get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]);

  571.             }

  572.         }

  573.     }

  574. 

  575.     if (get_bits_left(&s->gb) < 0)

  576.         return AVERROR_INVALIDDATA;

  577. 

  578.     for (j = base_channel; j < s->prim_channels; j++) {

  579.         const uint32_t *scale_table;

  580.         int scale_sum, log_size;

  581. 

  582.         memset(s->scale_factor[j], 0,

  583.                s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);

  584. 

  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. 

  593.         /* When huffman coded, only the difference is encoded */

  594.         scale_sum = 0;

  595. 

  596.         for (k = 0; k < s->subband_activity[j]; k++) {

  597.             if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) {

  598.                 scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size);

  599.                 s->scale_factor[j][k][0] = scale_table[scale_sum];

  600.             }

  601. 

  602.             if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) {

  603.                 /* Get second scale factor */

  604.                 scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size);

  605.                 s->scale_factor[j][k][1] = scale_table[scale_sum];

  606.             }

  607.         }

  608.     }

  609. 

  610.     /* Joint subband scale factor codebook select */

  611.     for (j = base_channel; j < s->prim_channels; j++) {

  612.         /* Transmitted only if joint subband coding enabled */

  613.         if (s->joint_intensity[j] > 0)

  614.             s->joint_huff[j] = get_bits(&s->gb, 3);

  615.     }

  616. 

  617.     if (get_bits_left(&s->gb) < 0)

  618.         return AVERROR_INVALIDDATA;

  619. 

  620.     /* Scale factors for joint subband coding */

  621.     for (j = base_channel; j < s->prim_channels; j++) {

  622.         int source_channel;

  623. 

  624.         /* Transmitted only if joint subband coding enabled */

  625.         if (s->joint_intensity[j] > 0) {

  626.             int scale = 0;

  627.             source_channel = s->joint_intensity[j] - 1;

  628. 

  629.             /* When huffman coded, only the difference is encoded

  630.              * (is this valid as well for joint scales ???) */

  631. 

  632.             for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) {

  633.                 scale = get_scale(&s->gb, s->joint_huff[j], 64 /* bias */, 7);

  634.                 s->joint_scale_factor[j][k] = scale;    /*joint_scale_table[scale]; */

  635.             }

  636. 

  637.             if (!(s->debug_flag & 0x02)) {

  638.                 av_log(s->avctx, AV_LOG_DEBUG,

  639.                        "Joint stereo coding not supported\n");

  640.                 s->debug_flag |= 0x02;

  641.             }

  642.         }

  643.     }

  644. 

  645.     /* Dynamic range coefficient */

  646.     if (!base_channel && s->dynrange)

  647.         s->dynrange_coef = get_bits(&s->gb, 8);

  648. 

  649.     /* Side information CRC check word */

  650.     if (s->crc_present) {

  651.         get_bits(&s->gb, 16);

  652.     }

  653. 

  654.     /*

  655.      * Primary audio data arrays

  656.      */

  657. 

  658.     /* VQ encoded high frequency subbands */

  659.     for (j = base_channel; j < s->prim_channels; j++)

  660.         for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)

  661.             /* 1 vector -> 32 samples */

  662.             s->high_freq_vq[j][k] = get_bits(&s->gb, 10);

  663. 

  664.     /* Low frequency effect data */

  665.     if (!base_channel && s->lfe) {

  666.         /* LFE samples */

  667.         int lfe_samples    = 2 * s->lfe * (4 + block_index);

  668.         int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);

  669.         float lfe_scale;

  670. 

  671.         for (j = lfe_samples; j < lfe_end_sample; j++) {

  672.             /* Signed 8 bits int */

  673.             s->lfe_data[j] = get_sbits(&s->gb, 8);

  674.         }

  675. 

  676.         /* Scale factor index */

  677.         skip_bits(&s->gb, 1);

  678.         s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 7)];

  679. 

  680.         /* Quantization step size * scale factor */

  681.         lfe_scale = 0.035 * s->lfe_scale_factor;

  682. 

  683.         for (j = lfe_samples; j < lfe_end_sample; j++)

  684.             s->lfe_data[j] *= lfe_scale;

  685.     }

  686. 

  687. #ifdef TRACE

  688.     av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n",

  689.            s->subsubframes[s->current_subframe]);

  690.     av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n",

  691.            s->partial_samples[s->current_subframe]);

  692. 

  693.     for (j = base_channel; j < s->prim_channels; j++) {

  694.         av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:");

  695.         for (k = 0; k < s->subband_activity[j]; k++)

  696.             av_log(s->avctx, AV_LOG_DEBUG, " %i", s->prediction_mode[j][k]);

  697.         av_log(s->avctx, AV_LOG_DEBUG, "\n");

  698.     }

  699.     for (j = base_channel; j < s->prim_channels; j++) {

  700.         for (k = 0; k < s->subband_activity[j]; k++)

  701.             av_log(s->avctx, AV_LOG_DEBUG,

  702.                    "prediction coefs: %f, %f, %f, %f\n",

  703.                    (float) adpcm_vb[s->prediction_vq[j][k]][0] / 8192,

  704.                    (float) adpcm_vb[s->prediction_vq[j][k]][1] / 8192,

  705.                    (float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192,

  706.                    (float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192);

  707.     }

  708.     for (j = base_channel; j < s->prim_channels; j++) {

  709.         av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: ");

  710.         for (k = 0; k < s->vq_start_subband[j]; k++)

  711.             av_log(s->avctx, AV_LOG_DEBUG, "%2.2i ", s->bitalloc[j][k]);

  712.         av_log(s->avctx, AV_LOG_DEBUG, "\n");

  713.     }

  714.     for (j = base_channel; j < s->prim_channels; j++) {

  715.         av_log(s->avctx, AV_LOG_DEBUG, "Transition mode:");

  716.         for (k = 0; k < s->subband_activity[j]; k++)

  717.             av_log(s->avctx, AV_LOG_DEBUG, " %i", s->transition_mode[j][k]);

  718.         av_log(s->avctx, AV_LOG_DEBUG, "\n");

  719.     }

  720.     for (j = base_channel; j < s->prim_channels; j++) {

  721.         av_log(s->avctx, AV_LOG_DEBUG, "Scale factor:");

  722.         for (k = 0; k < s->subband_activity[j]; k++) {

  723.             if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0)

  724.                 av_log(s->avctx, AV_LOG_DEBUG, " %i", s->scale_factor[j][k][0]);

  725.             if (k < s->vq_start_subband[j] && s->transition_mode[j][k])

  726.                 av_log(s->avctx, AV_LOG_DEBUG, " %i(t)", s->scale_factor[j][k][1]);

  727.         }

  728.         av_log(s->avctx, AV_LOG_DEBUG, "\n");

  729.     }

  730.     for (j = base_channel; j < s->prim_channels; j++) {

  731.         if (s->joint_intensity[j] > 0) {

  732.             int source_channel = s->joint_intensity[j] - 1;

  733.             av_log(s->avctx, AV_LOG_DEBUG, "Joint scale factor index:\n");

  734.             for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++)

  735.                 av_log(s->avctx, AV_LOG_DEBUG, " %i", s->joint_scale_factor[j][k]);

  736.             av_log(s->avctx, AV_LOG_DEBUG, "\n");

  737.         }

  738.     }

  739.     for (j = base_channel; j < s->prim_channels; j++)

  740.         for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)

  741.             av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]);

  742.     if (!base_channel && s->lfe) {

  743.         int lfe_samples    = 2 * s->lfe * (4 + block_index);

  744.         int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);

  745. 

  746.         av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n");

  747.         for (j = lfe_samples; j < lfe_end_sample; j++)

  748.             av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]);

  749.         av_log(s->avctx, AV_LOG_DEBUG, "\n");

  750.     }

  751. #endif

  752. 

  753.     return 0;

  754. }

  755. 

  756. static void qmf_32_subbands(DCAContext *s, int chans,

  757.                             float samples_in[32][8], float *samples_out,

  758.                             float scale)

  759. {

  760.     const float *prCoeff;

  761. 

  762.     int sb_act = s->subband_activity[chans];

  763. 

  764.     scale *= sqrt(1 / 8.0);

  765. 

  766.     /* Select filter */

  767.     if (!s->multirate_inter)    /* Non-perfect reconstruction */

  768.         prCoeff = fir_32bands_nonperfect;

  769.     else                        /* Perfect reconstruction */

  770.         prCoeff = fir_32bands_perfect;

  771. 

  772.     s->dcadsp.qmf_32_subbands(samples_in, sb_act, &s->synth, &s->imdct,

  773.                               s->subband_fir_hist[chans],

  774.                               &s->hist_index[chans],

  775.                               s->subband_fir_noidea[chans], prCoeff,

  776.                               samples_out, s->raXin, scale);

  777. }

  778. 

  779. static void lfe_interpolation_fir(DCAContext *s, int decimation_select,

  780.                                   int num_deci_sample, float *samples_in,

  781.                                   float *samples_out)

  782. {

  783.     /* samples_in: An array holding decimated samples.

  784.      *   Samples in current subframe starts from samples_in[0],

  785.      *   while samples_in[-1], samples_in[-2], ..., stores samples

  786.      *   from last subframe as history.

  787.      *

  788.      * samples_out: An array holding interpolated samples

  789.      */

  790. 

  791.     int idx;

  792.     const float *prCoeff;

  793.     int deciindex;

  794. 

  795.     /* Select decimation filter */

  796.     if (decimation_select == 1) {

  797.         idx     = 1;

  798.         prCoeff = lfe_fir_128;

  799.     } else {

  800.         idx     = 0;

  801.         prCoeff = lfe_fir_64;

  802.     }

  803.     /* Interpolation */

  804.     for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {

  805.         s->dcadsp.lfe_fir[idx](samples_out, samples_in, prCoeff);

  806.         samples_in++;

  807.         samples_out += 2 * 32 * (1 + idx);

  808.     }

  809. }

  810. 

  811. /* downmixing routines */

  812. #define MIX_REAR1(samples, s1, rs, coef)            \

  813.     samples[0][i] += samples[s1][i] * coef[rs][0];  \

  814.     samples[1][i] += samples[s1][i] * coef[rs][1];

  815. 

  816. #define MIX_REAR2(samples, s1, s2, rs, coef)                                          \

  817.     samples[0][i] += samples[s1][i] * coef[rs][0] + samples[s2][i] * coef[rs + 1][0]; \

  818.     samples[1][i] += samples[s1][i] * coef[rs][1] + samples[s2][i] * coef[rs + 1][1];

  819. 

  820. #define MIX_FRONT3(samples, coef)                                      \

  821.     t = samples[c][i];                                                 \

  822.     u = samples[l][i];                                                 \

  823.     v = samples[r][i];                                                 \

  824.     samples[0][i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0];  \

  825.     samples[1][i] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];

  826. 

  827. #define DOWNMIX_TO_STEREO(op1, op2)             \

  828.     for (i = 0; i < 256; i++) {                 \

  829.         op1                                     \

  830.         op2                                     \

  831.     }

  832. 

  833. static void dca_downmix(float **samples, int srcfmt, int lfe_present,

  834.                         float coef[DCA_PRIM_CHANNELS_MAX + 1][2],

  835.                         const int8_t *channel_mapping)

  836. {

  837.     int c, l, r, sl, sr, s;

  838.     int i;

  839.     float t, u, v;

  840. 

  841.     switch (srcfmt) {

  842.     case DCA_MONO:

  843.     case DCA_4F2R:

  844.         av_log(NULL, 0, "Not implemented!\n");

  845.         break;

  846.     case DCA_CHANNEL:

  847.     case DCA_STEREO:

  848.     case DCA_STEREO_TOTAL:

  849.     case DCA_STEREO_SUMDIFF:

  850.         break;

  851.     case DCA_3F:

  852.         c = channel_mapping[0];

  853.         l = channel_mapping[1];

  854.         r = channel_mapping[2];

  855.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );

  856.         break;

  857.     case DCA_2F1R:

  858.         s = channel_mapping[2];

  859.         DOWNMIX_TO_STEREO(MIX_REAR1(samples, s, 2, coef), );

  860.         break;

  861.     case DCA_3F1R:

  862.         c = channel_mapping[0];

  863.         l = channel_mapping[1];

  864.         r = channel_mapping[2];

  865.         s = channel_mapping[3];

  866.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  867.                           MIX_REAR1(samples, s, 3, coef));

  868.         break;

  869.     case DCA_2F2R:

  870.         sl = channel_mapping[2];

  871.         sr = channel_mapping[3];

  872.         DOWNMIX_TO_STEREO(MIX_REAR2(samples, sl, sr, 2, coef), );

  873.         break;

  874.     case DCA_3F2R:

  875.         c  = channel_mapping[0];

  876.         l  = channel_mapping[1];

  877.         r  = channel_mapping[2];

  878.         sl = channel_mapping[3];

  879.         sr = channel_mapping[4];

  880.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  881.                           MIX_REAR2(samples, sl, sr, 3, coef));

  882.         break;

  883.     }

  884.     if (lfe_present) {

  885.         int lf_buf = dca_lfe_index[srcfmt];

  886.         int lf_idx =  dca_channels[srcfmt];

  887.         for (i = 0; i < 256; i++) {

  888.             samples[0][i] += samples[lf_buf][i] * coef[lf_idx][0];

  889.             samples[1][i] += samples[lf_buf][i] * coef[lf_idx][1];

  890.         }

  891.     }

  892. }

  893. 

  894. #ifndef decode_blockcodes

  895. /* Very compact version of the block code decoder that does not use table

  896.  * look-up but is slightly slower */

  897. static int decode_blockcode(int code, int levels, int32_t *values)

  898. {

  899.     int i;

  900.     int offset = (levels - 1) >> 1;

  901. 

  902.     for (i = 0; i < 4; i++) {

  903.         int div = FASTDIV(code, levels);

  904.         values[i] = code - offset - div * levels;

  905.         code      = div;

  906.     }

  907. 

  908.     return code;

  909. }

  910. 

  911. static int decode_blockcodes(int code1, int code2, int levels, int32_t *values)

  912. {

  913.     return decode_blockcode(code1, levels, values) |

  914.            decode_blockcode(code2, levels, values + 4);

  915. }

  916. #endif

  917. 

  918. static const uint8_t abits_sizes[7]  = { 7, 10, 12, 13, 15, 17, 19 };

  919. static const uint8_t abits_levels[7] = { 3,  5,  7,  9, 13, 17, 25 };

  920. 

  921. static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)

  922. {

  923.     int k, l;

  924.     int subsubframe = s->current_subsubframe;

  925. 

  926.     const float *quant_step_table;

  927. 

  928.     /* FIXME */

  929.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  930.     LOCAL_ALIGNED_16(int32_t, block, [8 * DCA_SUBBANDS]);

  931. 

  932.     /*

  933.      * Audio data

  934.      */

  935. 

  936.     /* Select quantization step size table */

  937.     if (s->bit_rate_index == 0x1f)

  938.         quant_step_table = lossless_quant_d;

  939.     else

  940.         quant_step_table = lossy_quant_d;

  941. 

  942.     for (k = base_channel; k < s->prim_channels; k++) {

  943.         float rscale[DCA_SUBBANDS];

  944. 

  945.         if (get_bits_left(&s->gb) < 0)

  946.             return AVERROR_INVALIDDATA;

  947. 

  948.         for (l = 0; l < s->vq_start_subband[k]; l++) {

  949.             int m;

  950. 

  951.             /* Select the mid-tread linear quantizer */

  952.             int abits = s->bitalloc[k][l];

  953. 

  954.             float quant_step_size = quant_step_table[abits];

  955. 

  956.             /*

  957.              * Determine quantization index code book and its type

  958.              */

  959. 

  960.             /* Select quantization index code book */

  961.             int sel = s->quant_index_huffman[k][abits];

  962. 

  963.             /*

  964.              * Extract bits from the bit stream

  965.              */

  966.             if (!abits) {

  967.                 rscale[l] = 0;

  968.                 memset(block + 8 * l, 0, 8 * sizeof(block[0]));

  969.             } else {

  970.                 /* Deal with transients */

  971.                 int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];

  972.                 rscale[l] = quant_step_size * s->scale_factor[k][l][sfi] *

  973.                             s->scalefactor_adj[k][sel];

  974. 

  975.                 if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {

  976.                     if (abits <= 7) {

  977.                         /* Block code */

  978.                         int block_code1, block_code2, size, levels, err;

  979. 

  980.                         size   = abits_sizes[abits - 1];

  981.                         levels = abits_levels[abits - 1];

  982. 

  983.                         block_code1 = get_bits(&s->gb, size);

  984.                         block_code2 = get_bits(&s->gb, size);

  985.                         err         = decode_blockcodes(block_code1, block_code2,

  986.                                                         levels, block + 8 * l);

  987.                         if (err) {

  988.                             av_log(s->avctx, AV_LOG_ERROR,

  989.                                    "ERROR: block code look-up failed\n");

  990.                             return AVERROR_INVALIDDATA;

  991.                         }

  992.                     } else {

  993.                         /* no coding */

  994.                         for (m = 0; m < 8; m++)

  995.                             block[8 * l + m] = get_sbits(&s->gb, abits - 3);

  996.                     }

  997.                 } else {

  998.                     /* Huffman coded */

  999.                     for (m = 0; m < 8; m++)

  1000.                         block[8 * l + m] = get_bitalloc(&s->gb,

  1001.                                                         &dca_smpl_bitalloc[abits], sel);

  1002.                 }

  1003.             }

  1004.         }

  1005. 

  1006.         s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[k][0],

  1007.                                                block, rscale, 8 * s->vq_start_subband[k]);

  1008. 

  1009.         for (l = 0; l < s->vq_start_subband[k]; l++) {

  1010.             int m;

  1011.             /*

  1012.              * Inverse ADPCM if in prediction mode

  1013.              */

  1014.             if (s->prediction_mode[k][l]) {

  1015.                 int n;

  1016.                 if (s->predictor_history)

  1017.                     subband_samples[k][l][0] += (adpcm_vb[s->prediction_vq[k][l]][0] *

  1018.                                                  s->subband_samples_hist[k][l][3] +

  1019.                                                  adpcm_vb[s->prediction_vq[k][l]][1] *

  1020.                                                  s->subband_samples_hist[k][l][2] +

  1021.                                                  adpcm_vb[s->prediction_vq[k][l]][2] *

  1022.                                                  s->subband_samples_hist[k][l][1] +

  1023.                                                  adpcm_vb[s->prediction_vq[k][l]][3] *

  1024.                                                  s->subband_samples_hist[k][l][0]) *

  1025.                                                 (1.0f / 8192);

  1026.                 for (m = 1; m < 8; m++) {

  1027.                     float sum = adpcm_vb[s->prediction_vq[k][l]][0] *

  1028.                                 subband_samples[k][l][m - 1];

  1029.                     for (n = 2; n <= 4; n++)

  1030.                         if (m >= n)

  1031.                             sum += adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1032.                                    subband_samples[k][l][m - n];

  1033.                         else if (s->predictor_history)

  1034.                             sum += adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1035.                                    s->subband_samples_hist[k][l][m - n + 4];

  1036.                     subband_samples[k][l][m] += sum * 1.0f / 8192;

  1037.                 }

  1038.             }

  1039.         }

  1040. 

  1041.         /*

  1042.          * Decode VQ encoded high frequencies

  1043.          */

  1044.         if (s->subband_activity[k] > s->vq_start_subband[k]) {

  1045.             if (!s->debug_flag & 0x01) {

  1046.                 av_log(s->avctx, AV_LOG_DEBUG,

  1047.                        "Stream with high frequencies VQ coding\n");

  1048.                 s->debug_flag |= 0x01;

  1049.             }

  1050.             s->dcadsp.decode_hf(subband_samples[k], s->high_freq_vq[k],

  1051.                                 high_freq_vq, subsubframe * 8,

  1052.                                 s->scale_factor[k], s->vq_start_subband[k],

  1053.                                 s->subband_activity[k]);

  1054.         }

  1055.     }

  1056. 

  1057.     /* Check for DSYNC after subsubframe */

  1058.     if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {

  1059.         if (0xFFFF == get_bits(&s->gb, 16)) {   /* 0xFFFF */

  1060. #ifdef TRACE

  1061.             av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");

  1062. #endif

  1063.         } else {

  1064.             av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");

  1065.             return AVERROR_INVALIDDATA;

  1066.         }

  1067.     }

  1068. 

  1069.     /* Backup predictor history for adpcm */

  1070.     for (k = base_channel; k < s->prim_channels; k++)

  1071.         for (l = 0; l < s->vq_start_subband[k]; l++)

  1072.             AV_COPY128(s->subband_samples_hist[k][l], &subband_samples[k][l][4]);

  1073. 

  1074.     return 0;

  1075. }

  1076. 

  1077. static int dca_filter_channels(DCAContext *s, int block_index)

  1078. {

  1079.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1080.     int k;

  1081. 

  1082.     /* 32 subbands QMF */

  1083.     for (k = 0; k < s->prim_channels; k++) {

  1084.         if (s->channel_order_tab[k] >= 0)

  1085.             qmf_32_subbands(s, k, subband_samples[k],

  1086.                             s->samples_chanptr[s->channel_order_tab[k]],

  1087.                             M_SQRT1_2 / 32768.0);

  1088.     }

  1089. 

  1090.     /* Generate LFE samples for this subsubframe FIXME!!! */

  1091.     if (s->lfe) {

  1092.         lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,

  1093.                               s->lfe_data + 2 * s->lfe * (block_index + 4),

  1094.                               s->samples_chanptr[dca_lfe_index[s->amode]]);

  1095.         /* Outputs 20bits pcm samples */

  1096.     }

  1097. 

  1098.     /* Downmixing to Stereo */

  1099.     if (s->prim_channels + !!s->lfe > 2 &&

  1100.         s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {

  1101.         dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef,

  1102.                     s->channel_order_tab);

  1103.     }

  1104. 

  1105.     return 0;

  1106. }

  1107. 

  1108. static int dca_subframe_footer(DCAContext *s, int base_channel)

  1109. {

  1110.     int in, out, aux_data_count, aux_data_end, reserved;

  1111.     uint32_t nsyncaux;

  1112. 

  1113.     /*

  1114.      * Unpack optional information

  1115.      */

  1116. 

  1117.     /* presumably optional information only appears in the core? */

  1118.     if (!base_channel) {

  1119.         if (s->timestamp)

  1120.             skip_bits_long(&s->gb, 32);

  1121. 

  1122.         if (s->aux_data) {

  1123.             aux_data_count = get_bits(&s->gb, 6);

  1124. 

  1125.             // align (32-bit)

  1126.             skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1127. 

  1128.             aux_data_end = 8 * aux_data_count + get_bits_count(&s->gb);

  1129. 

  1130.             if ((nsyncaux = get_bits_long(&s->gb, 32)) != DCA_NSYNCAUX) {

  1131.                 av_log(s->avctx, AV_LOG_ERROR, "nSYNCAUX mismatch %#"PRIx32"\n",

  1132.                        nsyncaux);

  1133.                 return AVERROR_INVALIDDATA;

  1134.             }

  1135. 

  1136.             if (get_bits1(&s->gb)) { // bAUXTimeStampFlag

  1137.                 avpriv_request_sample(s->avctx,

  1138.                                       "Auxiliary Decode Time Stamp Flag");

  1139.                 // align (4-bit)

  1140.                 skip_bits(&s->gb, (-get_bits_count(&s->gb)) & 4);

  1141.                 // 44 bits: nMSByte (8), nMarker (4), nLSByte (28), nMarker (4)

  1142.                 skip_bits_long(&s->gb, 44);

  1143.             }

  1144. 

  1145.             if ((s->core_downmix = get_bits1(&s->gb))) {

  1146.                 int am = get_bits(&s->gb, 3);

  1147.                 switch (am) {

  1148.                 case 0:

  1149.                     s->core_downmix_amode = DCA_MONO;

  1150.                     break;

  1151.                 case 1:

  1152.                     s->core_downmix_amode = DCA_STEREO;

  1153.                     break;

  1154.                 case 2:

  1155.                     s->core_downmix_amode = DCA_STEREO_TOTAL;

  1156.                     break;

  1157.                 case 3:

  1158.                     s->core_downmix_amode = DCA_3F;

  1159.                     break;

  1160.                 case 4:

  1161.                     s->core_downmix_amode = DCA_2F1R;

  1162.                     break;

  1163.                 case 5:

  1164.                     s->core_downmix_amode = DCA_2F2R;

  1165.                     break;

  1166.                 case 6:

  1167.                     s->core_downmix_amode = DCA_3F1R;

  1168.                     break;

  1169.                 default:

  1170.                     av_log(s->avctx, AV_LOG_ERROR,

  1171.                            "Invalid mode %d for embedded downmix coefficients\n",

  1172.                            am);

  1173.                     return AVERROR_INVALIDDATA;

  1174.                 }

  1175.                 for (out = 0; out < dca_channels[s->core_downmix_amode]; out++) {

  1176.                     for (in = 0; in < s->prim_channels + !!s->lfe; in++) {

  1177.                         uint16_t tmp = get_bits(&s->gb, 9);

  1178.                         if ((tmp & 0xFF) > 241) {

  1179.                             av_log(s->avctx, AV_LOG_ERROR,

  1180.                                    "Invalid downmix coefficient code %"PRIu16"\n",

  1181.                                    tmp);

  1182.                             return AVERROR_INVALIDDATA;

  1183.                         }

  1184.                         s->core_downmix_codes[in][out] = tmp;

  1185.                     }

  1186.                 }

  1187.             }

  1188. 

  1189.             align_get_bits(&s->gb); // byte align

  1190.             skip_bits(&s->gb, 16);  // nAUXCRC16

  1191. 

  1192.             // additional data (reserved, cf. ETSI TS 102 114 V1.4.1)

  1193.             if ((reserved = (aux_data_end - get_bits_count(&s->gb))) < 0) {

  1194.                 av_log(s->avctx, AV_LOG_ERROR,

  1195.                        "Overread auxiliary data by %d bits\n", -reserved);

  1196.                 return AVERROR_INVALIDDATA;

  1197.             } else if (reserved) {

  1198.                 avpriv_request_sample(s->avctx,

  1199.                                       "Core auxiliary data reserved content");

  1200.                 skip_bits_long(&s->gb, reserved);

  1201.             }

  1202.         }

  1203. 

  1204.         if (s->crc_present && s->dynrange)

  1205.             get_bits(&s->gb, 16);

  1206.     }

  1207. 

  1208.     return 0;

  1209. }

  1210. 

  1211. /**

  1212.  * Decode a dca frame block

  1213.  *

  1214.  * @param s     pointer to the DCAContext

  1215.  */

  1216. 

  1217. static int dca_decode_block(DCAContext *s, int base_channel, int block_index)

  1218. {

  1219.     int ret;

  1220. 

  1221.     /* Sanity check */

  1222.     if (s->current_subframe >= s->subframes) {

  1223.         av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",

  1224.                s->current_subframe, s->subframes);

  1225.         return AVERROR_INVALIDDATA;

  1226.     }

  1227. 

  1228.     if (!s->current_subsubframe) {

  1229. #ifdef TRACE

  1230.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");

  1231. #endif

  1232.         /* Read subframe header */

  1233.         if ((ret = dca_subframe_header(s, base_channel, block_index)))

  1234.             return ret;

  1235.     }

  1236. 

  1237.     /* Read subsubframe */

  1238. #ifdef TRACE

  1239.     av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");

  1240. #endif

  1241.     if ((ret = dca_subsubframe(s, base_channel, block_index)))

  1242.         return ret;

  1243. 

  1244.     /* Update state */

  1245.     s->current_subsubframe++;

  1246.     if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {

  1247.         s->current_subsubframe = 0;

  1248.         s->current_subframe++;

  1249.     }

  1250.     if (s->current_subframe >= s->subframes) {

  1251. #ifdef TRACE

  1252.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");

  1253. #endif

  1254.         /* Read subframe footer */

  1255.         if ((ret = dca_subframe_footer(s, base_channel)))

  1256.             return ret;

  1257.     }

  1258. 

  1259.     return 0;

  1260. }

  1261. 

  1262. static float dca_dmix_code(unsigned code)

  1263. {

  1264.     int sign = (code >> 8) - 1;

  1265.     code &= 0xff;

  1266.     return ((dca_dmixtable[code] ^ sign) - sign) * (1.0 / (1U << 15));

  1267. }

  1268. 

  1269. /**

  1270.  * Main frame decoding function

  1271.  * FIXME add arguments

  1272.  */

  1273. static int dca_decode_frame(AVCodecContext *avctx, void *data,

  1274.                             int *got_frame_ptr, AVPacket *avpkt)

  1275. {

  1276.     AVFrame *frame     = data;

  1277.     const uint8_t *buf = avpkt->data;

  1278.     int buf_size       = avpkt->size;

  1279. 

  1280.     int lfe_samples;

  1281.     int num_core_channels = 0;

  1282.     int i, ret;

  1283.     float  **samples_flt;

  1284.     DCAContext *s = avctx->priv_data;

  1285.     int channels, full_channels;

  1286.     int core_ss_end;

  1287. 

  1288.     s->xch_present = 0;

  1289. 

  1290.     s->dca_buffer_size = ff_dca_convert_bitstream(buf, buf_size, s->dca_buffer,

  1291.                                                   DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);

  1292.     if (s->dca_buffer_size == AVERROR_INVALIDDATA) {

  1293.         av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");

  1294.         return AVERROR_INVALIDDATA;

  1295.     }

  1296. 

  1297.     if ((ret = dca_parse_frame_header(s)) < 0) {

  1298.         // seems like the frame is corrupt, try with the next one

  1299.         return ret;

  1300.     }

  1301.     // set AVCodec values with parsed data

  1302.     avctx->sample_rate = s->sample_rate;

  1303.     avctx->bit_rate    = s->bit_rate;

  1304. 

  1305.     s->profile = FF_PROFILE_DTS;

  1306. 

  1307.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1308.         if ((ret = dca_decode_block(s, 0, i))) {

  1309.             av_log(avctx, AV_LOG_ERROR, "error decoding block\n");

  1310.             return ret;

  1311.         }

  1312.     }

  1313. 

  1314.     /* record number of core channels incase less than max channels are requested */

  1315.     num_core_channels = s->prim_channels;

  1316. 

  1317.     if (s->ext_coding)

  1318.         s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];

  1319.     else

  1320.         s->core_ext_mask = 0;

  1321. 

  1322.     core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;

  1323. 

  1324.     /* only scan for extensions if ext_descr was unknown or indicated a

  1325.      * supported XCh extension */

  1326.     if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {

  1327.         /* if ext_descr was unknown, clear s->core_ext_mask so that the

  1328.          * extensions scan can fill it up */

  1329.         s->core_ext_mask = FFMAX(s->core_ext_mask, 0);

  1330. 

  1331.         /* extensions start at 32-bit boundaries into bitstream */

  1332.         skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1333. 

  1334.         while (core_ss_end - get_bits_count(&s->gb) >= 32) {

  1335.             uint32_t bits = get_bits_long(&s->gb, 32);

  1336. 

  1337.             switch (bits) {

  1338.             case 0x5a5a5a5a: {

  1339.                 int ext_amode, xch_fsize;

  1340. 

  1341.                 s->xch_base_channel = s->prim_channels;

  1342. 

  1343.                 /* validate sync word using XCHFSIZE field */

  1344.                 xch_fsize = show_bits(&s->gb, 10);

  1345.                 if ((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&

  1346.                     (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))

  1347.                     continue;

  1348. 

  1349.                 /* skip length-to-end-of-frame field for the moment */

  1350.                 skip_bits(&s->gb, 10);

  1351. 

  1352.                 s->core_ext_mask |= DCA_EXT_XCH;

  1353. 

  1354.                 /* extension amode(number of channels in extension) should be 1 */

  1355.                 /* AFAIK XCh is not used for more channels */

  1356.                 if ((ext_amode = get_bits(&s->gb, 4)) != 1) {

  1357.                     av_log(avctx, AV_LOG_ERROR,

  1358.                            "XCh extension amode %d not supported!\n",

  1359.                            ext_amode);

  1360.                     continue;

  1361.                 }

  1362. 

  1363.                 /* much like core primary audio coding header */

  1364.                 dca_parse_audio_coding_header(s, s->xch_base_channel);

  1365. 

  1366.                 for (i = 0; i < (s->sample_blocks / 8); i++)

  1367.                     if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {

  1368.                         av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");

  1369.                         continue;

  1370.                     }

  1371. 

  1372.                 s->xch_present = 1;

  1373.                 break;

  1374.             }

  1375.             case 0x47004a03:

  1376.                 /* XXCh: extended channels */

  1377.                 /* usually found either in core or HD part in DTS-HD HRA streams,

  1378.                  * but not in DTS-ES which contains XCh extensions instead */

  1379.                 s->core_ext_mask |= DCA_EXT_XXCH;

  1380.                 break;

  1381. 

  1382.             case 0x1d95f262: {

  1383.                 int fsize96 = show_bits(&s->gb, 12) + 1;

  1384.                 if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)

  1385.                     continue;

  1386. 

  1387.                 av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n",

  1388.                        get_bits_count(&s->gb));

  1389.                 skip_bits(&s->gb, 12);

  1390.                 av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);

  1391.                 av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));

  1392. 

  1393.                 s->core_ext_mask |= DCA_EXT_X96;

  1394.                 break;

  1395.             }

  1396.             }

  1397. 

  1398.             skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1399.         }

  1400.     } else {

  1401.         /* no supported extensions, skip the rest of the core substream */

  1402.         skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));

  1403.     }

  1404. 

  1405.     if (s->core_ext_mask & DCA_EXT_X96)

  1406.         s->profile = FF_PROFILE_DTS_96_24;

  1407.     else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))

  1408.         s->profile = FF_PROFILE_DTS_ES;

  1409. 

  1410.     /* check for ExSS (HD part) */

  1411.     if (s->dca_buffer_size - s->frame_size > 32 &&

  1412.         get_bits_long(&s->gb, 32) == DCA_HD_MARKER)

  1413.         ff_dca_exss_parse_header(s);

  1414. 

  1415.     avctx->profile = s->profile;

  1416. 

  1417.     full_channels = channels = s->prim_channels + !!s->lfe;

  1418. 

  1419.     if (s->amode < 16) {

  1420.         avctx->channel_layout = dca_core_channel_layout[s->amode];

  1421. 

  1422.         if (s->prim_channels + !!s->lfe > 2 &&

  1423.             avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {

  1424.             /*

  1425.              * Neither the core's auxiliary data nor our default tables contain

  1426.              * downmix coefficients for the additional channel coded in the XCh

  1427.              * extension, so when we're doing a Stereo downmix, don't decode it.

  1428.              */

  1429.             s->xch_disable = 1;

  1430.         }

  1431. 

  1432. #if FF_API_REQUEST_CHANNELS

  1433. FF_DISABLE_DEPRECATION_WARNINGS

  1434.         if (s->xch_present && !s->xch_disable &&

  1435.             (!avctx->request_channels ||

  1436.              avctx->request_channels > num_core_channels + !!s->lfe)) {

  1437. FF_ENABLE_DEPRECATION_WARNINGS

  1438. #else

  1439.         if (s->xch_present && !s->xch_disable) {

  1440. #endif

  1441.             avctx->channel_layout |= AV_CH_BACK_CENTER;

  1442.             if (s->lfe) {

  1443.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1444.                 s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode];

  1445.             } else {

  1446.                 s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];

  1447.             }

  1448.         } else {

  1449.             channels       = num_core_channels + !!s->lfe;

  1450.             s->xch_present = 0; /* disable further xch processing */

  1451.             if (s->lfe) {

  1452.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1453.                 s->channel_order_tab = dca_channel_reorder_lfe[s->amode];

  1454.             } else

  1455.                 s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];

  1456.         }

  1457. 

  1458.         if (channels > !!s->lfe &&

  1459.             s->channel_order_tab[channels - 1 - !!s->lfe] < 0)

  1460.             return AVERROR_INVALIDDATA;

  1461. 

  1462.         if (num_core_channels + !!s->lfe > 2 &&

  1463.             avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {

  1464.             channels              = 2;

  1465.             s->output             = s->prim_channels == 2 ? s->amode : DCA_STEREO;

  1466.             avctx->channel_layout = AV_CH_LAYOUT_STEREO;

  1467. 

  1468.             /* Stereo downmix coefficients

  1469.              *

  1470.              * The decoder can only downmix to 2-channel, so we need to ensure

  1471.              * embedded downmix coefficients are actually targeting 2-channel.

  1472.              */

  1473.             if (s->core_downmix && (s->core_downmix_amode == DCA_STEREO ||

  1474.                                     s->core_downmix_amode == DCA_STEREO_TOTAL)) {

  1475.                 for (i = 0; i < num_core_channels + !!s->lfe; i++) {

  1476.                     /* Range checked earlier */

  1477.                     s->downmix_coef[i][0] = dca_dmix_code(s->core_downmix_codes[i][0]);

  1478.                     s->downmix_coef[i][1] = dca_dmix_code(s->core_downmix_codes[i][1]);

  1479.                 }

  1480.                 s->output = s->core_downmix_amode;

  1481.             } else {

  1482.                 int am = s->amode & DCA_CHANNEL_MASK;

  1483.                 if (am >= FF_ARRAY_ELEMS(dca_default_coeffs)) {

  1484.                     av_log(s->avctx, AV_LOG_ERROR,

  1485.                            "Invalid channel mode %d\n", am);

  1486.                     return AVERROR_INVALIDDATA;

  1487.                 }

  1488.                 if (num_core_channels + !!s->lfe >

  1489.                     FF_ARRAY_ELEMS(dca_default_coeffs[0])) {

  1490.                     avpriv_request_sample(s->avctx, "Downmixing %d channels",

  1491.                                           s->prim_channels + !!s->lfe);

  1492.                     return AVERROR_PATCHWELCOME;

  1493.                 }

  1494.                 for (i = 0; i < num_core_channels + !!s->lfe; i++) {

  1495.                     s->downmix_coef[i][0] = dca_default_coeffs[am][i][0];

  1496.                     s->downmix_coef[i][1] = dca_default_coeffs[am][i][1];

  1497.                 }

  1498.             }

  1499.             av_dlog(s->avctx, "Stereo downmix coeffs:\n");

  1500.             for (i = 0; i < num_core_channels + !!s->lfe; i++) {

  1501.                 av_dlog(s->avctx, "L, input channel %d = %f\n", i,

  1502.                         s->downmix_coef[i][0]);

  1503.                 av_dlog(s->avctx, "R, input channel %d = %f\n", i,

  1504.                         s->downmix_coef[i][1]);

  1505.             }

  1506.             av_dlog(s->avctx, "\n");

  1507.         }

  1508.     } else {

  1509.         av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);

  1510.         return AVERROR_INVALIDDATA;

  1511.     }

  1512.     avctx->channels = channels;

  1513. 

  1514.     /* get output buffer */

  1515.     frame->nb_samples = 256 * (s->sample_blocks / 8);

  1516.     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {

  1517.         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");

  1518.         return ret;

  1519.     }

  1520.     samples_flt = (float **) frame->extended_data;

  1521. 

  1522.     /* allocate buffer for extra channels if downmixing */

  1523.     if (avctx->channels < full_channels) {

  1524.         ret = av_samples_get_buffer_size(NULL, full_channels - channels,

  1525.                                          frame->nb_samples,

  1526.                                          avctx->sample_fmt, 0);

  1527.         if (ret < 0)

  1528.             return ret;

  1529. 

  1530.         av_fast_malloc(&s->extra_channels_buffer,

  1531.                        &s->extra_channels_buffer_size, ret);

  1532.         if (!s->extra_channels_buffer)

  1533.             return AVERROR(ENOMEM);

  1534. 

  1535.         ret = av_samples_fill_arrays((uint8_t **) s->extra_channels, NULL,

  1536.                                      s->extra_channels_buffer,

  1537.                                      full_channels - channels,

  1538.                                      frame->nb_samples, avctx->sample_fmt, 0);

  1539.         if (ret < 0)

  1540.             return ret;

  1541.     }

  1542. 

  1543.     /* filter to get final output */

  1544.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1545.         int ch;

  1546. 

  1547.         for (ch = 0; ch < channels; ch++)

  1548.             s->samples_chanptr[ch] = samples_flt[ch] + i * 256;

  1549.         for (; ch < full_channels; ch++)

  1550.             s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * 256;

  1551. 

  1552.         dca_filter_channels(s, i);

  1553. 

  1554.         /* If this was marked as a DTS-ES stream we need to subtract back- */

  1555.         /* channel from SL & SR to remove matrixed back-channel signal */

  1556.         if ((s->source_pcm_res & 1) && s->xch_present) {

  1557.             float *back_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel]];

  1558.             float *lt_chan   = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 2]];

  1559.             float *rt_chan   = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 1]];

  1560.             s->fdsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);

  1561.             s->fdsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);

  1562.         }

  1563.     }

  1564. 

  1565.     /* update lfe history */

  1566.     lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);

  1567.     for (i = 0; i < 2 * s->lfe * 4; i++)

  1568.         s->lfe_data[i] = s->lfe_data[i + lfe_samples];

  1569. 

  1570.     /* AVMatrixEncoding

  1571.      *

  1572.      * DCA_STEREO_TOTAL (Lt/Rt) is equivalent to Dolby Surround */

  1573.     ret = ff_side_data_update_matrix_encoding(frame,

  1574.                                               (s->output & ~DCA_LFE) == DCA_STEREO_TOTAL ?

  1575.                                               AV_MATRIX_ENCODING_DOLBY : AV_MATRIX_ENCODING_NONE);

  1576.     if (ret < 0)

  1577.         return ret;

  1578. 

  1579.     *got_frame_ptr = 1;

  1580. 

  1581.     return buf_size;

  1582. }

  1583. 

  1584. /**

  1585.  * DCA initialization

  1586.  *

  1587.  * @param avctx     pointer to the AVCodecContext

  1588.  */

  1589. 

  1590. static av_cold int dca_decode_init(AVCodecContext *avctx)

  1591. {

  1592.     DCAContext *s = avctx->priv_data;

  1593. 

  1594.     s->avctx = avctx;

  1595.     dca_init_vlcs();

  1596. 

  1597.     avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);

  1598.     ff_mdct_init(&s->imdct, 6, 1, 1.0);

  1599.     ff_synth_filter_init(&s->synth);

  1600.     ff_dcadsp_init(&s->dcadsp);

  1601.     ff_fmt_convert_init(&s->fmt_conv, avctx);

  1602. 

  1603.     avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

  1604. 

  1605.     /* allow downmixing to stereo */

  1606. #if FF_API_REQUEST_CHANNELS

  1607. FF_DISABLE_DEPRECATION_WARNINGS

  1608.     if (avctx->request_channels == 2)

  1609.         avctx->request_channel_layout = AV_CH_LAYOUT_STEREO;

  1610. FF_ENABLE_DEPRECATION_WARNINGS

  1611. #endif

  1612.     if (avctx->channels > 2 &&

  1613.         avctx->request_channel_layout == AV_CH_LAYOUT_STEREO)

  1614.         avctx->channels = 2;

  1615. 

  1616.     return 0;

  1617. }

  1618. 

  1619. static av_cold int dca_decode_end(AVCodecContext *avctx)

  1620. {

  1621.     DCAContext *s = avctx->priv_data;

  1622.     ff_mdct_end(&s->imdct);

  1623.     av_freep(&s->extra_channels_buffer);

  1624.     return 0;

  1625. }

  1626. 

  1627. static const AVProfile profiles[] = {

  1628.     { FF_PROFILE_DTS,        "DTS"        },

  1629.     { FF_PROFILE_DTS_ES,     "DTS-ES"     },

  1630.     { FF_PROFILE_DTS_96_24,  "DTS 96/24"  },

  1631.     { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },

  1632.     { FF_PROFILE_DTS_HD_MA,  "DTS-HD MA"  },

  1633.     { FF_PROFILE_UNKNOWN },

  1634. };

  1635. 

  1636. static const AVOption options[] = {

  1637.     { "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 },

  1638.     { NULL },

  1639. };

  1640. 

  1641. static const AVClass dca_decoder_class = {

  1642.     .class_name = "DCA decoder",

  1643.     .item_name  = av_default_item_name,

  1644.     .option     = options,

  1645.     .version    = LIBAVUTIL_VERSION_INT,

  1646. };

  1647. 

  1648. AVCodec ff_dca_decoder = {

  1649.     .name            = "dca",

  1650.     .long_name       = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),

  1651.     .type            = AVMEDIA_TYPE_AUDIO,

  1652.     .id              = AV_CODEC_ID_DTS,

  1653.     .priv_data_size  = sizeof(DCAContext),

  1654.     .init            = dca_decode_init,

  1655.     .decode          = dca_decode_frame,

  1656.     .close           = dca_decode_end,

  1657.     .capabilities    = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,

  1658.     .sample_fmts     = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,

  1659.                                                        AV_SAMPLE_FMT_NONE },

  1660.     .profiles        = NULL_IF_CONFIG_SMALL(profiles),

  1661.     .priv_class      = &dca_decoder_class,

  1662. };