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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. #include "avcodec.h"

  38. #include "fft.h"

  39. #include "get_bits.h"

  40. #include "put_bits.h"

  41. #include "dcadata.h"

  42. #include "dcahuff.h"

  43. #include "dca.h"

  44. #include "mathops.h"

  45. #include "synth_filter.h"

  46. #include "dcadsp.h"

  47. #include "fmtconvert.h"

  48. #include "internal.h"

  49. 

  50. #if ARCH_ARM

  51. #   include "arm/dca.h"

  52. #endif

  53. 

  54. //#define TRACE

  55. 

  56. #define DCA_PRIM_CHANNELS_MAX  (7)

  57. #define DCA_SUBBANDS          (32)

  58. #define DCA_ABITS_MAX         (32)      /* Should be 28 */

  59. #define DCA_SUBSUBFRAMES_MAX   (4)

  60. #define DCA_SUBFRAMES_MAX     (16)

  61. #define DCA_BLOCKS_MAX        (16)

  62. #define DCA_LFE_MAX            (3)

  63. 

  64. enum DCAMode {

  65.     DCA_MONO = 0,

  66.     DCA_CHANNEL,

  67.     DCA_STEREO,

  68.     DCA_STEREO_SUMDIFF,

  69.     DCA_STEREO_TOTAL,

  70.     DCA_3F,

  71.     DCA_2F1R,

  72.     DCA_3F1R,

  73.     DCA_2F2R,

  74.     DCA_3F2R,

  75.     DCA_4F2R

  76. };

  77. 

  78. /* these are unconfirmed but should be mostly correct */

  79. enum DCAExSSSpeakerMask {

  80.     DCA_EXSS_FRONT_CENTER          = 0x0001,

  81.     DCA_EXSS_FRONT_LEFT_RIGHT      = 0x0002,

  82.     DCA_EXSS_SIDE_REAR_LEFT_RIGHT  = 0x0004,

  83.     DCA_EXSS_LFE                   = 0x0008,

  84.     DCA_EXSS_REAR_CENTER           = 0x0010,

  85.     DCA_EXSS_FRONT_HIGH_LEFT_RIGHT = 0x0020,

  86.     DCA_EXSS_REAR_LEFT_RIGHT       = 0x0040,

  87.     DCA_EXSS_FRONT_HIGH_CENTER     = 0x0080,

  88.     DCA_EXSS_OVERHEAD              = 0x0100,

  89.     DCA_EXSS_CENTER_LEFT_RIGHT     = 0x0200,

  90.     DCA_EXSS_WIDE_LEFT_RIGHT       = 0x0400,

  91.     DCA_EXSS_SIDE_LEFT_RIGHT       = 0x0800,

  92.     DCA_EXSS_LFE2                  = 0x1000,

  93.     DCA_EXSS_SIDE_HIGH_LEFT_RIGHT  = 0x2000,

  94.     DCA_EXSS_REAR_HIGH_CENTER      = 0x4000,

  95.     DCA_EXSS_REAR_HIGH_LEFT_RIGHT  = 0x8000,

  96. };

  97. 

  98. enum DCAExtensionMask {

  99.     DCA_EXT_CORE       = 0x001, ///< core in core substream

  100.     DCA_EXT_XXCH       = 0x002, ///< XXCh channels extension in core substream

  101.     DCA_EXT_X96        = 0x004, ///< 96/24 extension in core substream

  102.     DCA_EXT_XCH        = 0x008, ///< XCh channel extension in core substream

  103.     DCA_EXT_EXSS_CORE  = 0x010, ///< core in ExSS (extension substream)

  104.     DCA_EXT_EXSS_XBR   = 0x020, ///< extended bitrate extension in ExSS

  105.     DCA_EXT_EXSS_XXCH  = 0x040, ///< XXCh channels extension in ExSS

  106.     DCA_EXT_EXSS_X96   = 0x080, ///< 96/24 extension in ExSS

  107.     DCA_EXT_EXSS_LBR   = 0x100, ///< low bitrate component in ExSS

  108.     DCA_EXT_EXSS_XLL   = 0x200, ///< lossless extension in ExSS

  109. };

  110. 

  111. /* -1 are reserved or unknown */

  112. static const int dca_ext_audio_descr_mask[] = {

  113.     DCA_EXT_XCH,

  114.     -1,

  115.     DCA_EXT_X96,

  116.     DCA_EXT_XCH | DCA_EXT_X96,

  117.     -1,

  118.     -1,

  119.     DCA_EXT_XXCH,

  120.     -1,

  121. };

  122. 

  123. /* extensions that reside in core substream */

  124. #define DCA_CORE_EXTS (DCA_EXT_XCH | DCA_EXT_XXCH | DCA_EXT_X96)

  125. 

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

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

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

  129.  *

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

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

  132.  * OV -> center back

  133.  * All 2 channel configurations -> AV_CH_LAYOUT_STEREO

  134.  */

  135. static const uint64_t dca_core_channel_layout[] = {

  136.     AV_CH_FRONT_CENTER,                                                     ///< 1, A

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

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

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

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

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

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

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

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

  145. 

  146.     AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_SIDE_LEFT |

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

  148. 

  149.     AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |

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

  151. 

  152.     AV_CH_LAYOUT_STEREO | AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT |

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

  154. 

  155.     AV_CH_FRONT_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |

  156.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_BACK_CENTER   |

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

  158. 

  159.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER   |

  160.     AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |

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

  162. 

  163.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |

  164.     AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |

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

  166. 

  167.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER   |

  168.     AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |

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

  170. };

  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. 

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

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

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

  183.     { 0,  1,  3, -1, -1, -1, -1, -1, -1},

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

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

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

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

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

  189.     { 0,  6,  4,  5,  2,  3, -1, -1, -1},

  190.     { 4,  2,  5,  0,  1,  6,  7, -1, -1},

  191.     { 5,  6,  0,  1,  7,  3,  8,  4, -1},

  192.     { 4,  2,  5,  0,  1,  6,  8,  7, -1},

  193. };

  194. 

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

  196.     { 0,  2, -1, -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.     { 0,  1,  3, -1, -1, -1, -1, -1, -1},

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

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

  202.     { 0,  1,  3,  4, -1, -1, -1, -1, -1},

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

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

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

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

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

  208.     { 0,  6,  4,  5,  2,  3,  7, -1, -1},

  209.     { 4,  2,  5,  0,  1,  7,  8,  6, -1},

  210.     { 5,  6,  0,  1,  8,  3,  9,  4,  7},

  211.     { 4,  2,  5,  0,  1,  6,  9,  8,  7},

  212. };

  213. 

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

  215.     { 0, -1, -1, -1, -1, -1, -1, -1, -1},

  216.     { 0,  1, -1, -1, -1, -1, -1, -1, -1},

  217.     { 0,  1, -1, -1, -1, -1, -1, -1, -1},

  218.     { 0,  1, -1, -1, -1, -1, -1, -1, -1},

  219.     { 0,  1, -1, -1, -1, -1, -1, -1, -1},

  220.     { 2,  0,  1, -1, -1, -1, -1, -1, -1},

  221.     { 0,  1,  2, -1, -1, -1, -1, -1, -1},

  222.     { 2,  0,  1,  3, -1, -1, -1, -1, -1},

  223.     { 0,  1,  2,  3, -1, -1, -1, -1, -1},

  224.     { 2,  0,  1,  3,  4, -1, -1, -1, -1},

  225.     { 2,  3,  0,  1,  4,  5, -1, -1, -1},

  226.     { 2,  0,  1,  3,  4,  5, -1, -1, -1},

  227.     { 0,  5,  3,  4,  1,  2, -1, -1, -1},

  228.     { 3,  2,  4,  0,  1,  5,  6, -1, -1},

  229.     { 4,  5,  0,  1,  6,  2,  7,  3, -1},

  230.     { 3,  2,  4,  0,  1,  5,  7,  6, -1},

  231. };

  232. 

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

  234.     { 0,  1, -1, -1, -1, -1, -1, -1, -1},

  235.     { 0,  1,  2, -1, -1, -1, -1, -1, -1},

  236.     { 0,  1,  2, -1, -1, -1, -1, -1, -1},

  237.     { 0,  1,  2, -1, -1, -1, -1, -1, -1},

  238.     { 0,  1,  2, -1, -1, -1, -1, -1, -1},

  239.     { 2,  0,  1,  3, -1, -1, -1, -1, -1},

  240.     { 0,  1,  2,  3, -1, -1, -1, -1, -1},

  241.     { 2,  0,  1,  3,  4, -1, -1, -1, -1},

  242.     { 0,  1,  3,  4,  2, -1, -1, -1, -1},

  243.     { 2,  0,  1,  4,  5,  3, -1, -1, -1},

  244.     { 2,  3,  0,  1,  5,  6,  4, -1, -1},

  245.     { 2,  0,  1,  3,  4,  5,  6, -1, -1},

  246.     { 0,  5,  3,  4,  1,  2,  6, -1, -1},

  247.     { 3,  2,  4,  0,  1,  6,  7,  5, -1},

  248.     { 4,  5,  0,  1,  7,  2,  8,  3,  6},

  249.     { 3,  2,  4,  0,  1,  5,  8,  7,  6},

  250. };

  251. 

  252. #define DCA_DOLBY                  101           /* FIXME */

  253. 

  254. #define DCA_CHANNEL_BITS             6

  255. #define DCA_CHANNEL_MASK          0x3F

  256. 

  257. #define DCA_LFE                   0x80

  258. 

  259. #define HEADER_SIZE                 14

  260. 

  261. #define DCA_MAX_FRAME_SIZE       16384

  262. #define DCA_MAX_EXSS_HEADER_SIZE  4096

  263. 

  264. #define DCA_BUFFER_PADDING_SIZE   1024

  265. 

  266. #define DCA_NSYNCAUX        0x9A1105A0

  267. 

  268. /** Bit allocation */

  269. typedef struct {

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

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

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

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

  274. } BitAlloc;

  275. 

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

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

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

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

  280. 

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

  282.                                          int idx)

  283. {

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

  285.            ba->offset;

  286. }

  287. 

  288. typedef struct {

  289.     AVClass *class;             ///< class for AVOptions

  290.     AVCodecContext *avctx;

  291.     /* Frame header */

  292.     int frame_type;             ///< type of the current frame

  293.     int samples_deficit;        ///< deficit sample count

  294.     int crc_present;            ///< crc is present in the bitstream

  295.     int sample_blocks;          ///< number of PCM sample blocks

  296.     int frame_size;             ///< primary frame byte size

  297.     int amode;                  ///< audio channels arrangement

  298.     int sample_rate;            ///< audio sampling rate

  299.     int bit_rate;               ///< transmission bit rate

  300.     int bit_rate_index;         ///< transmission bit rate index

  301. 

  302.     int dynrange;               ///< embedded dynamic range flag

  303.     int timestamp;              ///< embedded time stamp flag

  304.     int aux_data;               ///< auxiliary data flag

  305.     int hdcd;                   ///< source material is mastered in HDCD

  306.     int ext_descr;              ///< extension audio descriptor flag

  307.     int ext_coding;             ///< extended coding flag

  308.     int aspf;                   ///< audio sync word insertion flag

  309.     int lfe;                    ///< low frequency effects flag

  310.     int predictor_history;      ///< predictor history flag

  311.     int header_crc;             ///< header crc check bytes

  312.     int multirate_inter;        ///< multirate interpolator switch

  313.     int version;                ///< encoder software revision

  314.     int copy_history;           ///< copy history

  315.     int source_pcm_res;         ///< source pcm resolution

  316.     int front_sum;              ///< front sum/difference flag

  317.     int surround_sum;           ///< surround sum/difference flag

  318.     int dialog_norm;            ///< dialog normalisation parameter

  319. 

  320.     /* Primary audio coding header */

  321.     int subframes;              ///< number of subframes

  322.     int total_channels;         ///< number of channels including extensions

  323.     int prim_channels;          ///< number of primary audio channels

  324.     int subband_activity[DCA_PRIM_CHANNELS_MAX];    ///< subband activity count

  325.     int vq_start_subband[DCA_PRIM_CHANNELS_MAX];    ///< high frequency vq start subband

  326.     int joint_intensity[DCA_PRIM_CHANNELS_MAX];     ///< joint intensity coding index

  327.     int transient_huffman[DCA_PRIM_CHANNELS_MAX];   ///< transient mode code book

  328.     int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book

  329.     int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX];    ///< bit allocation quantizer select

  330.     int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select

  331.     float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX];   ///< scale factor adjustment

  332. 

  333.     /* Primary audio coding side information */

  334.     int subsubframes[DCA_SUBFRAMES_MAX];                         ///< number of subsubframes

  335.     int partial_samples[DCA_SUBFRAMES_MAX];                      ///< partial subsubframe samples count

  336.     int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];    ///< prediction mode (ADPCM used or not)

  337.     int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];      ///< prediction VQ coefs

  338.     int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];           ///< bit allocation index

  339.     int transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];    ///< transition mode (transients)

  340.     int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2];    ///< scale factors (2 if transient)

  341.     int joint_huff[DCA_PRIM_CHANNELS_MAX];                       ///< joint subband scale factors codebook

  342.     int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors

  343.     float downmix_coef[DCA_PRIM_CHANNELS_MAX + 1][2];            ///< stereo downmix coefficients

  344.     int dynrange_coef;                                           ///< dynamic range coefficient

  345. 

  346.     /* Core substream's embedded downmix coefficients (cf. ETSI TS 102 114 V1.4.1)

  347.      * Input:  primary audio channels (incl. LFE if present)

  348.      * Output: downmix audio channels (up to 4, no LFE) */

  349.     uint8_t  core_downmix;                                       ///< embedded downmix coefficients available

  350.     uint8_t  core_downmix_amode;                                 ///< audio channel arrangement of embedded downmix

  351.     uint16_t core_downmix_codes[DCA_PRIM_CHANNELS_MAX + 1][4];   ///< embedded downmix coefficients (9-bit codes)

  352. 

  353.     int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];       ///< VQ encoded high frequency subbands

  354. 

  355.     float lfe_data[2 * DCA_LFE_MAX * (DCA_BLOCKS_MAX + 4)];      ///< Low frequency effect data

  356.     int lfe_scale_factor;

  357. 

  358.     /* Subband samples history (for ADPCM) */

  359.     DECLARE_ALIGNED(16, float, subband_samples_hist)[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];

  360.     DECLARE_ALIGNED(32, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512];

  361.     DECLARE_ALIGNED(32, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32];

  362.     int hist_index[DCA_PRIM_CHANNELS_MAX];

  363.     DECLARE_ALIGNED(32, float, raXin)[32];

  364. 

  365.     int output;                 ///< type of output

  366. 

  367.     DECLARE_ALIGNED(32, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];

  368.     float *samples_chanptr[DCA_PRIM_CHANNELS_MAX + 1];

  369.     float *extra_channels[DCA_PRIM_CHANNELS_MAX + 1];

  370.     uint8_t *extra_channels_buffer;

  371.     unsigned int extra_channels_buffer_size;

  372. 

  373.     uint8_t dca_buffer[DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE + DCA_BUFFER_PADDING_SIZE];

  374.     int dca_buffer_size;        ///< how much data is in the dca_buffer

  375. 

  376.     const int8_t *channel_order_tab;  ///< channel reordering table, lfe and non lfe

  377.     GetBitContext gb;

  378.     /* Current position in DCA frame */

  379.     int current_subframe;

  380.     int current_subsubframe;

  381. 

  382.     int core_ext_mask;          ///< present extensions in the core substream

  383. 

  384.     /* XCh extension information */

  385.     int xch_present;            ///< XCh extension present and valid

  386.     int xch_base_channel;       ///< index of first (only) channel containing XCH data

  387.     int xch_disable;            ///< whether the XCh extension should be decoded or not

  388. 

  389.     /* ExSS header parser */

  390.     int static_fields;          ///< static fields present

  391.     int mix_metadata;           ///< mixing metadata present

  392.     int num_mix_configs;        ///< number of mix out configurations

  393.     int mix_config_num_ch[4];   ///< number of channels in each mix out configuration

  394. 

  395.     int profile;

  396. 

  397.     int debug_flag;             ///< used for suppressing repeated error messages output

  398.     AVFloatDSPContext fdsp;

  399.     FFTContext imdct;

  400.     SynthFilterContext synth;

  401.     DCADSPContext dcadsp;

  402.     FmtConvertContext fmt_conv;

  403. } DCAContext;

  404. 

  405. static const uint16_t dca_vlc_offs[] = {

  406.         0,   512,   640,   768,  1282,  1794,  2436,  3080,  3770,  4454,  5364,

  407.      5372,  5380,  5388,  5392,  5396,  5412,  5420,  5428,  5460,  5492,  5508,

  408.      5572,  5604,  5668,  5796,  5860,  5892,  6412,  6668,  6796,  7308,  7564,

  409.      7820,  8076,  8620,  9132,  9388,  9910, 10166, 10680, 11196, 11726, 12240,

  410.     12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264,

  411.     18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622,

  412. };

  413. 

  414. static av_cold void dca_init_vlcs(void)

  415. {

  416.     static int vlcs_initialized = 0;

  417.     int i, j, c = 14;

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

  419. 

  420.     if (vlcs_initialized)

  421.         return;

  422. 

  423.     dca_bitalloc_index.offset = 1;

  424.     dca_bitalloc_index.wrap = 2;

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

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

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

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

  429.                  bitalloc_12_bits[i], 1, 1,

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

  431.     }

  432.     dca_scalefactor.offset = -64;

  433.     dca_scalefactor.wrap = 2;

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

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

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

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

  438.                  scales_bits[i], 1, 1,

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

  440.     }

  441.     dca_tmode.offset = 0;

  442.     dca_tmode.wrap = 1;

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

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

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

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

  447.                  tmode_bits[i], 1, 1,

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

  449.     }

  450. 

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

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

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

  454.                 break;

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

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

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

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

  459. 

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

  461.                      bitalloc_sizes[i],

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

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

  464.             c++;

  465.         }

  466.     vlcs_initialized = 1;

  467. }

  468. 

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

  470. {

  471.     while (len--)

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

  473. }

  474. 

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

  476. {

  477.     int i, j;

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

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

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

  481. 

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

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

  484. 

  485.     if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)

  486.         s->prim_channels = DCA_PRIM_CHANNELS_MAX;

  487. 

  488. 

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

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

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

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

  493.     }

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

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

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

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

  498.     }

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

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

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

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

  503. 

  504.     /* Get codebooks quantization indexes */

  505.     if (!base_channel)

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

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

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

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

  510. 

  511.     /* Get scale factor adjustment */

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

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

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

  515. 

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

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

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

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

  520. 

  521.     if (s->crc_present) {

  522.         /* Audio header CRC check */

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

  524.     }

  525. 

  526.     s->current_subframe    = 0;

  527.     s->current_subsubframe = 0;

  528. 

  529. #ifdef TRACE

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

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

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

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

  534.                s->subband_activity[i]);

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

  536.                s->vq_start_subband[i]);

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

  538.                s->joint_intensity[i]);

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

  540.                s->transient_huffman[i]);

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

  542.                s->scalefactor_huffman[i]);

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

  544.                s->bitalloc_huffman[i]);

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

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

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

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

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

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

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

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

  553.     }

  554. #endif

  555. 

  556.     return 0;

  557. }

  558. 

  559. static int dca_parse_frame_header(DCAContext *s)

  560. {

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

  562. 

  563.     /* Sync code */

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

  565. 

  566.     /* Frame header */

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

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

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

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

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

  572.     if (s->frame_size < 95)

  573.         return AVERROR_INVALIDDATA;

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

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

  576.     if (!s->sample_rate)

  577.         return AVERROR_INVALIDDATA;

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

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

  580.     if (!s->bit_rate)

  581.         return AVERROR_INVALIDDATA;

  582. 

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

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

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

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

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

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

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

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

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

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

  593. 

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

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

  596.         return AVERROR_INVALIDDATA;

  597.     }

  598. 

  599.     /* TODO: check CRC */

  600.     if (s->crc_present)

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

  602. 

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

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

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

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

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

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

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

  610. 

  611.     /* FIXME: channels mixing levels */

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

  613.     if (s->lfe)

  614.         s->output |= DCA_LFE;

  615. 

  616. #ifdef TRACE

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

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

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

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

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

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

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

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

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

  626.            s->sample_rate);

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

  628.            s->bit_rate);

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

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

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

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

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

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

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

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

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

  638.            s->predictor_history);

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

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

  641.            s->multirate_inter);

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

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

  644.     av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

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

  651. #endif

  652. 

  653.     /* Primary audio coding header */

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

  655. 

  656.     return dca_parse_audio_coding_header(s, 0);

  657. }

  658. 

  659. 

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

  661. {

  662.     if (level < 5) {

  663.         /* huffman encoded */

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

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

  666.     } else if (level < 8) {

  667.         if (level + 1 > log2range) {

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

  669.             value = get_bits(gb, log2range);

  670.         } else {

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

  672.         }

  673.     }

  674.     return value;

  675. }

  676. 

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

  678. {

  679.     /* Primary audio coding side information */

  680.     int j, k;

  681. 

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

  683.         return AVERROR_INVALIDDATA;

  684. 

  685.     if (!base_channel) {

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

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

  688.     }

  689. 

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

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

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

  693.     }

  694. 

  695.     /* Get prediction codebook */

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

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

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

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

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

  701.             }

  702.         }

  703.     }

  704. 

  705.     /* Bit allocation index */

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

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

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

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

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

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

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

  713.                 av_log(s->avctx, AV_LOG_ERROR,

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

  715.                 return AVERROR_INVALIDDATA;

  716.             } else {

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

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

  719.             }

  720. 

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

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

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

  724.                 return AVERROR_INVALIDDATA;

  725.             }

  726.         }

  727.     }

  728. 

  729.     /* Transition mode */

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

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

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

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

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

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

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

  737.             }

  738.         }

  739.     }

  740. 

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

  742.         return AVERROR_INVALIDDATA;

  743. 

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

  745.         const uint32_t *scale_table;

  746.         int scale_sum, log_size;

  747. 

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

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

  750. 

  751.         if (s->scalefactor_huffman[j] == 6) {

  752.             scale_table = scale_factor_quant7;

  753.             log_size = 7;

  754.         } else {

  755.             scale_table = scale_factor_quant6;

  756.             log_size = 6;

  757.         }

  758. 

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

  760.         scale_sum = 0;

  761. 

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

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

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

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

  766.             }

  767. 

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

  769.                 /* Get second scale factor */

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

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

  772.             }

  773.         }

  774.     }

  775. 

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

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

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

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

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

  781.     }

  782. 

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

  784.         return AVERROR_INVALIDDATA;

  785. 

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

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

  788.         int source_channel;

  789. 

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

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

  792.             int scale = 0;

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

  794. 

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

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

  797. 

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

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

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

  801.             }

  802. 

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

  804.                 av_log(s->avctx, AV_LOG_DEBUG,

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

  806.                 s->debug_flag |= 0x02;

  807.             }

  808.         }

  809.     }

  810. 

  811.     /* Dynamic range coefficient */

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

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

  814. 

  815.     /* Side information CRC check word */

  816.     if (s->crc_present) {

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

  818.     }

  819. 

  820.     /*

  821.      * Primary audio data arrays

  822.      */

  823. 

  824.     /* VQ encoded high frequency subbands */

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

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

  827.             /* 1 vector -> 32 samples */

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

  829. 

  830.     /* Low frequency effect data */

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

  832.         /* LFE samples */

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

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

  835.         float lfe_scale;

  836. 

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

  838.             /* Signed 8 bits int */

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

  840.         }

  841. 

  842.         /* Scale factor index */

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

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

  845. 

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

  847.         lfe_scale = 0.035 * s->lfe_scale_factor;

  848. 

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

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

  851.     }

  852. 

  853. #ifdef TRACE

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

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

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

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

  858. 

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

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

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

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

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

  864.     }

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

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

  867.             av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

  873.     }

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

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

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

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

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

  879.     }

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

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

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

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

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

  885.     }

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

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

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

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

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

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

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

  893.         }

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

  895.     }

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

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

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

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

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

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

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

  903.         }

  904.     }

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

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

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

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

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

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

  911. 

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

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

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

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

  916.     }

  917. #endif

  918. 

  919.     return 0;

  920. }

  921. 

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

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

  924.                             float scale)

  925. {

  926.     const float *prCoeff;

  927. 

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

  929. 

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

  931. 

  932.     /* Select filter */

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

  934.         prCoeff = fir_32bands_nonperfect;

  935.     else                        /* Perfect reconstruction */

  936.         prCoeff = fir_32bands_perfect;

  937. 

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

  939.                               s->subband_fir_hist[chans],

  940.                               &s->hist_index[chans],

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

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

  943. }

  944. 

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

  946.                                   int num_deci_sample, float *samples_in,

  947.                                   float *samples_out, float scale)

  948. {

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

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

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

  952.      *   from last subframe as history.

  953.      *

  954.      * samples_out: An array holding interpolated samples

  955.      */

  956. 

  957.     int decifactor;

  958.     const float *prCoeff;

  959.     int deciindex;

  960. 

  961.     /* Select decimation filter */

  962.     if (decimation_select == 1) {

  963.         decifactor = 64;

  964.         prCoeff = lfe_fir_128;

  965.     } else {

  966.         decifactor = 32;

  967.         prCoeff = lfe_fir_64;

  968.     }

  969.     /* Interpolation */

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

  971.         s->dcadsp.lfe_fir(samples_out, samples_in, prCoeff, decifactor, scale);

  972.         samples_in++;

  973.         samples_out += 2 * decifactor;

  974.     }

  975. }

  976. 

  977. /* downmixing routines */

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

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

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

  981. 

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

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

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

  985. 

  986. #define MIX_FRONT3(samples, coef)                                      \

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

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

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

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

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

  992. 

  993. #define DOWNMIX_TO_STEREO(op1, op2)             \

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

  995.         op1                                     \

  996.         op2                                     \

  997.     }

  998. 

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

  1000.                         float coef[DCA_PRIM_CHANNELS_MAX + 1][2],

  1001.                         const int8_t *channel_mapping)

  1002. {

  1003.     int c, l, r, sl, sr, s;

  1004.     int i;

  1005.     float t, u, v;

  1006. 

  1007.     switch (srcfmt) {

  1008.     case DCA_MONO:

  1009.     case DCA_4F2R:

  1010.         av_log(NULL, 0, "Not implemented!\n");

  1011.         break;

  1012.     case DCA_CHANNEL:

  1013.     case DCA_STEREO:

  1014.     case DCA_STEREO_TOTAL:

  1015.     case DCA_STEREO_SUMDIFF:

  1016.         break;

  1017.     case DCA_3F:

  1018.         c = channel_mapping[0];

  1019.         l = channel_mapping[1];

  1020.         r = channel_mapping[2];

  1021.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );

  1022.         break;

  1023.     case DCA_2F1R:

  1024.         s = channel_mapping[2];

  1025.         DOWNMIX_TO_STEREO(MIX_REAR1(samples, s, 2, coef), );

  1026.         break;

  1027.     case DCA_3F1R:

  1028.         c = channel_mapping[0];

  1029.         l = channel_mapping[1];

  1030.         r = channel_mapping[2];

  1031.         s = channel_mapping[3];

  1032.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1033.                           MIX_REAR1(samples, s, 3, coef));

  1034.         break;

  1035.     case DCA_2F2R:

  1036.         sl = channel_mapping[2];

  1037.         sr = channel_mapping[3];

  1038.         DOWNMIX_TO_STEREO(MIX_REAR2(samples, sl, sr, 2, coef), );

  1039.         break;

  1040.     case DCA_3F2R:

  1041.         c  = channel_mapping[0];

  1042.         l  = channel_mapping[1];

  1043.         r  = channel_mapping[2];

  1044.         sl = channel_mapping[3];

  1045.         sr = channel_mapping[4];

  1046.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1047.                           MIX_REAR2(samples, sl, sr, 3, coef));

  1048.         break;

  1049.     }

  1050.     if (lfe_present) {

  1051.         int lf_buf = dca_lfe_index[srcfmt];

  1052.         int lf_idx = dca_channels [srcfmt];

  1053.         for (i = 0; i < 256; i++) {

  1054.             samples[0][i] += samples[lf_buf][i] * coef[lf_idx][0];

  1055.             samples[1][i] += samples[lf_buf][i] * coef[lf_idx][1];

  1056.         }

  1057.     }

  1058. }

  1059. 

  1060. 

  1061. #ifndef decode_blockcodes

  1062. /* Very compact version of the block code decoder that does not use table

  1063.  * look-up but is slightly slower */

  1064. static int decode_blockcode(int code, int levels, int32_t *values)

  1065. {

  1066.     int i;

  1067.     int offset = (levels - 1) >> 1;

  1068. 

  1069.     for (i = 0; i < 4; i++) {

  1070.         int div = FASTDIV(code, levels);

  1071.         values[i] = code - offset - div * levels;

  1072.         code = div;

  1073.     }

  1074. 

  1075.     return code;

  1076. }

  1077. 

  1078. static int decode_blockcodes(int code1, int code2, int levels, int32_t *values)

  1079. {

  1080.     return decode_blockcode(code1, levels, values) |

  1081.            decode_blockcode(code2, levels, values + 4);

  1082. }

  1083. #endif

  1084. 

  1085. static const uint8_t abits_sizes[7]  = { 7, 10, 12, 13, 15, 17, 19 };

  1086. static const uint8_t abits_levels[7] = { 3,  5,  7,  9, 13, 17, 25 };

  1087. 

  1088. #ifndef int8x8_fmul_int32

  1089. static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)

  1090. {

  1091.     float fscale = scale / 16.0;

  1092.     int i;

  1093.     for (i = 0; i < 8; i++)

  1094.         dst[i] = src[i] * fscale;

  1095. }

  1096. #endif

  1097. 

  1098. static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)

  1099. {

  1100.     int k, l;

  1101.     int subsubframe = s->current_subsubframe;

  1102. 

  1103.     const float *quant_step_table;

  1104. 

  1105.     /* FIXME */

  1106.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1107.     LOCAL_ALIGNED_16(int32_t, block, [8 * DCA_SUBBANDS]);

  1108. 

  1109.     /*

  1110.      * Audio data

  1111.      */

  1112. 

  1113.     /* Select quantization step size table */

  1114.     if (s->bit_rate_index == 0x1f)

  1115.         quant_step_table = lossless_quant_d;

  1116.     else

  1117.         quant_step_table = lossy_quant_d;

  1118. 

  1119.     for (k = base_channel; k < s->prim_channels; k++) {

  1120.         float rscale[DCA_SUBBANDS];

  1121. 

  1122.         if (get_bits_left(&s->gb) < 0)

  1123.             return AVERROR_INVALIDDATA;

  1124. 

  1125.         for (l = 0; l < s->vq_start_subband[k]; l++) {

  1126.             int m;

  1127. 

  1128.             /* Select the mid-tread linear quantizer */

  1129.             int abits = s->bitalloc[k][l];

  1130. 

  1131.             float quant_step_size = quant_step_table[abits];

  1132. 

  1133.             /*

  1134.              * Determine quantization index code book and its type

  1135.              */

  1136. 

  1137.             /* Select quantization index code book */

  1138.             int sel = s->quant_index_huffman[k][abits];

  1139. 

  1140.             /*

  1141.              * Extract bits from the bit stream

  1142.              */

  1143.             if (!abits) {

  1144.                 rscale[l] = 0;

  1145.                 memset(block + 8 * l, 0, 8 * sizeof(block[0]));

  1146.             } else {

  1147.                 /* Deal with transients */

  1148.                 int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];

  1149.                 rscale[l] = quant_step_size * s->scale_factor[k][l][sfi] *

  1150.                                s->scalefactor_adj[k][sel];

  1151. 

  1152.                 if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {

  1153.                     if (abits <= 7) {

  1154.                         /* Block code */

  1155.                         int block_code1, block_code2, size, levels, err;

  1156. 

  1157.                         size   = abits_sizes[abits - 1];

  1158.                         levels = abits_levels[abits - 1];

  1159. 

  1160.                         block_code1 = get_bits(&s->gb, size);

  1161.                         block_code2 = get_bits(&s->gb, size);

  1162.                         err = decode_blockcodes(block_code1, block_code2,

  1163.                                                 levels, block + 8 * l);

  1164.                         if (err) {

  1165.                             av_log(s->avctx, AV_LOG_ERROR,

  1166.                                    "ERROR: block code look-up failed\n");

  1167.                             return AVERROR_INVALIDDATA;

  1168.                         }

  1169.                     } else {

  1170.                         /* no coding */

  1171.                         for (m = 0; m < 8; m++)

  1172.                             block[8 * l + m] = get_sbits(&s->gb, abits - 3);

  1173.                     }

  1174.                 } else {

  1175.                     /* Huffman coded */

  1176.                     for (m = 0; m < 8; m++)

  1177.                         block[8 * l + m] = get_bitalloc(&s->gb,

  1178.                                                 &dca_smpl_bitalloc[abits], sel);

  1179.                 }

  1180. 

  1181.             }

  1182.         }

  1183. 

  1184.         s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[k][0],

  1185.                                                block, rscale, 8 * s->vq_start_subband[k]);

  1186. 

  1187.         for (l = 0; l < s->vq_start_subband[k]; l++) {

  1188.             int m;

  1189.             /*

  1190.              * Inverse ADPCM if in prediction mode

  1191.              */

  1192.             if (s->prediction_mode[k][l]) {

  1193.                 int n;

  1194.                 for (m = 0; m < 8; m++) {

  1195.                     for (n = 1; n <= 4; n++)

  1196.                         if (m >= n)

  1197.                             subband_samples[k][l][m] +=

  1198.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1199.                                  subband_samples[k][l][m - n] / 8192);

  1200.                         else if (s->predictor_history)

  1201.                             subband_samples[k][l][m] +=

  1202.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1203.                                  s->subband_samples_hist[k][l][m - n + 4] / 8192);

  1204.                 }

  1205.             }

  1206.         }

  1207. 

  1208.         /*

  1209.          * Decode VQ encoded high frequencies

  1210.          */

  1211.         for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {

  1212.             /* 1 vector -> 32 samples but we only need the 8 samples

  1213.              * for this subsubframe. */

  1214.             int hfvq = s->high_freq_vq[k][l];

  1215. 

  1216.             if (!s->debug_flag & 0x01) {

  1217.                 av_log(s->avctx, AV_LOG_DEBUG,

  1218.                        "Stream with high frequencies VQ coding\n");

  1219.                 s->debug_flag |= 0x01;

  1220.             }

  1221. 

  1222.             int8x8_fmul_int32(subband_samples[k][l],

  1223.                               &high_freq_vq[hfvq][subsubframe * 8],

  1224.                               s->scale_factor[k][l][0]);

  1225.         }

  1226.     }

  1227. 

  1228.     /* Check for DSYNC after subsubframe */

  1229.     if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {

  1230.         if (0xFFFF == get_bits(&s->gb, 16)) {   /* 0xFFFF */

  1231. #ifdef TRACE

  1232.             av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");

  1233. #endif

  1234.         } else {

  1235.             av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");

  1236.             return AVERROR_INVALIDDATA;

  1237.         }

  1238.     }

  1239. 

  1240.     /* Backup predictor history for adpcm */

  1241.     for (k = base_channel; k < s->prim_channels; k++)

  1242.         for (l = 0; l < s->vq_start_subband[k]; l++)

  1243.             memcpy(s->subband_samples_hist[k][l],

  1244.                    &subband_samples[k][l][4],

  1245.                    4 * sizeof(subband_samples[0][0][0]));

  1246. 

  1247.     return 0;

  1248. }

  1249. 

  1250. static int dca_filter_channels(DCAContext *s, int block_index)

  1251. {

  1252.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1253.     int k;

  1254. 

  1255.     /* 32 subbands QMF */

  1256.     for (k = 0; k < s->prim_channels; k++) {

  1257. /*        static float pcm_to_double[8] = { 32768.0, 32768.0, 524288.0, 524288.0,

  1258.                                             0, 8388608.0, 8388608.0 };*/

  1259.         if (s->channel_order_tab[k] >= 0)

  1260.             qmf_32_subbands(s, k, subband_samples[k],

  1261.                             s->samples_chanptr[s->channel_order_tab[k]],

  1262.                             M_SQRT1_2 / 32768.0 /* pcm_to_double[s->source_pcm_res] */);

  1263.     }

  1264. 

  1265.     /* Generate LFE samples for this subsubframe FIXME!!! */

  1266.     if (s->lfe) {

  1267.         lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,

  1268.                               s->lfe_data + 2 * s->lfe * (block_index + 4),

  1269.                               s->samples_chanptr[dca_lfe_index[s->amode]],

  1270.                               1.0 / (256.0 * 32768.0));

  1271.         /* Outputs 20bits pcm samples */

  1272.     }

  1273. 

  1274.     /* Downmixing to Stereo */

  1275.     if (s->prim_channels + !!s->lfe > 2 &&

  1276.         s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {

  1277.         dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef,

  1278.                     s->channel_order_tab);

  1279.     }

  1280. 

  1281.     return 0;

  1282. }

  1283. 

  1284. 

  1285. static int dca_subframe_footer(DCAContext *s, int base_channel)

  1286. {

  1287.     int in, out, aux_data_count, aux_data_end, reserved;

  1288.     uint32_t nsyncaux;

  1289. 

  1290.     /*

  1291.      * Unpack optional information

  1292.      */

  1293. 

  1294.     /* presumably optional information only appears in the core? */

  1295.     if (!base_channel) {

  1296.         if (s->timestamp)

  1297.             skip_bits_long(&s->gb, 32);

  1298. 

  1299.         if (s->aux_data) {

  1300.             aux_data_count = get_bits(&s->gb, 6);

  1301. 

  1302.             // align (32-bit)

  1303.             skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1304. 

  1305.             aux_data_end = 8 * aux_data_count + get_bits_count(&s->gb);

  1306. 

  1307.             if ((nsyncaux = get_bits_long(&s->gb, 32)) != DCA_NSYNCAUX) {

  1308.                 av_log(s->avctx, AV_LOG_ERROR, "nSYNCAUX mismatch %#"PRIx32"\n",

  1309.                        nsyncaux);

  1310.                 return AVERROR_INVALIDDATA;

  1311.             }

  1312. 

  1313.             if (get_bits1(&s->gb)) { // bAUXTimeStampFlag

  1314.                 avpriv_request_sample(s->avctx,

  1315.                                       "Auxiliary Decode Time Stamp Flag");

  1316.                 // align (4-bit)

  1317.                 skip_bits(&s->gb, (-get_bits_count(&s->gb)) & 4);

  1318.                 // 44 bits: nMSByte (8), nMarker (4), nLSByte (28), nMarker (4)

  1319.                 skip_bits_long(&s->gb, 44);

  1320.             }

  1321. 

  1322.             if ((s->core_downmix = get_bits1(&s->gb))) {

  1323.                 int am = get_bits(&s->gb, 3);

  1324.                 switch (am) {

  1325.                 case 0:

  1326.                     s->core_downmix_amode = DCA_MONO;

  1327.                     break;

  1328.                 case 1:

  1329.                     s->core_downmix_amode = DCA_STEREO;

  1330.                     break;

  1331.                 case 2:

  1332.                     s->core_downmix_amode = DCA_STEREO_TOTAL;

  1333.                     break;

  1334.                 case 3:

  1335.                     s->core_downmix_amode = DCA_3F;

  1336.                     break;

  1337.                 case 4:

  1338.                     s->core_downmix_amode = DCA_2F1R;

  1339.                     break;

  1340.                 case 5:

  1341.                     s->core_downmix_amode = DCA_2F2R;

  1342.                     break;

  1343.                 case 6:

  1344.                     s->core_downmix_amode = DCA_3F1R;

  1345.                     break;

  1346.                 default:

  1347.                     av_log(s->avctx, AV_LOG_ERROR,

  1348.                            "Invalid mode %d for embedded downmix coefficients\n",

  1349.                            am);

  1350.                     return AVERROR_INVALIDDATA;

  1351.                 }

  1352.                 for (out = 0; out < dca_channels[s->core_downmix_amode]; out++) {

  1353.                     for (in = 0; in < s->prim_channels + !!s->lfe; in++) {

  1354.                         uint16_t tmp = get_bits(&s->gb, 9);

  1355.                         if ((tmp & 0xFF) > 241) {

  1356.                             av_log(s->avctx, AV_LOG_ERROR,

  1357.                                    "Invalid downmix coefficient code %"PRIu16"\n",

  1358.                                    tmp);

  1359.                             return AVERROR_INVALIDDATA;

  1360.                         }

  1361.                         s->core_downmix_codes[in][out] = tmp;

  1362.                     }

  1363.                 }

  1364.             }

  1365. 

  1366.             align_get_bits(&s->gb); // byte align

  1367.             skip_bits(&s->gb, 16);  // nAUXCRC16

  1368. 

  1369.             // additional data (reserved, cf. ETSI TS 102 114 V1.4.1)

  1370.             if ((reserved = (aux_data_end - get_bits_count(&s->gb))) < 0) {

  1371.                  av_log(s->avctx, AV_LOG_ERROR,

  1372.                         "Overread auxiliary data by %d bits\n", -reserved);

  1373.                 return AVERROR_INVALIDDATA;

  1374.             } else if (reserved) {

  1375.                 avpriv_request_sample(s->avctx,

  1376.                                       "Core auxiliary data reserved content");

  1377.                 skip_bits_long(&s->gb, reserved);

  1378.             }

  1379.         }

  1380. 

  1381.         if (s->crc_present && s->dynrange)

  1382.             get_bits(&s->gb, 16);

  1383.     }

  1384. 

  1385.     return 0;

  1386. }

  1387. 

  1388. /**

  1389.  * Decode a dca frame block

  1390.  *

  1391.  * @param s     pointer to the DCAContext

  1392.  */

  1393. 

  1394. static int dca_decode_block(DCAContext *s, int base_channel, int block_index)

  1395. {

  1396.     int ret;

  1397. 

  1398.     /* Sanity check */

  1399.     if (s->current_subframe >= s->subframes) {

  1400.         av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",

  1401.                s->current_subframe, s->subframes);

  1402.         return AVERROR_INVALIDDATA;

  1403.     }

  1404. 

  1405.     if (!s->current_subsubframe) {

  1406. #ifdef TRACE

  1407.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");

  1408. #endif

  1409.         /* Read subframe header */

  1410.         if ((ret = dca_subframe_header(s, base_channel, block_index)))

  1411.             return ret;

  1412.     }

  1413. 

  1414.     /* Read subsubframe */

  1415. #ifdef TRACE

  1416.     av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");

  1417. #endif

  1418.     if ((ret = dca_subsubframe(s, base_channel, block_index)))

  1419.         return ret;

  1420. 

  1421.     /* Update state */

  1422.     s->current_subsubframe++;

  1423.     if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {

  1424.         s->current_subsubframe = 0;

  1425.         s->current_subframe++;

  1426.     }

  1427.     if (s->current_subframe >= s->subframes) {

  1428. #ifdef TRACE

  1429.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");

  1430. #endif

  1431.         /* Read subframe footer */

  1432.         if ((ret = dca_subframe_footer(s, base_channel)))

  1433.             return ret;

  1434.     }

  1435. 

  1436.     return 0;

  1437. }

  1438. 

  1439. /**

  1440.  * Return the number of channels in an ExSS speaker mask (HD)

  1441.  */

  1442. static int dca_exss_mask2count(int mask)

  1443. {

  1444.     /* count bits that mean speaker pairs twice */

  1445.     return av_popcount(mask) +

  1446.            av_popcount(mask & (DCA_EXSS_CENTER_LEFT_RIGHT      |

  1447.                                DCA_EXSS_FRONT_LEFT_RIGHT       |

  1448.                                DCA_EXSS_FRONT_HIGH_LEFT_RIGHT  |

  1449.                                DCA_EXSS_WIDE_LEFT_RIGHT        |

  1450.                                DCA_EXSS_SIDE_LEFT_RIGHT        |

  1451.                                DCA_EXSS_SIDE_HIGH_LEFT_RIGHT   |

  1452.                                DCA_EXSS_SIDE_REAR_LEFT_RIGHT   |

  1453.                                DCA_EXSS_REAR_LEFT_RIGHT        |

  1454.                                DCA_EXSS_REAR_HIGH_LEFT_RIGHT));

  1455. }

  1456. 

  1457. /**

  1458.  * Skip mixing coefficients of a single mix out configuration (HD)

  1459.  */

  1460. static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch)

  1461. {

  1462.     int i;

  1463. 

  1464.     for (i = 0; i < channels; i++) {

  1465.         int mix_map_mask = get_bits(gb, out_ch);

  1466.         int num_coeffs = av_popcount(mix_map_mask);

  1467.         skip_bits_long(gb, num_coeffs * 6);

  1468.     }

  1469. }

  1470. 

  1471. /**

  1472.  * Parse extension substream asset header (HD)

  1473.  */

  1474. static int dca_exss_parse_asset_header(DCAContext *s)

  1475. {

  1476.     int header_pos = get_bits_count(&s->gb);

  1477.     int header_size;

  1478.     int channels;

  1479.     int embedded_stereo = 0;

  1480.     int embedded_6ch    = 0;

  1481.     int drc_code_present;

  1482.     int extensions_mask;

  1483.     int i, j;

  1484. 

  1485.     if (get_bits_left(&s->gb) < 16)

  1486.         return -1;

  1487. 

  1488.     /* We will parse just enough to get to the extensions bitmask with which

  1489.      * we can set the profile value. */

  1490. 

  1491.     header_size = get_bits(&s->gb, 9) + 1;

  1492.     skip_bits(&s->gb, 3); // asset index

  1493. 

  1494.     if (s->static_fields) {

  1495.         if (get_bits1(&s->gb))

  1496.             skip_bits(&s->gb, 4); // asset type descriptor

  1497.         if (get_bits1(&s->gb))

  1498.             skip_bits_long(&s->gb, 24); // language descriptor

  1499. 

  1500.         if (get_bits1(&s->gb)) {

  1501.             /* How can one fit 1024 bytes of text here if the maximum value

  1502.              * for the asset header size field above was 512 bytes? */

  1503.             int text_length = get_bits(&s->gb, 10) + 1;

  1504.             if (get_bits_left(&s->gb) < text_length * 8)

  1505.                 return -1;

  1506.             skip_bits_long(&s->gb, text_length * 8); // info text

  1507.         }

  1508. 

  1509.         skip_bits(&s->gb, 5); // bit resolution - 1

  1510.         skip_bits(&s->gb, 4); // max sample rate code

  1511.         channels = get_bits(&s->gb, 8) + 1;

  1512. 

  1513.         if (get_bits1(&s->gb)) { // 1-to-1 channels to speakers

  1514.             int spkr_remap_sets;

  1515.             int spkr_mask_size = 16;

  1516.             int num_spkrs[7];

  1517. 

  1518.             if (channels > 2)

  1519.                 embedded_stereo = get_bits1(&s->gb);

  1520.             if (channels > 6)

  1521.                 embedded_6ch = get_bits1(&s->gb);

  1522. 

  1523.             if (get_bits1(&s->gb)) {

  1524.                 spkr_mask_size = (get_bits(&s->gb, 2) + 1) << 2;

  1525.                 skip_bits(&s->gb, spkr_mask_size); // spkr activity mask

  1526.             }

  1527. 

  1528.             spkr_remap_sets = get_bits(&s->gb, 3);

  1529. 

  1530.             for (i = 0; i < spkr_remap_sets; i++) {

  1531.                 /* std layout mask for each remap set */

  1532.                 num_spkrs[i] = dca_exss_mask2count(get_bits(&s->gb, spkr_mask_size));

  1533.             }

  1534. 

  1535.             for (i = 0; i < spkr_remap_sets; i++) {

  1536.                 int num_dec_ch_remaps = get_bits(&s->gb, 5) + 1;

  1537.                 if (get_bits_left(&s->gb) < 0)

  1538.                     return -1;

  1539. 

  1540.                 for (j = 0; j < num_spkrs[i]; j++) {

  1541.                     int remap_dec_ch_mask = get_bits_long(&s->gb, num_dec_ch_remaps);

  1542.                     int num_dec_ch = av_popcount(remap_dec_ch_mask);

  1543.                     skip_bits_long(&s->gb, num_dec_ch * 5); // remap codes

  1544.                 }

  1545.             }

  1546. 

  1547.         } else {

  1548.             skip_bits(&s->gb, 3); // representation type

  1549.         }

  1550.     }

  1551. 

  1552.     drc_code_present = get_bits1(&s->gb);

  1553.     if (drc_code_present)

  1554.         get_bits(&s->gb, 8); // drc code

  1555. 

  1556.     if (get_bits1(&s->gb))

  1557.         skip_bits(&s->gb, 5); // dialog normalization code

  1558. 

  1559.     if (drc_code_present && embedded_stereo)

  1560.         get_bits(&s->gb, 8); // drc stereo code

  1561. 

  1562.     if (s->mix_metadata && get_bits1(&s->gb)) {

  1563.         skip_bits(&s->gb, 1); // external mix

  1564.         skip_bits(&s->gb, 6); // post mix gain code

  1565. 

  1566.         if (get_bits(&s->gb, 2) != 3) // mixer drc code

  1567.             skip_bits(&s->gb, 3); // drc limit

  1568.         else

  1569.             skip_bits(&s->gb, 8); // custom drc code

  1570. 

  1571.         if (get_bits1(&s->gb)) // channel specific scaling

  1572.             for (i = 0; i < s->num_mix_configs; i++)

  1573.                 skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6); // scale codes

  1574.         else

  1575.             skip_bits_long(&s->gb, s->num_mix_configs * 6); // scale codes

  1576. 

  1577.         for (i = 0; i < s->num_mix_configs; i++) {

  1578.             if (get_bits_left(&s->gb) < 0)

  1579.                 return -1;

  1580.             dca_exss_skip_mix_coeffs(&s->gb, channels, s->mix_config_num_ch[i]);

  1581.             if (embedded_6ch)

  1582.                 dca_exss_skip_mix_coeffs(&s->gb, 6, s->mix_config_num_ch[i]);

  1583.             if (embedded_stereo)

  1584.                 dca_exss_skip_mix_coeffs(&s->gb, 2, s->mix_config_num_ch[i]);

  1585.         }

  1586.     }

  1587. 

  1588.     switch (get_bits(&s->gb, 2)) {

  1589.     case 0: extensions_mask = get_bits(&s->gb, 12); break;

  1590.     case 1: extensions_mask = DCA_EXT_EXSS_XLL;     break;

  1591.     case 2: extensions_mask = DCA_EXT_EXSS_LBR;     break;

  1592.     case 3: extensions_mask = 0; /* aux coding */   break;

  1593.     }

  1594. 

  1595.     /* not parsed further, we were only interested in the extensions mask */

  1596. 

  1597.     if (get_bits_left(&s->gb) < 0)

  1598.         return -1;

  1599. 

  1600.     if (get_bits_count(&s->gb) - header_pos > header_size * 8) {

  1601.         av_log(s->avctx, AV_LOG_WARNING, "Asset header size mismatch.\n");

  1602.         return -1;

  1603.     }

  1604.     skip_bits_long(&s->gb, header_pos + header_size * 8 - get_bits_count(&s->gb));

  1605. 

  1606.     if (extensions_mask & DCA_EXT_EXSS_XLL)

  1607.         s->profile = FF_PROFILE_DTS_HD_MA;

  1608.     else if (extensions_mask & (DCA_EXT_EXSS_XBR | DCA_EXT_EXSS_X96 |

  1609.                                 DCA_EXT_EXSS_XXCH))

  1610.         s->profile = FF_PROFILE_DTS_HD_HRA;

  1611. 

  1612.     if (!(extensions_mask & DCA_EXT_CORE))

  1613.         av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n");

  1614.     if ((extensions_mask & DCA_CORE_EXTS) != s->core_ext_mask)

  1615.         av_log(s->avctx, AV_LOG_WARNING,

  1616.                "DTS extensions detection mismatch (%d, %d)\n",

  1617.                extensions_mask & DCA_CORE_EXTS, s->core_ext_mask);

  1618. 

  1619.     return 0;

  1620. }

  1621. 

  1622. /**

  1623.  * Parse extension substream header (HD)

  1624.  */

  1625. static void dca_exss_parse_header(DCAContext *s)

  1626. {

  1627.     int ss_index;

  1628.     int blownup;

  1629.     int num_audiop = 1;

  1630.     int num_assets = 1;

  1631.     int active_ss_mask[8];

  1632.     int i, j;

  1633. 

  1634.     if (get_bits_left(&s->gb) < 52)

  1635.         return;

  1636. 

  1637.     skip_bits(&s->gb, 8); // user data

  1638.     ss_index = get_bits(&s->gb, 2);

  1639. 

  1640.     blownup = get_bits1(&s->gb);

  1641.     skip_bits(&s->gb,  8 + 4 * blownup); // header_size

  1642.     skip_bits(&s->gb, 16 + 4 * blownup); // hd_size

  1643. 

  1644.     s->static_fields = get_bits1(&s->gb);

  1645.     if (s->static_fields) {

  1646.         skip_bits(&s->gb, 2); // reference clock code

  1647.         skip_bits(&s->gb, 3); // frame duration code

  1648. 

  1649.         if (get_bits1(&s->gb))

  1650.             skip_bits_long(&s->gb, 36); // timestamp

  1651. 

  1652.         /* a single stream can contain multiple audio assets that can be

  1653.          * combined to form multiple audio presentations */

  1654. 

  1655.         num_audiop = get_bits(&s->gb, 3) + 1;

  1656.         if (num_audiop > 1) {

  1657.             avpriv_request_sample(s->avctx,

  1658.                                   "Multiple DTS-HD audio presentations");

  1659.             /* ignore such streams for now */

  1660.             return;

  1661.         }

  1662. 

  1663.         num_assets = get_bits(&s->gb, 3) + 1;

  1664.         if (num_assets > 1) {

  1665.             avpriv_request_sample(s->avctx, "Multiple DTS-HD audio assets");

  1666.             /* ignore such streams for now */

  1667.             return;

  1668.         }

  1669. 

  1670.         for (i = 0; i < num_audiop; i++)

  1671.             active_ss_mask[i] = get_bits(&s->gb, ss_index + 1);

  1672. 

  1673.         for (i = 0; i < num_audiop; i++)

  1674.             for (j = 0; j <= ss_index; j++)

  1675.                 if (active_ss_mask[i] & (1 << j))

  1676.                     skip_bits(&s->gb, 8); // active asset mask

  1677. 

  1678.         s->mix_metadata = get_bits1(&s->gb);

  1679.         if (s->mix_metadata) {

  1680.             int mix_out_mask_size;

  1681. 

  1682.             skip_bits(&s->gb, 2); // adjustment level

  1683.             mix_out_mask_size  = (get_bits(&s->gb, 2) + 1) << 2;

  1684.             s->num_mix_configs =  get_bits(&s->gb, 2) + 1;

  1685. 

  1686.             for (i = 0; i < s->num_mix_configs; i++) {

  1687.                 int mix_out_mask        = get_bits(&s->gb, mix_out_mask_size);

  1688.                 s->mix_config_num_ch[i] = dca_exss_mask2count(mix_out_mask);

  1689.             }

  1690.         }

  1691.     }

  1692. 

  1693.     for (i = 0; i < num_assets; i++)

  1694.         skip_bits_long(&s->gb, 16 + 4 * blownup);  // asset size

  1695. 

  1696.     for (i = 0; i < num_assets; i++) {

  1697.         if (dca_exss_parse_asset_header(s))

  1698.             return;

  1699.     }

  1700. 

  1701.     /* not parsed further, we were only interested in the extensions mask

  1702.      * from the asset header */

  1703. }

  1704. 

  1705. /**

  1706.  * Main frame decoding function

  1707.  * FIXME add arguments

  1708.  */

  1709. static int dca_decode_frame(AVCodecContext *avctx, void *data,

  1710.                             int *got_frame_ptr, AVPacket *avpkt)

  1711. {

  1712.     AVFrame *frame     = data;

  1713.     const uint8_t *buf = avpkt->data;

  1714.     int buf_size = avpkt->size;

  1715. 

  1716.     int lfe_samples;

  1717.     int num_core_channels = 0;

  1718.     int i, ret;

  1719.     float  **samples_flt;

  1720.     DCAContext *s = avctx->priv_data;

  1721.     int channels, full_channels;

  1722.     int core_ss_end;

  1723. 

  1724. 

  1725.     s->xch_present = 0;

  1726. 

  1727.     s->dca_buffer_size = ff_dca_convert_bitstream(buf, buf_size, s->dca_buffer,

  1728.                                                   DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);

  1729.     if (s->dca_buffer_size == AVERROR_INVALIDDATA) {

  1730.         av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");

  1731.         return AVERROR_INVALIDDATA;

  1732.     }

  1733. 

  1734.     init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);

  1735.     if ((ret = dca_parse_frame_header(s)) < 0) {

  1736.         //seems like the frame is corrupt, try with the next one

  1737.         return ret;

  1738.     }

  1739.     //set AVCodec values with parsed data

  1740.     avctx->sample_rate = s->sample_rate;

  1741.     avctx->bit_rate    = s->bit_rate;

  1742. 

  1743.     s->profile = FF_PROFILE_DTS;

  1744. 

  1745.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1746.         if ((ret = dca_decode_block(s, 0, i))) {

  1747.             av_log(avctx, AV_LOG_ERROR, "error decoding block\n");

  1748.             return ret;

  1749.         }

  1750.     }

  1751. 

  1752.     /* record number of core channels incase less than max channels are requested */

  1753.     num_core_channels = s->prim_channels;

  1754. 

  1755.     if (s->ext_coding)

  1756.         s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];

  1757.     else

  1758.         s->core_ext_mask = 0;

  1759. 

  1760.     core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;

  1761. 

  1762.     /* only scan for extensions if ext_descr was unknown or indicated a

  1763.      * supported XCh extension */

  1764.     if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {

  1765. 

  1766.         /* if ext_descr was unknown, clear s->core_ext_mask so that the

  1767.          * extensions scan can fill it up */

  1768.         s->core_ext_mask = FFMAX(s->core_ext_mask, 0);

  1769. 

  1770.         /* extensions start at 32-bit boundaries into bitstream */

  1771.         skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1772. 

  1773.         while (core_ss_end - get_bits_count(&s->gb) >= 32) {

  1774.             uint32_t bits = get_bits_long(&s->gb, 32);

  1775. 

  1776.             switch (bits) {

  1777.             case 0x5a5a5a5a: {

  1778.                 int ext_amode, xch_fsize;

  1779. 

  1780.                 s->xch_base_channel = s->prim_channels;

  1781. 

  1782.                 /* validate sync word using XCHFSIZE field */

  1783.                 xch_fsize = show_bits(&s->gb, 10);

  1784.                 if ((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&

  1785.                     (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))

  1786.                     continue;

  1787. 

  1788.                 /* skip length-to-end-of-frame field for the moment */

  1789.                 skip_bits(&s->gb, 10);

  1790. 

  1791.                 s->core_ext_mask |= DCA_EXT_XCH;

  1792. 

  1793.                 /* extension amode(number of channels in extension) should be 1 */

  1794.                 /* AFAIK XCh is not used for more channels */

  1795.                 if ((ext_amode = get_bits(&s->gb, 4)) != 1) {

  1796.                     av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"

  1797.                            " supported!\n", ext_amode);

  1798.                     continue;

  1799.                 }

  1800. 

  1801.                 /* much like core primary audio coding header */

  1802.                 dca_parse_audio_coding_header(s, s->xch_base_channel);

  1803. 

  1804.                 for (i = 0; i < (s->sample_blocks / 8); i++)

  1805.                     if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {

  1806.                         av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");

  1807.                         continue;

  1808.                     }

  1809. 

  1810.                 s->xch_present = 1;

  1811.                 break;

  1812.             }

  1813.             case 0x47004a03:

  1814.                 /* XXCh: extended channels */

  1815.                 /* usually found either in core or HD part in DTS-HD HRA streams,

  1816.                  * but not in DTS-ES which contains XCh extensions instead */

  1817.                 s->core_ext_mask |= DCA_EXT_XXCH;

  1818.                 break;

  1819. 

  1820.             case 0x1d95f262: {

  1821.                 int fsize96 = show_bits(&s->gb, 12) + 1;

  1822.                 if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)

  1823.                     continue;

  1824. 

  1825.                 av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n",

  1826.                        get_bits_count(&s->gb));

  1827.                 skip_bits(&s->gb, 12);

  1828.                 av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);

  1829.                 av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));

  1830. 

  1831.                 s->core_ext_mask |= DCA_EXT_X96;

  1832.                 break;

  1833.             }

  1834.             }

  1835. 

  1836.             skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1837.         }

  1838.     } else {

  1839.         /* no supported extensions, skip the rest of the core substream */

  1840.         skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));

  1841.     }

  1842. 

  1843.     if (s->core_ext_mask & DCA_EXT_X96)

  1844.         s->profile = FF_PROFILE_DTS_96_24;

  1845.     else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))

  1846.         s->profile = FF_PROFILE_DTS_ES;

  1847. 

  1848.     /* check for ExSS (HD part) */

  1849.     if (s->dca_buffer_size - s->frame_size > 32 &&

  1850.         get_bits_long(&s->gb, 32) == DCA_HD_MARKER)

  1851.         dca_exss_parse_header(s);

  1852. 

  1853.     avctx->profile = s->profile;

  1854. 

  1855.     full_channels = channels = s->prim_channels + !!s->lfe;

  1856. 

  1857.     if (s->amode < 16) {

  1858.         avctx->channel_layout = dca_core_channel_layout[s->amode];

  1859. 

  1860. #if FF_API_REQUEST_CHANNELS

  1861. FF_DISABLE_DEPRECATION_WARNINGS

  1862.         if (s->xch_present && !s->xch_disable &&

  1863.             (!avctx->request_channels ||

  1864.              avctx->request_channels > num_core_channels + !!s->lfe)) {

  1865. FF_ENABLE_DEPRECATION_WARNINGS

  1866. #else

  1867.         if (s->xch_present && !s->xch_disable) {

  1868. #endif

  1869.             avctx->channel_layout |= AV_CH_BACK_CENTER;

  1870.             if (s->lfe) {

  1871.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1872.                 s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode];

  1873.             } else {

  1874.                 s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];

  1875.             }

  1876.         } else {

  1877.             channels = num_core_channels + !!s->lfe;

  1878.             s->xch_present = 0; /* disable further xch processing */

  1879.             if (s->lfe) {

  1880.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1881.                 s->channel_order_tab = dca_channel_reorder_lfe[s->amode];

  1882.             } else

  1883.                 s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];

  1884.         }

  1885. 

  1886.         if (channels > !!s->lfe &&

  1887.             s->channel_order_tab[channels - 1 - !!s->lfe] < 0)

  1888.             return AVERROR_INVALIDDATA;

  1889. 

  1890.         if (s->prim_channels + !!s->lfe > 2 &&

  1891.             avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {

  1892.             channels = 2;

  1893.             s->output = s->prim_channels == 2 ? s->amode : DCA_STEREO;

  1894.             avctx->channel_layout = AV_CH_LAYOUT_STEREO;

  1895. 

  1896.             /* Stereo downmix coefficients

  1897.              *

  1898.              * The decoder can only downmix to 2-channel, so we need to ensure

  1899.              * embedded downmix coefficients are actually targeting 2-channel.

  1900.              */

  1901.             if (s->core_downmix && (s->core_downmix_amode == DCA_STEREO ||

  1902.                                     s->core_downmix_amode == DCA_STEREO_TOTAL)) {

  1903.                 int sign, code;

  1904.                 for (i = 0; i < s->prim_channels + !!s->lfe; i++) {

  1905.                     sign = s->core_downmix_codes[i][0] & 0x100 ? 1 : -1;

  1906.                     code = s->core_downmix_codes[i][0] & 0x0FF;

  1907.                     s->downmix_coef[i][0] = (!code ? 0.0f :

  1908.                                              sign * dca_dmixtable[code - 1]);

  1909.                     sign = s->core_downmix_codes[i][1] & 0x100 ? 1 : -1;

  1910.                     code = s->core_downmix_codes[i][1] & 0x0FF;

  1911.                     s->downmix_coef[i][1] = (!code ? 0.0f :

  1912.                                              sign * dca_dmixtable[code - 1]);

  1913.                 }

  1914.                 s->output = s->core_downmix_amode;

  1915.             } else {

  1916.                 int am = s->amode & DCA_CHANNEL_MASK;

  1917.                 if (am >= FF_ARRAY_ELEMS(dca_default_coeffs)) {

  1918.                     av_log(s->avctx, AV_LOG_ERROR,

  1919.                            "Invalid channel mode %d\n", am);

  1920.                     return AVERROR_INVALIDDATA;

  1921.                 }

  1922.                 if (s->prim_channels + !!s->lfe >

  1923.                     FF_ARRAY_ELEMS(dca_default_coeffs[0])) {

  1924.                     avpriv_request_sample(s->avctx, "Downmixing %d channels",

  1925.                                           s->prim_channels + !!s->lfe);

  1926.                     return AVERROR_PATCHWELCOME;

  1927.                 }

  1928.                 for (i = 0; i < s->prim_channels + !!s->lfe; i++) {

  1929.                     s->downmix_coef[i][0] = dca_default_coeffs[am][i][0];

  1930.                     s->downmix_coef[i][1] = dca_default_coeffs[am][i][1];

  1931.                 }

  1932.             }

  1933.             av_dlog(s->avctx, "Stereo downmix coeffs:\n");

  1934.             for (i = 0; i < s->prim_channels + !!s->lfe; i++) {

  1935.                 av_dlog(s->avctx, "L, input channel %d = %f\n", i,

  1936.                         s->downmix_coef[i][0]);

  1937.                 av_dlog(s->avctx, "R, input channel %d = %f\n", i,

  1938.                         s->downmix_coef[i][1]);

  1939.             }

  1940.             av_dlog(s->avctx, "\n");

  1941.         }

  1942.     } else {

  1943.         av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);

  1944.         return AVERROR_INVALIDDATA;

  1945.     }

  1946.     avctx->channels = channels;

  1947. 

  1948.     /* get output buffer */

  1949.     frame->nb_samples = 256 * (s->sample_blocks / 8);

  1950.     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {

  1951.         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");

  1952.         return ret;

  1953.     }

  1954.     samples_flt = (float **)frame->extended_data;

  1955. 

  1956.     /* allocate buffer for extra channels if downmixing */

  1957.     if (avctx->channels < full_channels) {

  1958.         ret = av_samples_get_buffer_size(NULL, full_channels - channels,

  1959.                                          frame->nb_samples,

  1960.                                          avctx->sample_fmt, 0);

  1961.         if (ret < 0)

  1962.             return ret;

  1963. 

  1964.         av_fast_malloc(&s->extra_channels_buffer,

  1965.                        &s->extra_channels_buffer_size, ret);

  1966.         if (!s->extra_channels_buffer)

  1967.             return AVERROR(ENOMEM);

  1968. 

  1969.         ret = av_samples_fill_arrays((uint8_t **)s->extra_channels, NULL,

  1970.                                      s->extra_channels_buffer,

  1971.                                      full_channels - channels,

  1972.                                      frame->nb_samples, avctx->sample_fmt, 0);

  1973.         if (ret < 0)

  1974.             return ret;

  1975.     }

  1976. 

  1977.     /* filter to get final output */

  1978.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1979.         int ch;

  1980. 

  1981.         for (ch = 0; ch < channels; ch++)

  1982.             s->samples_chanptr[ch] = samples_flt[ch] + i * 256;

  1983.         for (; ch < full_channels; ch++)

  1984.             s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * 256;

  1985. 

  1986.         dca_filter_channels(s, i);

  1987. 

  1988.         /* If this was marked as a DTS-ES stream we need to subtract back- */

  1989.         /* channel from SL & SR to remove matrixed back-channel signal */

  1990.         if ((s->source_pcm_res & 1) && s->xch_present) {

  1991.             float *back_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel]];

  1992.             float *lt_chan   = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 2]];

  1993.             float *rt_chan   = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 1]];

  1994.             s->fdsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);

  1995.             s->fdsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);

  1996.         }

  1997.     }

  1998. 

  1999.     /* update lfe history */

  2000.     lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);

  2001.     for (i = 0; i < 2 * s->lfe * 4; i++)

  2002.         s->lfe_data[i] = s->lfe_data[i + lfe_samples];

  2003. 

  2004.     /* AVMatrixEncoding

  2005.      *

  2006.      * DCA_STEREO_TOTAL (Lt/Rt) is equivalent to Dolby Surround */

  2007.     ret = ff_side_data_update_matrix_encoding(frame,

  2008.                                               (s->output & ~DCA_LFE) == DCA_STEREO_TOTAL ?

  2009.                                               AV_MATRIX_ENCODING_DOLBY : AV_MATRIX_ENCODING_NONE);

  2010.     if (ret < 0)

  2011.         return ret;

  2012. 

  2013.     *got_frame_ptr = 1;

  2014. 

  2015.     return buf_size;

  2016. }

  2017. 

  2018. 

  2019. 

  2020. /**

  2021.  * DCA initialization

  2022.  *

  2023.  * @param avctx     pointer to the AVCodecContext

  2024.  */

  2025. 

  2026. static av_cold int dca_decode_init(AVCodecContext *avctx)

  2027. {

  2028.     DCAContext *s = avctx->priv_data;

  2029. 

  2030.     s->avctx = avctx;

  2031.     dca_init_vlcs();

  2032. 

  2033.     avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);

  2034.     ff_mdct_init(&s->imdct, 6, 1, 1.0);

  2035.     ff_synth_filter_init(&s->synth);

  2036.     ff_dcadsp_init(&s->dcadsp);

  2037.     ff_fmt_convert_init(&s->fmt_conv, avctx);

  2038. 

  2039.     avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

  2040. 

  2041.     /* allow downmixing to stereo */

  2042. #if FF_API_REQUEST_CHANNELS

  2043. FF_DISABLE_DEPRECATION_WARNINGS

  2044.     if (avctx->request_channels == 2)

  2045.         avctx->request_channel_layout = AV_CH_LAYOUT_STEREO;

  2046. FF_ENABLE_DEPRECATION_WARNINGS

  2047. #endif

  2048.     if (avctx->channels > 2 &&

  2049.         avctx->request_channel_layout == AV_CH_LAYOUT_STEREO)

  2050.         avctx->channels = 2;

  2051. 

  2052.     return 0;

  2053. }

  2054. 

  2055. static av_cold int dca_decode_end(AVCodecContext *avctx)

  2056. {

  2057.     DCAContext *s = avctx->priv_data;

  2058.     ff_mdct_end(&s->imdct);

  2059.     av_freep(&s->extra_channels_buffer);

  2060.     return 0;

  2061. }

  2062. 

  2063. static const AVProfile profiles[] = {

  2064.     { FF_PROFILE_DTS,        "DTS"        },

  2065.     { FF_PROFILE_DTS_ES,     "DTS-ES"     },

  2066.     { FF_PROFILE_DTS_96_24,  "DTS 96/24"  },

  2067.     { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },

  2068.     { FF_PROFILE_DTS_HD_MA,  "DTS-HD MA"  },

  2069.     { FF_PROFILE_UNKNOWN },

  2070. };

  2071. 

  2072. static const AVOption options[] = {

  2073.     { "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 },

  2074.     { NULL },

  2075. };

  2076. 

  2077. static const AVClass dca_decoder_class = {

  2078.     .class_name = "DCA decoder",

  2079.     .item_name  = av_default_item_name,

  2080.     .option     = options,

  2081.     .version    = LIBAVUTIL_VERSION_INT,

  2082. };

  2083. 

  2084. AVCodec ff_dca_decoder = {

  2085.     .name            = "dca",

  2086.     .long_name       = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),

  2087.     .type            = AVMEDIA_TYPE_AUDIO,

  2088.     .id              = AV_CODEC_ID_DTS,

  2089.     .priv_data_size  = sizeof(DCAContext),

  2090.     .init            = dca_decode_init,

  2091.     .decode          = dca_decode_frame,

  2092.     .close           = dca_decode_end,

  2093.     .capabilities    = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,

  2094.     .sample_fmts     = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,

  2095.                                                        AV_SAMPLE_FMT_NONE },

  2096.     .profiles        = NULL_IF_CONFIG_SMALL(profiles),

  2097.     .priv_class      = &dca_decoder_class,

  2098. };