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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/intmath.h"

  33. #include "libavutil/intreadwrite.h"

  34. #include "libavutil/mathematics.h"

  35. #include "libavutil/samplefmt.h"

  36. #include "avcodec.h"

  37. #include "dsputil.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 "dca_parser.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. /** Bit allocation */

  267. typedef struct {

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

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

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

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

  272. } BitAlloc;

  273. 

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

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

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

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

  278. 

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

  280.                                          int idx)

  281. {

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

  283.            ba->offset;

  284. }

  285. 

  286. typedef struct {

  287.     AVCodecContext *avctx;

  288.     AVFrame frame;

  289.     /* Frame header */

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

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

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

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

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

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

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

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

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

  299. 

  300.     int downmix;                ///< embedded downmix enabled

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

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

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

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

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

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

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

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

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

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

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

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

  313.     int copy_history;           ///< copy history

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

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

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

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

  318. 

  319.     /* Primary audio coding header */

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

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

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

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

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

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

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

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

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

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

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

  331. 

  332.     /* Primary audio coding side information */

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

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

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

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

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

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

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

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

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

  342.     int downmix_coef[DCA_PRIM_CHANNELS_MAX][2];                  ///< stereo downmix coefficients

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

  344. 

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

  346. 

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

  348.     int lfe_scale_factor;

  349. 

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

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

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

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

  354.     int hist_index[DCA_PRIM_CHANNELS_MAX];

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

  356. 

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

  358. 

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

  360.     float *samples_chanptr[DCA_PRIM_CHANNELS_MAX + 1];

  361.     float *extra_channels[DCA_PRIM_CHANNELS_MAX + 1];

  362.     uint8_t *extra_channels_buffer;

  363.     unsigned int extra_channels_buffer_size;

  364. 

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

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

  367. 

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

  369.     GetBitContext gb;

  370.     /* Current position in DCA frame */

  371.     int current_subframe;

  372.     int current_subsubframe;

  373. 

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

  375. 

  376.     /* XCh extension information */

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

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

  379. 

  380.     /* ExSS header parser */

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

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

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

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

  385. 

  386.     int profile;

  387. 

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

  389.     AVFloatDSPContext fdsp;

  390.     FFTContext imdct;

  391.     SynthFilterContext synth;

  392.     DCADSPContext dcadsp;

  393.     FmtConvertContext fmt_conv;

  394. } DCAContext;

  395. 

  396. static const uint16_t dca_vlc_offs[] = {

  397.         0,   512,   640,   768,  1282,  1794,  2436,  3080,  3770,  4454,  5364,

  398.      5372,  5380,  5388,  5392,  5396,  5412,  5420,  5428,  5460,  5492,  5508,

  399.      5572,  5604,  5668,  5796,  5860,  5892,  6412,  6668,  6796,  7308,  7564,

  400.      7820,  8076,  8620,  9132,  9388,  9910, 10166, 10680, 11196, 11726, 12240,

  401.     12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264,

  402.     18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622,

  403. };

  404. 

  405. static av_cold void dca_init_vlcs(void)

  406. {

  407.     static int vlcs_initialized = 0;

  408.     int i, j, c = 14;

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

  410. 

  411.     if (vlcs_initialized)

  412.         return;

  413. 

  414.     dca_bitalloc_index.offset = 1;

  415.     dca_bitalloc_index.wrap = 2;

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

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

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

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

  420.                  bitalloc_12_bits[i], 1, 1,

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

  422.     }

  423.     dca_scalefactor.offset = -64;

  424.     dca_scalefactor.wrap = 2;

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

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

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

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

  429.                  scales_bits[i], 1, 1,

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

  431.     }

  432.     dca_tmode.offset = 0;

  433.     dca_tmode.wrap = 1;

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

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

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

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

  438.                  tmode_bits[i], 1, 1,

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

  440.     }

  441. 

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

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

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

  445.                 break;

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

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

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

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

  450. 

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

  452.                      bitalloc_sizes[i],

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

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

  455.             c++;

  456.         }

  457.     vlcs_initialized = 1;

  458. }

  459. 

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

  461. {

  462.     while (len--)

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

  464. }

  465. 

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

  467. {

  468.     int i, j;

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

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

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

  472. 

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

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

  475. 

  476.     if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)

  477.         s->prim_channels = DCA_PRIM_CHANNELS_MAX;

  478. 

  479. 

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

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

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

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

  484.     }

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

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

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

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

  489.     }

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

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

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

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

  494. 

  495.     /* Get codebooks quantization indexes */

  496.     if (!base_channel)

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

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

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

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

  501. 

  502.     /* Get scale factor adjustment */

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

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

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

  506. 

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

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

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

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

  511. 

  512.     if (s->crc_present) {

  513.         /* Audio header CRC check */

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

  515.     }

  516. 

  517.     s->current_subframe    = 0;

  518.     s->current_subsubframe = 0;

  519. 

  520. #ifdef TRACE

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

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

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

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

  525.                s->subband_activity[i]);

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

  527.                s->vq_start_subband[i]);

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

  529.                s->joint_intensity[i]);

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

  531.                s->transient_huffman[i]);

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

  533.                s->scalefactor_huffman[i]);

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

  535.                s->bitalloc_huffman[i]);

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

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

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

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

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

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

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

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

  544.     }

  545. #endif

  546. 

  547.     return 0;

  548. }

  549. 

  550. static int dca_parse_frame_header(DCAContext *s)

  551. {

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

  553. 

  554.     /* Sync code */

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

  556. 

  557.     /* Frame header */

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

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

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

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

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

  563.     if (s->frame_size < 95)

  564.         return AVERROR_INVALIDDATA;

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

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

  567.     if (!s->sample_rate)

  568.         return AVERROR_INVALIDDATA;

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

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

  571.     if (!s->bit_rate)

  572.         return AVERROR_INVALIDDATA;

  573. 

  574.     s->downmix           = get_bits(&s->gb, 1);

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

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

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

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

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

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

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

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

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

  584. 

  585.     /* TODO: check CRC */

  586.     if (s->crc_present)

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

  588. 

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

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

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

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

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

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

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

  596. 

  597.     /* FIXME: channels mixing levels */

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

  599.     if (s->lfe)

  600.         s->output |= DCA_LFE;

  601. 

  602. #ifdef TRACE

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

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

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

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

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

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

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

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

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

  612.            s->sample_rate);

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

  614.            s->bit_rate);

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

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

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

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

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

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

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

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

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

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

  625.            s->predictor_history);

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

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

  628.            s->multirate_inter);

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

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

  631.     av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

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

  638. #endif

  639. 

  640.     /* Primary audio coding header */

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

  642. 

  643.     return dca_parse_audio_coding_header(s, 0);

  644. }

  645. 

  646. 

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

  648. {

  649.     if (level < 5) {

  650.         /* huffman encoded */

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

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

  653.     } else if (level < 8) {

  654.         if (level + 1 > log2range) {

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

  656.             value = get_bits(gb, log2range);

  657.         } else {

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

  659.         }

  660.     }

  661.     return value;

  662. }

  663. 

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

  665. {

  666.     /* Primary audio coding side information */

  667.     int j, k;

  668. 

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

  670.         return AVERROR_INVALIDDATA;

  671. 

  672.     if (!base_channel) {

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

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

  675.     }

  676. 

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

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

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

  680.     }

  681. 

  682.     /* Get prediction codebook */

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

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

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

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

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

  688.             }

  689.         }

  690.     }

  691. 

  692.     /* Bit allocation index */

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

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

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

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

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

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

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

  700.                 av_log(s->avctx, AV_LOG_ERROR,

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

  702.                 return AVERROR_INVALIDDATA;

  703.             } else {

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

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

  706.             }

  707. 

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

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

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

  711.                 return AVERROR_INVALIDDATA;

  712.             }

  713.         }

  714.     }

  715. 

  716.     /* Transition mode */

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

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

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

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

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

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

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

  724.             }

  725.         }

  726.     }

  727. 

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

  729.         return AVERROR_INVALIDDATA;

  730. 

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

  732.         const uint32_t *scale_table;

  733.         int scale_sum, log_size;

  734. 

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

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

  737. 

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

  739.             scale_table = scale_factor_quant7;

  740.             log_size = 7;

  741.         } else {

  742.             scale_table = scale_factor_quant6;

  743.             log_size = 6;

  744.         }

  745. 

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

  747.         scale_sum = 0;

  748. 

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

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

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

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

  753.             }

  754. 

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

  756.                 /* Get second scale factor */

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

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

  759.             }

  760.         }

  761.     }

  762. 

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

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

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

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

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

  768.     }

  769. 

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

  771.         return AVERROR_INVALIDDATA;

  772. 

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

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

  775.         int source_channel;

  776. 

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

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

  779.             int scale = 0;

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

  781. 

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

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

  784. 

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

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

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

  788.             }

  789. 

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

  791.                 av_log(s->avctx, AV_LOG_DEBUG,

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

  793.                 s->debug_flag |= 0x02;

  794.             }

  795.         }

  796.     }

  797. 

  798.     /* Stereo downmix coefficients */

  799.     if (!base_channel && s->prim_channels > 2) {

  800.         if (s->downmix) {

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

  802.                 s->downmix_coef[j][0] = get_bits(&s->gb, 7);

  803.                 s->downmix_coef[j][1] = get_bits(&s->gb, 7);

  804.             }

  805.         } else {

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

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

  808.                 av_log(s->avctx, AV_LOG_ERROR,

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

  810.                 return AVERROR_INVALIDDATA;

  811.             }

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

  813.                 s->downmix_coef[j][0] = dca_default_coeffs[am][j][0];

  814.                 s->downmix_coef[j][1] = dca_default_coeffs[am][j][1];

  815.             }

  816.         }

  817.     }

  818. 

  819.     /* Dynamic range coefficient */

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

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

  822. 

  823.     /* Side information CRC check word */

  824.     if (s->crc_present) {

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

  826.     }

  827. 

  828.     /*

  829.      * Primary audio data arrays

  830.      */

  831. 

  832.     /* VQ encoded high frequency subbands */

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

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

  835.             /* 1 vector -> 32 samples */

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

  837. 

  838.     /* Low frequency effect data */

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

  840.         /* LFE samples */

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

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

  843.         float lfe_scale;

  844. 

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

  846.             /* Signed 8 bits int */

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

  848.         }

  849. 

  850.         /* Scale factor index */

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

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

  853. 

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

  855.         lfe_scale = 0.035 * s->lfe_scale_factor;

  856. 

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

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

  859.     }

  860. 

  861. #ifdef TRACE

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

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

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

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

  866. 

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

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

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

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

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

  872.     }

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

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

  875.             av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

  881.     }

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

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

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

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

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

  887.     }

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

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

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

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

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

  893.     }

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

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

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

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

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

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

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

  901.         }

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

  903.     }

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

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

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

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

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

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

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

  911.         }

  912.     }

  913.     if (!base_channel && s->prim_channels > 2 && s->downmix) {

  914.         av_log(s->avctx, AV_LOG_DEBUG, "Downmix coeffs:\n");

  915.         for (j = 0; j < s->prim_channels; j++) {

  916.             av_log(s->avctx, AV_LOG_DEBUG, "Channel 0, %d = %f\n", j,

  917.                    dca_downmix_coeffs[s->downmix_coef[j][0]]);

  918.             av_log(s->avctx, AV_LOG_DEBUG, "Channel 1, %d = %f\n", j,

  919.                    dca_downmix_coeffs[s->downmix_coef[j][1]]);

  920.         }

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

  922.     }

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

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

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

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

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

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

  929. 

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

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

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

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

  934.     }

  935. #endif

  936. 

  937.     return 0;

  938. }

  939. 

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

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

  942.                             float scale)

  943. {

  944.     const float *prCoeff;

  945.     int i;

  946. 

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

  948.     int subindex;

  949. 

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

  951. 

  952.     /* Select filter */

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

  954.         prCoeff = fir_32bands_nonperfect;

  955.     else                        /* Perfect reconstruction */

  956.         prCoeff = fir_32bands_perfect;

  957. 

  958.     for (i = sb_act; i < 32; i++)

  959.         s->raXin[i] = 0.0;

  960. 

  961.     /* Reconstructed channel sample index */

  962.     for (subindex = 0; subindex < 8; subindex++) {

  963.         /* Load in one sample from each subband and clear inactive subbands */

  964.         for (i = 0; i < sb_act; i++) {

  965.             unsigned sign = (i - 1) & 2;

  966.             uint32_t v    = AV_RN32A(&samples_in[i][subindex]) ^ sign << 30;

  967.             AV_WN32A(&s->raXin[i], v);

  968.         }

  969. 

  970.         s->synth.synth_filter_float(&s->imdct,

  971.                                     s->subband_fir_hist[chans],

  972.                                     &s->hist_index[chans],

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

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

  975.         samples_out += 32;

  976.     }

  977. }

  978. 

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

  980.                                   int num_deci_sample, float *samples_in,

  981.                                   float *samples_out, float scale)

  982. {

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

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

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

  986.      *   from last subframe as history.

  987.      *

  988.      * samples_out: An array holding interpolated samples

  989.      */

  990. 

  991.     int decifactor;

  992.     const float *prCoeff;

  993.     int deciindex;

  994. 

  995.     /* Select decimation filter */

  996.     if (decimation_select == 1) {

  997.         decifactor = 64;

  998.         prCoeff = lfe_fir_128;

  999.     } else {

  1000.         decifactor = 32;

  1001.         prCoeff = lfe_fir_64;

  1002.     }

  1003.     /* Interpolation */

  1004.     for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {

  1005.         s->dcadsp.lfe_fir(samples_out, samples_in, prCoeff, decifactor, scale);

  1006.         samples_in++;

  1007.         samples_out += 2 * decifactor;

  1008.     }

  1009. }

  1010. 

  1011. /* downmixing routines */

  1012. #define MIX_REAR1(samples, s1, rs, coef)            \

  1013.     samples[0][i] += samples[s1][i] * coef[rs][0];  \

  1014.     samples[1][i] += samples[s1][i] * coef[rs][1];

  1015. 

  1016. #define MIX_REAR2(samples, s1, s2, rs, coef)                                          \

  1017.     samples[0][i] += samples[s1][i] * coef[rs][0] + samples[s2][i] * coef[rs + 1][0]; \

  1018.     samples[1][i] += samples[s1][i] * coef[rs][1] + samples[s2][i] * coef[rs + 1][1];

  1019. 

  1020. #define MIX_FRONT3(samples, coef)                                      \

  1021.     t = samples[c][i];                                                 \

  1022.     u = samples[l][i];                                                 \

  1023.     v = samples[r][i];                                                 \

  1024.     samples[0][i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0];  \

  1025.     samples[1][i] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];

  1026. 

  1027. #define DOWNMIX_TO_STEREO(op1, op2)             \

  1028.     for (i = 0; i < 256; i++) {                 \

  1029.         op1                                     \

  1030.         op2                                     \

  1031.     }

  1032. 

  1033. static void dca_downmix(float **samples, int srcfmt,

  1034.                         int downmix_coef[DCA_PRIM_CHANNELS_MAX][2],

  1035.                         const int8_t *channel_mapping)

  1036. {

  1037.     int c, l, r, sl, sr, s;

  1038.     int i;

  1039.     float t, u, v;

  1040.     float coef[DCA_PRIM_CHANNELS_MAX][2];

  1041. 

  1042.     for (i = 0; i < DCA_PRIM_CHANNELS_MAX; i++) {

  1043.         coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]];

  1044.         coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]];

  1045.     }

  1046. 

  1047.     switch (srcfmt) {

  1048.     case DCA_MONO:

  1049.     case DCA_CHANNEL:

  1050.     case DCA_STEREO_TOTAL:

  1051.     case DCA_STEREO_SUMDIFF:

  1052.     case DCA_4F2R:

  1053.         av_log(NULL, 0, "Not implemented!\n");

  1054.         break;

  1055.     case DCA_STEREO:

  1056.         break;

  1057.     case DCA_3F:

  1058.         c = channel_mapping[0];

  1059.         l = channel_mapping[1];

  1060.         r = channel_mapping[2];

  1061.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );

  1062.         break;

  1063.     case DCA_2F1R:

  1064.         s = channel_mapping[2];

  1065.         DOWNMIX_TO_STEREO(MIX_REAR1(samples, s, 2, coef), );

  1066.         break;

  1067.     case DCA_3F1R:

  1068.         c = channel_mapping[0];

  1069.         l = channel_mapping[1];

  1070.         r = channel_mapping[2];

  1071.         s = channel_mapping[3];

  1072.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1073.                           MIX_REAR1(samples, s, 3, coef));

  1074.         break;

  1075.     case DCA_2F2R:

  1076.         sl = channel_mapping[2];

  1077.         sr = channel_mapping[3];

  1078.         DOWNMIX_TO_STEREO(MIX_REAR2(samples, sl, sr, 2, coef), );

  1079.         break;

  1080.     case DCA_3F2R:

  1081.         c  = channel_mapping[0];

  1082.         l  = channel_mapping[1];

  1083.         r  = channel_mapping[2];

  1084.         sl = channel_mapping[3];

  1085.         sr = channel_mapping[4];

  1086.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1087.                           MIX_REAR2(samples, sl, sr, 3, coef));

  1088.         break;

  1089.     }

  1090. }

  1091. 

  1092. 

  1093. #ifndef decode_blockcodes

  1094. /* Very compact version of the block code decoder that does not use table

  1095.  * look-up but is slightly slower */

  1096. static int decode_blockcode(int code, int levels, int *values)

  1097. {

  1098.     int i;

  1099.     int offset = (levels - 1) >> 1;

  1100. 

  1101.     for (i = 0; i < 4; i++) {

  1102.         int div = FASTDIV(code, levels);

  1103.         values[i] = code - offset - div * levels;

  1104.         code = div;

  1105.     }

  1106. 

  1107.     return code;

  1108. }

  1109. 

  1110. static int decode_blockcodes(int code1, int code2, int levels, int *values)

  1111. {

  1112.     return decode_blockcode(code1, levels, values) |

  1113.            decode_blockcode(code2, levels, values + 4);

  1114. }

  1115. #endif

  1116. 

  1117. static const uint8_t abits_sizes[7]  = { 7, 10, 12, 13, 15, 17, 19 };

  1118. static const uint8_t abits_levels[7] = { 3,  5,  7,  9, 13, 17, 25 };

  1119. 

  1120. #ifndef int8x8_fmul_int32

  1121. static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)

  1122. {

  1123.     float fscale = scale / 16.0;

  1124.     int i;

  1125.     for (i = 0; i < 8; i++)

  1126.         dst[i] = src[i] * fscale;

  1127. }

  1128. #endif

  1129. 

  1130. static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)

  1131. {

  1132.     int k, l;

  1133.     int subsubframe = s->current_subsubframe;

  1134. 

  1135.     const float *quant_step_table;

  1136. 

  1137.     /* FIXME */

  1138.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1139.     LOCAL_ALIGNED_16(int, block, [8]);

  1140. 

  1141.     /*

  1142.      * Audio data

  1143.      */

  1144. 

  1145.     /* Select quantization step size table */

  1146.     if (s->bit_rate_index == 0x1f)

  1147.         quant_step_table = lossless_quant_d;

  1148.     else

  1149.         quant_step_table = lossy_quant_d;

  1150. 

  1151.     for (k = base_channel; k < s->prim_channels; k++) {

  1152.         if (get_bits_left(&s->gb) < 0)

  1153.             return AVERROR_INVALIDDATA;

  1154. 

  1155.         for (l = 0; l < s->vq_start_subband[k]; l++) {

  1156.             int m;

  1157. 

  1158.             /* Select the mid-tread linear quantizer */

  1159.             int abits = s->bitalloc[k][l];

  1160. 

  1161.             float quant_step_size = quant_step_table[abits];

  1162. 

  1163.             /*

  1164.              * Determine quantization index code book and its type

  1165.              */

  1166. 

  1167.             /* Select quantization index code book */

  1168.             int sel = s->quant_index_huffman[k][abits];

  1169. 

  1170.             /*

  1171.              * Extract bits from the bit stream

  1172.              */

  1173.             if (!abits) {

  1174.                 memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));

  1175.             } else {

  1176.                 /* Deal with transients */

  1177.                 int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];

  1178.                 float rscale = quant_step_size * s->scale_factor[k][l][sfi] *

  1179.                                s->scalefactor_adj[k][sel];

  1180. 

  1181.                 if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {

  1182.                     if (abits <= 7) {

  1183.                         /* Block code */

  1184.                         int block_code1, block_code2, size, levels, err;

  1185. 

  1186.                         size   = abits_sizes[abits - 1];

  1187.                         levels = abits_levels[abits - 1];

  1188. 

  1189.                         block_code1 = get_bits(&s->gb, size);

  1190.                         block_code2 = get_bits(&s->gb, size);

  1191.                         err = decode_blockcodes(block_code1, block_code2,

  1192.                                                 levels, block);

  1193.                         if (err) {

  1194.                             av_log(s->avctx, AV_LOG_ERROR,

  1195.                                    "ERROR: block code look-up failed\n");

  1196.                             return AVERROR_INVALIDDATA;

  1197.                         }

  1198.                     } else {

  1199.                         /* no coding */

  1200.                         for (m = 0; m < 8; m++)

  1201.                             block[m] = get_sbits(&s->gb, abits - 3);

  1202.                     }

  1203.                 } else {

  1204.                     /* Huffman coded */

  1205.                     for (m = 0; m < 8; m++)

  1206.                         block[m] = get_bitalloc(&s->gb,

  1207.                                                 &dca_smpl_bitalloc[abits], sel);

  1208.                 }

  1209. 

  1210.                 s->fmt_conv.int32_to_float_fmul_scalar(subband_samples[k][l],

  1211.                                                        block, rscale, 8);

  1212.             }

  1213. 

  1214.             /*

  1215.              * Inverse ADPCM if in prediction mode

  1216.              */

  1217.             if (s->prediction_mode[k][l]) {

  1218.                 int n;

  1219.                 for (m = 0; m < 8; m++) {

  1220.                     for (n = 1; n <= 4; n++)

  1221.                         if (m >= n)

  1222.                             subband_samples[k][l][m] +=

  1223.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1224.                                  subband_samples[k][l][m - n] / 8192);

  1225.                         else if (s->predictor_history)

  1226.                             subband_samples[k][l][m] +=

  1227.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1228.                                  s->subband_samples_hist[k][l][m - n + 4] / 8192);

  1229.                 }

  1230.             }

  1231.         }

  1232. 

  1233.         /*

  1234.          * Decode VQ encoded high frequencies

  1235.          */

  1236.         for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {

  1237.             /* 1 vector -> 32 samples but we only need the 8 samples

  1238.              * for this subsubframe. */

  1239.             int hfvq = s->high_freq_vq[k][l];

  1240. 

  1241.             if (!s->debug_flag & 0x01) {

  1242.                 av_log(s->avctx, AV_LOG_DEBUG,

  1243.                        "Stream with high frequencies VQ coding\n");

  1244.                 s->debug_flag |= 0x01;

  1245.             }

  1246. 

  1247.             int8x8_fmul_int32(subband_samples[k][l],

  1248.                               &high_freq_vq[hfvq][subsubframe * 8],

  1249.                               s->scale_factor[k][l][0]);

  1250.         }

  1251.     }

  1252. 

  1253.     /* Check for DSYNC after subsubframe */

  1254.     if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {

  1255.         if (0xFFFF == get_bits(&s->gb, 16)) {   /* 0xFFFF */

  1256. #ifdef TRACE

  1257.             av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");

  1258. #endif

  1259.         } else {

  1260.             av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");

  1261.         }

  1262.     }

  1263. 

  1264.     /* Backup predictor history for adpcm */

  1265.     for (k = base_channel; k < s->prim_channels; k++)

  1266.         for (l = 0; l < s->vq_start_subband[k]; l++)

  1267.             memcpy(s->subband_samples_hist[k][l],

  1268.                    &subband_samples[k][l][4],

  1269.                    4 * sizeof(subband_samples[0][0][0]));

  1270. 

  1271.     return 0;

  1272. }

  1273. 

  1274. static int dca_filter_channels(DCAContext *s, int block_index)

  1275. {

  1276.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1277.     int k;

  1278. 

  1279.     /* 32 subbands QMF */

  1280.     for (k = 0; k < s->prim_channels; k++) {

  1281. /*        static float pcm_to_double[8] = { 32768.0, 32768.0, 524288.0, 524288.0,

  1282.                                             0, 8388608.0, 8388608.0 };*/

  1283.         if (s->channel_order_tab[k] >= 0)

  1284.             qmf_32_subbands(s, k, subband_samples[k],

  1285.                             s->samples_chanptr[s->channel_order_tab[k]],

  1286.                             M_SQRT1_2 / 32768.0 /* pcm_to_double[s->source_pcm_res] */);

  1287.     }

  1288. 

  1289.     /* Down mixing */

  1290.     if (s->avctx->request_channels == 2 && s->prim_channels > 2) {

  1291.         dca_downmix(s->samples_chanptr, s->amode, s->downmix_coef, s->channel_order_tab);

  1292.     }

  1293. 

  1294.     /* Generate LFE samples for this subsubframe FIXME!!! */

  1295.     if (s->output & DCA_LFE) {

  1296.         lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,

  1297.                               s->lfe_data + 2 * s->lfe * (block_index + 4),

  1298.                               s->samples_chanptr[dca_lfe_index[s->amode]],

  1299.                               1.0 / (256.0 * 32768.0));

  1300.         /* Outputs 20bits pcm samples */

  1301.     }

  1302. 

  1303.     return 0;

  1304. }

  1305. 

  1306. 

  1307. static int dca_subframe_footer(DCAContext *s, int base_channel)

  1308. {

  1309.     int aux_data_count = 0, i;

  1310. 

  1311.     /*

  1312.      * Unpack optional information

  1313.      */

  1314. 

  1315.     /* presumably optional information only appears in the core? */

  1316.     if (!base_channel) {

  1317.         if (s->timestamp)

  1318.             skip_bits_long(&s->gb, 32);

  1319. 

  1320.         if (s->aux_data)

  1321.             aux_data_count = get_bits(&s->gb, 6);

  1322. 

  1323.         for (i = 0; i < aux_data_count; i++)

  1324.             get_bits(&s->gb, 8);

  1325. 

  1326.         if (s->crc_present && (s->downmix || s->dynrange))

  1327.             get_bits(&s->gb, 16);

  1328.     }

  1329. 

  1330.     return 0;

  1331. }

  1332. 

  1333. /**

  1334.  * Decode a dca frame block

  1335.  *

  1336.  * @param s     pointer to the DCAContext

  1337.  */

  1338. 

  1339. static int dca_decode_block(DCAContext *s, int base_channel, int block_index)

  1340. {

  1341.     int ret;

  1342. 

  1343.     /* Sanity check */

  1344.     if (s->current_subframe >= s->subframes) {

  1345.         av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",

  1346.                s->current_subframe, s->subframes);

  1347.         return AVERROR_INVALIDDATA;

  1348.     }

  1349. 

  1350.     if (!s->current_subsubframe) {

  1351. #ifdef TRACE

  1352.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");

  1353. #endif

  1354.         /* Read subframe header */

  1355.         if ((ret = dca_subframe_header(s, base_channel, block_index)))

  1356.             return ret;

  1357.     }

  1358. 

  1359.     /* Read subsubframe */

  1360. #ifdef TRACE

  1361.     av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");

  1362. #endif

  1363.     if ((ret = dca_subsubframe(s, base_channel, block_index)))

  1364.         return ret;

  1365. 

  1366.     /* Update state */

  1367.     s->current_subsubframe++;

  1368.     if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {

  1369.         s->current_subsubframe = 0;

  1370.         s->current_subframe++;

  1371.     }

  1372.     if (s->current_subframe >= s->subframes) {

  1373. #ifdef TRACE

  1374.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");

  1375. #endif

  1376.         /* Read subframe footer */

  1377.         if ((ret = dca_subframe_footer(s, base_channel)))

  1378.             return ret;

  1379.     }

  1380. 

  1381.     return 0;

  1382. }

  1383. 

  1384. /**

  1385.  * Return the number of channels in an ExSS speaker mask (HD)

  1386.  */

  1387. static int dca_exss_mask2count(int mask)

  1388. {

  1389.     /* count bits that mean speaker pairs twice */

  1390.     return av_popcount(mask) +

  1391.            av_popcount(mask & (DCA_EXSS_CENTER_LEFT_RIGHT      |

  1392.                                DCA_EXSS_FRONT_LEFT_RIGHT       |

  1393.                                DCA_EXSS_FRONT_HIGH_LEFT_RIGHT  |

  1394.                                DCA_EXSS_WIDE_LEFT_RIGHT        |

  1395.                                DCA_EXSS_SIDE_LEFT_RIGHT        |

  1396.                                DCA_EXSS_SIDE_HIGH_LEFT_RIGHT   |

  1397.                                DCA_EXSS_SIDE_REAR_LEFT_RIGHT   |

  1398.                                DCA_EXSS_REAR_LEFT_RIGHT        |

  1399.                                DCA_EXSS_REAR_HIGH_LEFT_RIGHT));

  1400. }

  1401. 

  1402. /**

  1403.  * Skip mixing coefficients of a single mix out configuration (HD)

  1404.  */

  1405. static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch)

  1406. {

  1407.     int i;

  1408. 

  1409.     for (i = 0; i < channels; i++) {

  1410.         int mix_map_mask = get_bits(gb, out_ch);

  1411.         int num_coeffs = av_popcount(mix_map_mask);

  1412.         skip_bits_long(gb, num_coeffs * 6);

  1413.     }

  1414. }

  1415. 

  1416. /**

  1417.  * Parse extension substream asset header (HD)

  1418.  */

  1419. static int dca_exss_parse_asset_header(DCAContext *s)

  1420. {

  1421.     int header_pos = get_bits_count(&s->gb);

  1422.     int header_size;

  1423.     int channels;

  1424.     int embedded_stereo = 0;

  1425.     int embedded_6ch    = 0;

  1426.     int drc_code_present;

  1427.     int extensions_mask;

  1428.     int i, j;

  1429. 

  1430.     if (get_bits_left(&s->gb) < 16)

  1431.         return -1;

  1432. 

  1433.     /* We will parse just enough to get to the extensions bitmask with which

  1434.      * we can set the profile value. */

  1435. 

  1436.     header_size = get_bits(&s->gb, 9) + 1;

  1437.     skip_bits(&s->gb, 3); // asset index

  1438. 

  1439.     if (s->static_fields) {

  1440.         if (get_bits1(&s->gb))

  1441.             skip_bits(&s->gb, 4); // asset type descriptor

  1442.         if (get_bits1(&s->gb))

  1443.             skip_bits_long(&s->gb, 24); // language descriptor

  1444. 

  1445.         if (get_bits1(&s->gb)) {

  1446.             /* How can one fit 1024 bytes of text here if the maximum value

  1447.              * for the asset header size field above was 512 bytes? */

  1448.             int text_length = get_bits(&s->gb, 10) + 1;

  1449.             if (get_bits_left(&s->gb) < text_length * 8)

  1450.                 return -1;

  1451.             skip_bits_long(&s->gb, text_length * 8); // info text

  1452.         }

  1453. 

  1454.         skip_bits(&s->gb, 5); // bit resolution - 1

  1455.         skip_bits(&s->gb, 4); // max sample rate code

  1456.         channels = get_bits(&s->gb, 8) + 1;

  1457. 

  1458.         if (get_bits1(&s->gb)) { // 1-to-1 channels to speakers

  1459.             int spkr_remap_sets;

  1460.             int spkr_mask_size = 16;

  1461.             int num_spkrs[7];

  1462. 

  1463.             if (channels > 2)

  1464.                 embedded_stereo = get_bits1(&s->gb);

  1465.             if (channels > 6)

  1466.                 embedded_6ch = get_bits1(&s->gb);

  1467. 

  1468.             if (get_bits1(&s->gb)) {

  1469.                 spkr_mask_size = (get_bits(&s->gb, 2) + 1) << 2;

  1470.                 skip_bits(&s->gb, spkr_mask_size); // spkr activity mask

  1471.             }

  1472. 

  1473.             spkr_remap_sets = get_bits(&s->gb, 3);

  1474. 

  1475.             for (i = 0; i < spkr_remap_sets; i++) {

  1476.                 /* std layout mask for each remap set */

  1477.                 num_spkrs[i] = dca_exss_mask2count(get_bits(&s->gb, spkr_mask_size));

  1478.             }

  1479. 

  1480.             for (i = 0; i < spkr_remap_sets; i++) {

  1481.                 int num_dec_ch_remaps = get_bits(&s->gb, 5) + 1;

  1482.                 if (get_bits_left(&s->gb) < 0)

  1483.                     return -1;

  1484. 

  1485.                 for (j = 0; j < num_spkrs[i]; j++) {

  1486.                     int remap_dec_ch_mask = get_bits_long(&s->gb, num_dec_ch_remaps);

  1487.                     int num_dec_ch = av_popcount(remap_dec_ch_mask);

  1488.                     skip_bits_long(&s->gb, num_dec_ch * 5); // remap codes

  1489.                 }

  1490.             }

  1491. 

  1492.         } else {

  1493.             skip_bits(&s->gb, 3); // representation type

  1494.         }

  1495.     }

  1496. 

  1497.     drc_code_present = get_bits1(&s->gb);

  1498.     if (drc_code_present)

  1499.         get_bits(&s->gb, 8); // drc code

  1500. 

  1501.     if (get_bits1(&s->gb))

  1502.         skip_bits(&s->gb, 5); // dialog normalization code

  1503. 

  1504.     if (drc_code_present && embedded_stereo)

  1505.         get_bits(&s->gb, 8); // drc stereo code

  1506. 

  1507.     if (s->mix_metadata && get_bits1(&s->gb)) {

  1508.         skip_bits(&s->gb, 1); // external mix

  1509.         skip_bits(&s->gb, 6); // post mix gain code

  1510. 

  1511.         if (get_bits(&s->gb, 2) != 3) // mixer drc code

  1512.             skip_bits(&s->gb, 3); // drc limit

  1513.         else

  1514.             skip_bits(&s->gb, 8); // custom drc code

  1515. 

  1516.         if (get_bits1(&s->gb)) // channel specific scaling

  1517.             for (i = 0; i < s->num_mix_configs; i++)

  1518.                 skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6); // scale codes

  1519.         else

  1520.             skip_bits_long(&s->gb, s->num_mix_configs * 6); // scale codes

  1521. 

  1522.         for (i = 0; i < s->num_mix_configs; i++) {

  1523.             if (get_bits_left(&s->gb) < 0)

  1524.                 return -1;

  1525.             dca_exss_skip_mix_coeffs(&s->gb, channels, s->mix_config_num_ch[i]);

  1526.             if (embedded_6ch)

  1527.                 dca_exss_skip_mix_coeffs(&s->gb, 6, s->mix_config_num_ch[i]);

  1528.             if (embedded_stereo)

  1529.                 dca_exss_skip_mix_coeffs(&s->gb, 2, s->mix_config_num_ch[i]);

  1530.         }

  1531.     }

  1532. 

  1533.     switch (get_bits(&s->gb, 2)) {

  1534.     case 0: extensions_mask = get_bits(&s->gb, 12); break;

  1535.     case 1: extensions_mask = DCA_EXT_EXSS_XLL;     break;

  1536.     case 2: extensions_mask = DCA_EXT_EXSS_LBR;     break;

  1537.     case 3: extensions_mask = 0; /* aux coding */   break;

  1538.     }

  1539. 

  1540.     /* not parsed further, we were only interested in the extensions mask */

  1541. 

  1542.     if (get_bits_left(&s->gb) < 0)

  1543.         return -1;

  1544. 

  1545.     if (get_bits_count(&s->gb) - header_pos > header_size * 8) {

  1546.         av_log(s->avctx, AV_LOG_WARNING, "Asset header size mismatch.\n");

  1547.         return -1;

  1548.     }

  1549.     skip_bits_long(&s->gb, header_pos + header_size * 8 - get_bits_count(&s->gb));

  1550. 

  1551.     if (extensions_mask & DCA_EXT_EXSS_XLL)

  1552.         s->profile = FF_PROFILE_DTS_HD_MA;

  1553.     else if (extensions_mask & (DCA_EXT_EXSS_XBR | DCA_EXT_EXSS_X96 |

  1554.                                 DCA_EXT_EXSS_XXCH))

  1555.         s->profile = FF_PROFILE_DTS_HD_HRA;

  1556. 

  1557.     if (!(extensions_mask & DCA_EXT_CORE))

  1558.         av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n");

  1559.     if ((extensions_mask & DCA_CORE_EXTS) != s->core_ext_mask)

  1560.         av_log(s->avctx, AV_LOG_WARNING,

  1561.                "DTS extensions detection mismatch (%d, %d)\n",

  1562.                extensions_mask & DCA_CORE_EXTS, s->core_ext_mask);

  1563. 

  1564.     return 0;

  1565. }

  1566. 

  1567. /**

  1568.  * Parse extension substream header (HD)

  1569.  */

  1570. static void dca_exss_parse_header(DCAContext *s)

  1571. {

  1572.     int ss_index;

  1573.     int blownup;

  1574.     int num_audiop = 1;

  1575.     int num_assets = 1;

  1576.     int active_ss_mask[8];

  1577.     int i, j;

  1578. 

  1579.     if (get_bits_left(&s->gb) < 52)

  1580.         return;

  1581. 

  1582.     skip_bits(&s->gb, 8); // user data

  1583.     ss_index = get_bits(&s->gb, 2);

  1584. 

  1585.     blownup = get_bits1(&s->gb);

  1586.     skip_bits(&s->gb,  8 + 4 * blownup); // header_size

  1587.     skip_bits(&s->gb, 16 + 4 * blownup); // hd_size

  1588. 

  1589.     s->static_fields = get_bits1(&s->gb);

  1590.     if (s->static_fields) {

  1591.         skip_bits(&s->gb, 2); // reference clock code

  1592.         skip_bits(&s->gb, 3); // frame duration code

  1593. 

  1594.         if (get_bits1(&s->gb))

  1595.             skip_bits_long(&s->gb, 36); // timestamp

  1596. 

  1597.         /* a single stream can contain multiple audio assets that can be

  1598.          * combined to form multiple audio presentations */

  1599. 

  1600.         num_audiop = get_bits(&s->gb, 3) + 1;

  1601.         if (num_audiop > 1) {

  1602.             av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio presentations.");

  1603.             /* ignore such streams for now */

  1604.             return;

  1605.         }

  1606. 

  1607.         num_assets = get_bits(&s->gb, 3) + 1;

  1608.         if (num_assets > 1) {

  1609.             av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio assets.");

  1610.             /* ignore such streams for now */

  1611.             return;

  1612.         }

  1613. 

  1614.         for (i = 0; i < num_audiop; i++)

  1615.             active_ss_mask[i] = get_bits(&s->gb, ss_index + 1);

  1616. 

  1617.         for (i = 0; i < num_audiop; i++)

  1618.             for (j = 0; j <= ss_index; j++)

  1619.                 if (active_ss_mask[i] & (1 << j))

  1620.                     skip_bits(&s->gb, 8); // active asset mask

  1621. 

  1622.         s->mix_metadata = get_bits1(&s->gb);

  1623.         if (s->mix_metadata) {

  1624.             int mix_out_mask_size;

  1625. 

  1626.             skip_bits(&s->gb, 2); // adjustment level

  1627.             mix_out_mask_size  = (get_bits(&s->gb, 2) + 1) << 2;

  1628.             s->num_mix_configs =  get_bits(&s->gb, 2) + 1;

  1629. 

  1630.             for (i = 0; i < s->num_mix_configs; i++) {

  1631.                 int mix_out_mask        = get_bits(&s->gb, mix_out_mask_size);

  1632.                 s->mix_config_num_ch[i] = dca_exss_mask2count(mix_out_mask);

  1633.             }

  1634.         }

  1635.     }

  1636. 

  1637.     for (i = 0; i < num_assets; i++)

  1638.         skip_bits_long(&s->gb, 16 + 4 * blownup);  // asset size

  1639. 

  1640.     for (i = 0; i < num_assets; i++) {

  1641.         if (dca_exss_parse_asset_header(s))

  1642.             return;

  1643.     }

  1644. 

  1645.     /* not parsed further, we were only interested in the extensions mask

  1646.      * from the asset header */

  1647. }

  1648. 

  1649. /**

  1650.  * Main frame decoding function

  1651.  * FIXME add arguments

  1652.  */

  1653. static int dca_decode_frame(AVCodecContext *avctx, void *data,

  1654.                             int *got_frame_ptr, AVPacket *avpkt)

  1655. {

  1656.     const uint8_t *buf = avpkt->data;

  1657.     int buf_size = avpkt->size;

  1658. 

  1659.     int lfe_samples;

  1660.     int num_core_channels = 0;

  1661.     int i, ret;

  1662.     float  **samples_flt;

  1663.     DCAContext *s = avctx->priv_data;

  1664.     int channels, full_channels;

  1665.     int core_ss_end;

  1666. 

  1667. 

  1668.     s->xch_present = 0;

  1669. 

  1670.     s->dca_buffer_size = ff_dca_convert_bitstream(buf, buf_size, s->dca_buffer,

  1671.                                                   DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);

  1672.     if (s->dca_buffer_size == AVERROR_INVALIDDATA) {

  1673.         av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");

  1674.         return AVERROR_INVALIDDATA;

  1675.     }

  1676. 

  1677.     init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);

  1678.     if ((ret = dca_parse_frame_header(s)) < 0) {

  1679.         //seems like the frame is corrupt, try with the next one

  1680.         return ret;

  1681.     }

  1682.     //set AVCodec values with parsed data

  1683.     avctx->sample_rate = s->sample_rate;

  1684.     avctx->bit_rate    = s->bit_rate;

  1685. 

  1686.     s->profile = FF_PROFILE_DTS;

  1687. 

  1688.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1689.         if ((ret = dca_decode_block(s, 0, i))) {

  1690.             av_log(avctx, AV_LOG_ERROR, "error decoding block\n");

  1691.             return ret;

  1692.         }

  1693.     }

  1694. 

  1695.     /* record number of core channels incase less than max channels are requested */

  1696.     num_core_channels = s->prim_channels;

  1697. 

  1698.     if (s->ext_coding)

  1699.         s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];

  1700.     else

  1701.         s->core_ext_mask = 0;

  1702. 

  1703.     core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;

  1704. 

  1705.     /* only scan for extensions if ext_descr was unknown or indicated a

  1706.      * supported XCh extension */

  1707.     if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {

  1708. 

  1709.         /* if ext_descr was unknown, clear s->core_ext_mask so that the

  1710.          * extensions scan can fill it up */

  1711.         s->core_ext_mask = FFMAX(s->core_ext_mask, 0);

  1712. 

  1713.         /* extensions start at 32-bit boundaries into bitstream */

  1714.         skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1715. 

  1716.         while (core_ss_end - get_bits_count(&s->gb) >= 32) {

  1717.             uint32_t bits = get_bits_long(&s->gb, 32);

  1718. 

  1719.             switch (bits) {

  1720.             case 0x5a5a5a5a: {

  1721.                 int ext_amode, xch_fsize;

  1722. 

  1723.                 s->xch_base_channel = s->prim_channels;

  1724. 

  1725.                 /* validate sync word using XCHFSIZE field */

  1726.                 xch_fsize = show_bits(&s->gb, 10);

  1727.                 if ((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&

  1728.                     (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))

  1729.                     continue;

  1730. 

  1731.                 /* skip length-to-end-of-frame field for the moment */

  1732.                 skip_bits(&s->gb, 10);

  1733. 

  1734.                 s->core_ext_mask |= DCA_EXT_XCH;

  1735. 

  1736.                 /* extension amode(number of channels in extension) should be 1 */

  1737.                 /* AFAIK XCh is not used for more channels */

  1738.                 if ((ext_amode = get_bits(&s->gb, 4)) != 1) {

  1739.                     av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"

  1740.                            " supported!\n", ext_amode);

  1741.                     continue;

  1742.                 }

  1743. 

  1744.                 /* much like core primary audio coding header */

  1745.                 dca_parse_audio_coding_header(s, s->xch_base_channel);

  1746. 

  1747.                 for (i = 0; i < (s->sample_blocks / 8); i++)

  1748.                     if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {

  1749.                         av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");

  1750.                         continue;

  1751.                     }

  1752. 

  1753.                 s->xch_present = 1;

  1754.                 break;

  1755.             }

  1756.             case 0x47004a03:

  1757.                 /* XXCh: extended channels */

  1758.                 /* usually found either in core or HD part in DTS-HD HRA streams,

  1759.                  * but not in DTS-ES which contains XCh extensions instead */

  1760.                 s->core_ext_mask |= DCA_EXT_XXCH;

  1761.                 break;

  1762. 

  1763.             case 0x1d95f262: {

  1764.                 int fsize96 = show_bits(&s->gb, 12) + 1;

  1765.                 if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)

  1766.                     continue;

  1767. 

  1768.                 av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n",

  1769.                        get_bits_count(&s->gb));

  1770.                 skip_bits(&s->gb, 12);

  1771.                 av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);

  1772.                 av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));

  1773. 

  1774.                 s->core_ext_mask |= DCA_EXT_X96;

  1775.                 break;

  1776.             }

  1777.             }

  1778. 

  1779.             skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1780.         }

  1781.     } else {

  1782.         /* no supported extensions, skip the rest of the core substream */

  1783.         skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));

  1784.     }

  1785. 

  1786.     if (s->core_ext_mask & DCA_EXT_X96)

  1787.         s->profile = FF_PROFILE_DTS_96_24;

  1788.     else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))

  1789.         s->profile = FF_PROFILE_DTS_ES;

  1790. 

  1791.     /* check for ExSS (HD part) */

  1792.     if (s->dca_buffer_size - s->frame_size > 32 &&

  1793.         get_bits_long(&s->gb, 32) == DCA_HD_MARKER)

  1794.         dca_exss_parse_header(s);

  1795. 

  1796.     avctx->profile = s->profile;

  1797. 

  1798.     full_channels = channels = s->prim_channels + !!s->lfe;

  1799. 

  1800.     if (s->amode < 16) {

  1801.         avctx->channel_layout = dca_core_channel_layout[s->amode];

  1802. 

  1803.         if (s->xch_present && (!avctx->request_channels ||

  1804.                                avctx->request_channels > num_core_channels + !!s->lfe)) {

  1805.             avctx->channel_layout |= AV_CH_BACK_CENTER;

  1806.             if (s->lfe) {

  1807.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1808.                 s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode];

  1809.             } else {

  1810.                 s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];

  1811.             }

  1812.         } else {

  1813.             channels = num_core_channels + !!s->lfe;

  1814.             s->xch_present = 0; /* disable further xch processing */

  1815.             if (s->lfe) {

  1816.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1817.                 s->channel_order_tab = dca_channel_reorder_lfe[s->amode];

  1818.             } else

  1819.                 s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];

  1820.         }

  1821. 

  1822.         if (channels > !!s->lfe &&

  1823.             s->channel_order_tab[channels - 1 - !!s->lfe] < 0)

  1824.             return AVERROR_INVALIDDATA;

  1825. 

  1826.         if (avctx->request_channels == 2 && s->prim_channels > 2) {

  1827.             channels = 2;

  1828.             s->output = DCA_STEREO;

  1829.             avctx->channel_layout = AV_CH_LAYOUT_STEREO;

  1830.         }

  1831.     } else {

  1832.         av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);

  1833.         return AVERROR_INVALIDDATA;

  1834.     }

  1835.     avctx->channels = channels;

  1836. 

  1837.     /* get output buffer */

  1838.     s->frame.nb_samples = 256 * (s->sample_blocks / 8);

  1839.     if ((ret = ff_get_buffer(avctx, &s->frame)) < 0) {

  1840.         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");

  1841.         return ret;

  1842.     }

  1843.     samples_flt = (float  **) s->frame.extended_data;

  1844. 

  1845.     /* allocate buffer for extra channels if downmixing */

  1846.     if (avctx->channels < full_channels) {

  1847.         ret = av_samples_get_buffer_size(NULL, full_channels - channels,

  1848.                                          s->frame.nb_samples,

  1849.                                          avctx->sample_fmt, 0);

  1850.         if (ret < 0)

  1851.             return ret;

  1852. 

  1853.         av_fast_malloc(&s->extra_channels_buffer,

  1854.                        &s->extra_channels_buffer_size, ret);

  1855.         if (!s->extra_channels_buffer)

  1856.             return AVERROR(ENOMEM);

  1857. 

  1858.         ret = av_samples_fill_arrays((uint8_t **)s->extra_channels, NULL,

  1859.                                      s->extra_channels_buffer,

  1860.                                      full_channels - channels,

  1861.                                      s->frame.nb_samples, avctx->sample_fmt, 0);

  1862.         if (ret < 0)

  1863.             return ret;

  1864.     }

  1865. 

  1866.     /* filter to get final output */

  1867.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1868.         int ch;

  1869. 

  1870.         for (ch = 0; ch < channels; ch++)

  1871.             s->samples_chanptr[ch] = samples_flt[ch] + i * 256;

  1872.         for (; ch < full_channels; ch++)

  1873.             s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * 256;

  1874. 

  1875.         dca_filter_channels(s, i);

  1876. 

  1877.         /* If this was marked as a DTS-ES stream we need to subtract back- */

  1878.         /* channel from SL & SR to remove matrixed back-channel signal */

  1879.         if ((s->source_pcm_res & 1) && s->xch_present) {

  1880.             float *back_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel]];

  1881.             float *lt_chan   = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 2]];

  1882.             float *rt_chan   = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 1]];

  1883.             s->fdsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);

  1884.             s->fdsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);

  1885.         }

  1886.     }

  1887. 

  1888.     /* update lfe history */

  1889.     lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);

  1890.     for (i = 0; i < 2 * s->lfe * 4; i++)

  1891.         s->lfe_data[i] = s->lfe_data[i + lfe_samples];

  1892. 

  1893.     *got_frame_ptr    = 1;

  1894.     *(AVFrame *) data = s->frame;

  1895. 

  1896.     return buf_size;

  1897. }

  1898. 

  1899. 

  1900. 

  1901. /**

  1902.  * DCA initialization

  1903.  *

  1904.  * @param avctx     pointer to the AVCodecContext

  1905.  */

  1906. 

  1907. static av_cold int dca_decode_init(AVCodecContext *avctx)

  1908. {

  1909.     DCAContext *s = avctx->priv_data;

  1910. 

  1911.     s->avctx = avctx;

  1912.     dca_init_vlcs();

  1913. 

  1914.     avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);

  1915.     ff_mdct_init(&s->imdct, 6, 1, 1.0);

  1916.     ff_synth_filter_init(&s->synth);

  1917.     ff_dcadsp_init(&s->dcadsp);

  1918.     ff_fmt_convert_init(&s->fmt_conv, avctx);

  1919. 

  1920.     avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

  1921. 

  1922.     /* allow downmixing to stereo */

  1923.     if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&

  1924.         avctx->request_channels == 2) {

  1925.         avctx->channels = avctx->request_channels;

  1926.     }

  1927. 

  1928.     avcodec_get_frame_defaults(&s->frame);

  1929.     avctx->coded_frame = &s->frame;

  1930. 

  1931.     return 0;

  1932. }

  1933. 

  1934. static av_cold int dca_decode_end(AVCodecContext *avctx)

  1935. {

  1936.     DCAContext *s = avctx->priv_data;

  1937.     ff_mdct_end(&s->imdct);

  1938.     av_freep(&s->extra_channels_buffer);

  1939.     return 0;

  1940. }

  1941. 

  1942. static const AVProfile profiles[] = {

  1943.     { FF_PROFILE_DTS,        "DTS"        },

  1944.     { FF_PROFILE_DTS_ES,     "DTS-ES"     },

  1945.     { FF_PROFILE_DTS_96_24,  "DTS 96/24"  },

  1946.     { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },

  1947.     { FF_PROFILE_DTS_HD_MA,  "DTS-HD MA"  },

  1948.     { FF_PROFILE_UNKNOWN },

  1949. };

  1950. 

  1951. AVCodec ff_dca_decoder = {

  1952.     .name            = "dca",

  1953.     .type            = AVMEDIA_TYPE_AUDIO,

  1954.     .id              = AV_CODEC_ID_DTS,

  1955.     .priv_data_size  = sizeof(DCAContext),

  1956.     .init            = dca_decode_init,

  1957.     .decode          = dca_decode_frame,

  1958.     .close           = dca_decode_end,

  1959.     .long_name       = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),

  1960.     .capabilities    = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,

  1961.     .sample_fmts     = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,

  1962.                                                        AV_SAMPLE_FMT_NONE },

  1963.     .profiles        = NULL_IF_CONFIG_SMALL(profiles),

  1964. };