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FFmpeg/libavcodec/dca.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 FFmpeg.

  9.  *

  10.  * FFmpeg is free software; you can redistribute it and/or

  11.  * modify it under the terms of the GNU Lesser General Public

  12.  * License as published by the Free Software Foundation; either

  13.  * version 2.1 of the License, or (at your option) any later version.

  14.  *

  15.  * FFmpeg is distributed in the hope that it will be useful,

  16.  * but WITHOUT ANY WARRANTY; without even the implied warranty of

  17.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU

  18.  * Lesser General Public License for more details.

  19.  *

  20.  * You should have received a copy of the GNU Lesser General Public

  21.  * License along with FFmpeg; if not, write to the Free Software

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

  23.  */

  24. 

  25. #include <math.h>

  26. #include <stddef.h>

  27. #include <stdio.h>

  28. 

  29. #include "libavutil/common.h"

  30. #include "libavutil/intmath.h"

  31. #include "libavutil/intreadwrite.h"

  32. #include "libavutil/mathematics.h"

  33. #include "libavutil/audioconvert.h"

  34. #include "avcodec.h"

  35. #include "dsputil.h"

  36. #include "fft.h"

  37. #include "get_bits.h"

  38. #include "put_bits.h"

  39. #include "dcadata.h"

  40. #include "dcahuff.h"

  41. #include "dca.h"

  42. #include "dca_parser.h"

  43. #include "synth_filter.h"

  44. #include "dcadsp.h"

  45. #include "fmtconvert.h"

  46. 

  47. #if ARCH_ARM

  48. #   include "arm/dca.h"

  49. #endif

  50. 

  51. //#define TRACE

  52. 

  53. #define DCA_PRIM_CHANNELS_MAX  (7)

  54. #define DCA_SUBBANDS          (64)

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

  56. #define DCA_SUBSUBFRAMES_MAX   (4)

  57. #define DCA_SUBFRAMES_MAX     (16)

  58. #define DCA_BLOCKS_MAX        (16)

  59. #define DCA_LFE_MAX            (3)

  60. #define DCA_CHSETS_MAX         (4)

  61. #define DCA_CHSET_CHANS_MAX    (8)

  62. 

  63. enum DCAMode {

  64.     DCA_MONO = 0,

  65.     DCA_CHANNEL,

  66.     DCA_STEREO,

  67.     DCA_STEREO_SUMDIFF,

  68.     DCA_STEREO_TOTAL,

  69.     DCA_3F,

  70.     DCA_2F1R,

  71.     DCA_3F1R,

  72.     DCA_2F2R,

  73.     DCA_3F2R,

  74.     DCA_4F2R

  75. };

  76. 

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

  78. enum DCAExSSSpeakerMask {

  79.     DCA_EXSS_FRONT_CENTER          = 0x0001,

  80.     DCA_EXSS_FRONT_LEFT_RIGHT      = 0x0002,

  81.     DCA_EXSS_SIDE_REAR_LEFT_RIGHT  = 0x0004,

  82.     DCA_EXSS_LFE                   = 0x0008,

  83.     DCA_EXSS_REAR_CENTER           = 0x0010,

  84.     DCA_EXSS_FRONT_HIGH_LEFT_RIGHT = 0x0020,

  85.     DCA_EXSS_REAR_LEFT_RIGHT       = 0x0040,

  86.     DCA_EXSS_FRONT_HIGH_CENTER     = 0x0080,

  87.     DCA_EXSS_OVERHEAD              = 0x0100,

  88.     DCA_EXSS_CENTER_LEFT_RIGHT     = 0x0200,

  89.     DCA_EXSS_WIDE_LEFT_RIGHT       = 0x0400,

  90.     DCA_EXSS_SIDE_LEFT_RIGHT       = 0x0800,

  91.     DCA_EXSS_LFE2                  = 0x1000,

  92.     DCA_EXSS_SIDE_HIGH_LEFT_RIGHT  = 0x2000,

  93.     DCA_EXSS_REAR_HIGH_CENTER      = 0x4000,

  94.     DCA_EXSS_REAR_HIGH_LEFT_RIGHT  = 0x8000,

  95. };

  96. 

  97. enum DCAExtensionMask {

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

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

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

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

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

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

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

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

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

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

  108. };

  109. 

  110. /* -1 are reserved or unknown */

  111. static const int dca_ext_audio_descr_mask[] = {

  112.     DCA_EXT_XCH,

  113.     -1,

  114.     DCA_EXT_X96,

  115.     DCA_EXT_XCH | DCA_EXT_X96,

  116.     -1,

  117.     -1,

  118.     DCA_EXT_XXCH,

  119.     -1,

  120. };

  121. 

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

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

  124. 

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

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

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

  128.  *

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

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

  131.  * OV -> center back

  132.  * All 2 channel configurations -> AV_CH_LAYOUT_STEREO

  133.  */

  134. static const uint64_t dca_core_channel_layout[] = {

  135.     AV_CH_FRONT_CENTER,                                                     ///< 1, A

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

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

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

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

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

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

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

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

  144. 

  145.     AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_SIDE_LEFT |

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

  147. 

  148.     AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |

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

  150. 

  151.     AV_CH_LAYOUT_STEREO | AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT |

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

  153. 

  154.     AV_CH_FRONT_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |

  155.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_BACK_CENTER   |

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

  157. 

  158.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER   |

  159.     AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |

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

  161. 

  162.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |

  163.     AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |

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

  165. 

  166.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER   |

  167.     AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |

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

  169. };

  170. 

  171. static const int8_t dca_lfe_index[] = {

  172.     1, 2, 2, 2, 2, 3, 2, 3, 2, 3, 2, 3, 1, 3, 2, 3

  173. };

  174. 

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

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

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

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

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

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

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

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

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

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

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

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

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

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

  189.     { 4,  2,  5,  0,  1,  6,  7, -1, -1},

  190.     { 5,  6,  0,  1,  7,  3,  8,  4, -1},

  191.     { 4,  2,  5,  0,  1,  6,  8,  7, -1},

  192. };

  193. 

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

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

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

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

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

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

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

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

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

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

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

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

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

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

  208.     { 4,  2,  5,  0,  1,  7,  8,  6, -1},

  209.     { 5,  6,  0,  1,  8,  3,  9,  4,  7},

  210.     { 4,  2,  5,  0,  1,  6,  9,  8,  7},

  211. };

  212. 

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

  214.     { 0, -1, -1, -1, -1, -1, -1, -1, -1},

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

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

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

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

  219.     { 2,  0,  1, -1, -1, -1, -1, -1, -1},

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

  221.     { 2,  0,  1,  3, -1, -1, -1, -1, -1},

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

  223.     { 2,  0,  1,  3,  4, -1, -1, -1, -1},

  224.     { 2,  3,  0,  1,  4,  5, -1, -1, -1},

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

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

  227.     { 3,  2,  4,  0,  1,  5,  6, -1, -1},

  228.     { 4,  5,  0,  1,  6,  2,  7,  3, -1},

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

  230. };

  231. 

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

  233.     { 0,  1, -1, -1, -1, -1, -1, -1, -1},

  234.     { 0,  1,  2, -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.     { 2,  0,  1,  3, -1, -1, -1, -1, -1},

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

  240.     { 2,  0,  1,  3,  4, -1, -1, -1, -1},

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

  242.     { 2,  0,  1,  4,  5,  3, -1, -1, -1},

  243.     { 2,  3,  0,  1,  5,  6,  4, -1, -1},

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

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

  246.     { 3,  2,  4,  0,  1,  6,  7,  5, -1},

  247.     { 4,  5,  0,  1,  7,  2,  8,  3,  6},

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

  249. };

  250. 

  251. #define DCA_DOLBY                  101           /* FIXME */

  252. 

  253. #define DCA_CHANNEL_BITS             6

  254. #define DCA_CHANNEL_MASK          0x3F

  255. 

  256. #define DCA_LFE                   0x80

  257. 

  258. #define HEADER_SIZE                 14

  259. 

  260. #define DCA_MAX_FRAME_SIZE       16384

  261. #define DCA_MAX_EXSS_HEADER_SIZE  4096

  262. 

  263. #define DCA_BUFFER_PADDING_SIZE   1024

  264. 

  265. /** Bit allocation */

  266. typedef struct {

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

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

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

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

  271. } BitAlloc;

  272. 

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

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

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

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

  277. 

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

  279.                                          int idx)

  280. {

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

  282.            ba->offset;

  283. }

  284. 

  285. typedef struct {

  286.     AVCodecContext *avctx;

  287.     AVFrame frame;

  288.     /* Frame header */

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

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

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

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

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

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

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

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

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

  298. 

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

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

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

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

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

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

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

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

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

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

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

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

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

  312.     int copy_history;           ///< copy history

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

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

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

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

  317. 

  318.     /* Primary audio coding header */

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

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

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

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

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

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

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

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

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

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

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

  330. 

  331.     /* Primary audio coding side information */

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

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

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

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

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

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

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

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

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

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

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

  343. 

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

  345. 

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

  347.     int lfe_scale_factor;

  348. 

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

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

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

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

  353.     int hist_index[DCA_PRIM_CHANNELS_MAX];

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

  355. 

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

  357.     float scale_bias;           ///< output scale

  358. 

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

  360.     DECLARE_ALIGNED(32, float, samples)[(DCA_PRIM_CHANNELS_MAX + 1) * 256];

  361.     const float *samples_chanptr[DCA_PRIM_CHANNELS_MAX + 1];

  362. 

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

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

  365. 

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

  367.     GetBitContext gb;

  368.     /* Current position in DCA frame */

  369.     int current_subframe;

  370.     int current_subsubframe;

  371. 

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

  373. 

  374.     /* XCh extension information */

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

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

  377. 

  378.     /* ExSS header parser */

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

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

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

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

  383. 

  384.     int profile;

  385. 

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

  387.     DSPContext dsp;

  388.     FFTContext imdct;

  389.     SynthFilterContext synth;

  390.     DCADSPContext dcadsp;

  391.     FmtConvertContext fmt_conv;

  392. } DCAContext;

  393. 

  394. static const uint16_t dca_vlc_offs[] = {

  395.         0,   512,   640,   768,  1282,  1794,  2436,  3080,  3770,  4454,  5364,

  396.      5372,  5380,  5388,  5392,  5396,  5412,  5420,  5428,  5460,  5492,  5508,

  397.      5572,  5604,  5668,  5796,  5860,  5892,  6412,  6668,  6796,  7308,  7564,

  398.      7820,  8076,  8620,  9132,  9388,  9910, 10166, 10680, 11196, 11726, 12240,

  399.     12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264,

  400.     18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622,

  401. };

  402. 

  403. static av_cold void dca_init_vlcs(void)

  404. {

  405.     static int vlcs_initialized = 0;

  406.     int i, j, c = 14;

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

  408. 

  409.     if (vlcs_initialized)

  410.         return;

  411. 

  412.     dca_bitalloc_index.offset = 1;

  413.     dca_bitalloc_index.wrap = 2;

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

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

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

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

  418.                  bitalloc_12_bits[i], 1, 1,

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

  420.     }

  421.     dca_scalefactor.offset = -64;

  422.     dca_scalefactor.wrap = 2;

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

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

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

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

  427.                  scales_bits[i], 1, 1,

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

  429.     }

  430.     dca_tmode.offset = 0;

  431.     dca_tmode.wrap = 1;

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

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

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

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

  436.                  tmode_bits[i], 1, 1,

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

  438.     }

  439. 

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

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

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

  443.                 break;

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

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

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

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

  448. 

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

  450.                      bitalloc_sizes[i],

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

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

  453.             c++;

  454.         }

  455.     vlcs_initialized = 1;

  456. }

  457. 

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

  459. {

  460.     while (len--)

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

  462. }

  463. 

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

  465. {

  466.     int i, j;

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

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

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

  470. 

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

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

  473. 

  474.     if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)

  475.         s->prim_channels = DCA_PRIM_CHANNELS_MAX;

  476. 

  477. 

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

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

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

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

  482.     }

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

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

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

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

  487.     }

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

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

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

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

  492. 

  493.     /* Get codebooks quantization indexes */

  494.     if (!base_channel)

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

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

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

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

  499. 

  500.     /* Get scale factor adjustment */

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

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

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

  504. 

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

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

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

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

  509. 

  510.     if (s->crc_present) {

  511.         /* Audio header CRC check */

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

  513.     }

  514. 

  515.     s->current_subframe    = 0;

  516.     s->current_subsubframe = 0;

  517. 

  518. #ifdef TRACE

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

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

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

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

  523.                s->subband_activity[i]);

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

  525.                s->vq_start_subband[i]);

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

  527.                s->joint_intensity[i]);

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

  529.                s->transient_huffman[i]);

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

  531.                s->scalefactor_huffman[i]);

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

  533.                s->bitalloc_huffman[i]);

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

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

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

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

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

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

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

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

  542.     }

  543. #endif

  544. 

  545.     return 0;

  546. }

  547. 

  548. static int dca_parse_frame_header(DCAContext *s)

  549. {

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

  551. 

  552.     /* Sync code */

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

  554. 

  555.     /* Frame header */

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

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

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

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

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

  561.     if (s->frame_size < 95)

  562.         return AVERROR_INVALIDDATA;

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

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

  565.     if (!s->sample_rate)

  566.         return AVERROR_INVALIDDATA;

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

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

  569.     if (!s->bit_rate)

  570.         return AVERROR_INVALIDDATA;

  571. 

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

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

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

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

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

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

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

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

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

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

  582. 

  583.     /* TODO: check CRC */

  584.     if (s->crc_present)

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

  586. 

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

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

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

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

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

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

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

  594. 

  595.     /* FIXME: channels mixing levels */

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

  597.     if (s->lfe)

  598.         s->output |= DCA_LFE;

  599. 

  600. #ifdef TRACE

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

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

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

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

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

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

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

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

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

  610.            s->sample_rate);

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

  612.            s->bit_rate);

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

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

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

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

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

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

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

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

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

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

  623.            s->predictor_history);

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

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

  626.            s->multirate_inter);

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

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

  629.     av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

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

  636. #endif

  637. 

  638.     /* Primary audio coding header */

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

  640. 

  641.     return dca_parse_audio_coding_header(s, 0);

  642. }

  643. 

  644. 

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

  646. {

  647.     if (level < 5) {

  648.         /* huffman encoded */

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

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

  651.     } else if (level < 8) {

  652.         if (level + 1 > log2range) {

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

  654.             value = get_bits(gb, log2range);

  655.         } else {

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

  657.         }

  658.     }

  659.     return value;

  660. }

  661. 

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

  663. {

  664.     /* Primary audio coding side information */

  665.     int j, k;

  666. 

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

  668.         return AVERROR_INVALIDDATA;

  669. 

  670.     if (!base_channel) {

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

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

  673.     }

  674. 

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

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

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

  678.     }

  679. 

  680.     /* Get prediction codebook */

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

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

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

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

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

  686.             }

  687.         }

  688.     }

  689. 

  690.     /* Bit allocation index */

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

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

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

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

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

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

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

  698.                 av_log(s->avctx, AV_LOG_ERROR,

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

  700.                 return AVERROR_INVALIDDATA;

  701.             } else {

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

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

  704.             }

  705. 

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

  707.                 // av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index [%i][%i] too big (%i)\n",

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

  709.                 return AVERROR_INVALIDDATA;

  710.             }

  711.         }

  712.     }

  713. 

  714.     /* Transition mode */

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

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

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

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

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

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

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

  722.             }

  723.         }

  724.     }

  725. 

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

  727.         return AVERROR_INVALIDDATA;

  728. 

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

  730.         const uint32_t *scale_table;

  731.         int scale_sum, log_size;

  732. 

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

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

  735. 

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

  737.             scale_table = scale_factor_quant7;

  738.             log_size = 7;

  739.         } else {

  740.             scale_table = scale_factor_quant6;

  741.             log_size = 6;

  742.         }

  743. 

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

  745.         scale_sum = 0;

  746. 

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

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

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

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

  751.             }

  752. 

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

  754.                 /* Get second scale factor */

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

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

  757.             }

  758.         }

  759.     }

  760. 

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

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

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

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

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

  766.     }

  767. 

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

  769.         return AVERROR_INVALIDDATA;

  770. 

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

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

  773.         int source_channel;

  774. 

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

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

  777.             int scale = 0;

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

  779. 

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

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

  782. 

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

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

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

  786.             }

  787. 

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

  789.                 av_log(s->avctx, AV_LOG_DEBUG,

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

  791.                 s->debug_flag |= 0x02;

  792.             }

  793.         }

  794.     }

  795. 

  796.     /* Stereo downmix coefficients */

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

  798.         if (s->downmix) {

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

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

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

  802.             }

  803.         } else {

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

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

  806.                 av_log(s->avctx, AV_LOG_ERROR,

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

  808.                 return AVERROR_INVALIDDATA;

  809.             }

  810.             for (j = base_channel; j < FFMIN(s->prim_channels, FF_ARRAY_ELEMS(dca_default_coeffs[am])); j++) {

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

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

  813.             }

  814.         }

  815.     }

  816. 

  817.     /* Dynamic range coefficient */

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

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

  820. 

  821.     /* Side information CRC check word */

  822.     if (s->crc_present) {

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

  824.     }

  825. 

  826.     /*

  827.      * Primary audio data arrays

  828.      */

  829. 

  830.     /* VQ encoded high frequency subbands */

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

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

  833.             /* 1 vector -> 32 samples */

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

  835. 

  836.     /* Low frequency effect data */

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

  838.         int quant7;

  839.         /* LFE samples */

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

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

  842.         float lfe_scale;

  843. 

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

  845.             /* Signed 8 bits int */

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

  847.         }

  848. 

  849.         /* Scale factor index */

  850.         quant7 = get_bits(&s->gb, 8);

  851.         if (quant7 > 127) {

  852.             av_log_ask_for_sample(s->avctx, "LFEScaleIndex larger than 127\n");

  853.             return AVERROR_INVALIDDATA;

  854.         }

  855.         s->lfe_scale_factor = scale_factor_quant7[quant7];

  856. 

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

  858.         lfe_scale = 0.035 * s->lfe_scale_factor;

  859. 

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

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

  862.     }

  863. 

  864. #ifdef TRACE

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

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

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

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

  869. 

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

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

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

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

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

  875.     }

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

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

  878.             av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

  884.     }

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

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

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

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

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

  890.     }

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

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

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

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

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

  896.     }

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

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

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

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

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

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

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

  904.         }

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

  906.     }

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

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

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

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

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

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

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

  914.         }

  915.     }

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

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

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

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

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

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

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

  923.         }

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

  925.     }

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

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

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

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

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

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

  932. 

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

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

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

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

  937.     }

  938. #endif

  939. 

  940.     return 0;

  941. }

  942. 

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

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

  945.                             float scale)

  946. {

  947.     const float *prCoeff;

  948.     int i;

  949. 

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

  951.     int subindex;

  952. 

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

  954. 

  955.     /* Select filter */

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

  957.         prCoeff = fir_32bands_nonperfect;

  958.     else                        /* Perfect reconstruction */

  959.         prCoeff = fir_32bands_perfect;

  960. 

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

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

  963. 

  964.     /* Reconstructed channel sample index */

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

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

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

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

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

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

  971.         }

  972. 

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

  974.                                     s->subband_fir_hist[chans],

  975.                                     &s->hist_index[chans],

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

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

  978.         samples_out += 32;

  979.     }

  980. }

  981. 

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

  983.                                   int num_deci_sample, float *samples_in,

  984.                                   float *samples_out, float scale)

  985. {

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

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

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

  989.      *   from last subframe as history.

  990.      *

  991.      * samples_out: An array holding interpolated samples

  992.      */

  993. 

  994.     int decifactor;

  995.     const float *prCoeff;

  996.     int deciindex;

  997. 

  998.     /* Select decimation filter */

  999.     if (decimation_select == 1) {

  1000.         decifactor = 64;

  1001.         prCoeff = lfe_fir_128;

  1002.     } else {

  1003.         decifactor = 32;

  1004.         prCoeff = lfe_fir_64;

  1005.     }

  1006.     /* Interpolation */

  1007.     for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {

  1008.         s->dcadsp.lfe_fir(samples_out, samples_in, prCoeff, decifactor, scale);

  1009.         samples_in++;

  1010.         samples_out += 2 * decifactor;

  1011.     }

  1012. }

  1013. 

  1014. /* downmixing routines */

  1015. #define MIX_REAR1(samples, si1, rs, coef)           \

  1016.     samples[i]     += samples[si1] * coef[rs][0];   \

  1017.     samples[i+256] += samples[si1] * coef[rs][1];

  1018. 

  1019. #define MIX_REAR2(samples, si1, si2, rs, coef)                                     \

  1020.     samples[i]     += samples[si1] * coef[rs][0] + samples[si2] * coef[rs + 1][0]; \

  1021.     samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs + 1][1];

  1022. 

  1023. #define MIX_FRONT3(samples, coef)                                      \

  1024.     t = samples[i + c];                                                \

  1025.     u = samples[i + l];                                                \

  1026.     v = samples[i + r];                                                \

  1027.     samples[i]     = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \

  1028.     samples[i+256] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];

  1029. 

  1030. #define DOWNMIX_TO_STEREO(op1, op2)             \

  1031.     for (i = 0; i < 256; i++) {                 \

  1032.         op1                                     \

  1033.         op2                                     \

  1034.     }

  1035. 

  1036. static void dca_downmix(float *samples, int srcfmt,

  1037.                         int downmix_coef[DCA_PRIM_CHANNELS_MAX][2],

  1038.                         const int8_t *channel_mapping)

  1039. {

  1040.     int c, l, r, sl, sr, s;

  1041.     int i;

  1042.     float t, u, v;

  1043.     float coef[DCA_PRIM_CHANNELS_MAX][2];

  1044. 

  1045.     for (i = 0; i < DCA_PRIM_CHANNELS_MAX; i++) {

  1046.         coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]];

  1047.         coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]];

  1048.     }

  1049. 

  1050.     switch (srcfmt) {

  1051.     case DCA_MONO:

  1052.     case DCA_CHANNEL:

  1053.     case DCA_STEREO_TOTAL:

  1054.     case DCA_STEREO_SUMDIFF:

  1055.     case DCA_4F2R:

  1056.         av_log(NULL, 0, "Not implemented!\n");

  1057.         break;

  1058.     case DCA_STEREO:

  1059.         break;

  1060.     case DCA_3F:

  1061.         c = channel_mapping[0] * 256;

  1062.         l = channel_mapping[1] * 256;

  1063.         r = channel_mapping[2] * 256;

  1064.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );

  1065.         break;

  1066.     case DCA_2F1R:

  1067.         s = channel_mapping[2] * 256;

  1068.         DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + s, 2, coef), );

  1069.         break;

  1070.     case DCA_3F1R:

  1071.         c = channel_mapping[0] * 256;

  1072.         l = channel_mapping[1] * 256;

  1073.         r = channel_mapping[2] * 256;

  1074.         s = channel_mapping[3] * 256;

  1075.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1076.                           MIX_REAR1(samples, i + s, 3, coef));

  1077.         break;

  1078.     case DCA_2F2R:

  1079.         sl = channel_mapping[2] * 256;

  1080.         sr = channel_mapping[3] * 256;

  1081.         DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + sl, i + sr, 2, coef), );

  1082.         break;

  1083.     case DCA_3F2R:

  1084.         c  = channel_mapping[0] * 256;

  1085.         l  = channel_mapping[1] * 256;

  1086.         r  = channel_mapping[2] * 256;

  1087.         sl = channel_mapping[3] * 256;

  1088.         sr = channel_mapping[4] * 256;

  1089.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1090.                           MIX_REAR2(samples, i + sl, i + sr, 3, coef));

  1091.         break;

  1092.     }

  1093. }

  1094. 

  1095. 

  1096. #ifndef decode_blockcodes

  1097. /* Very compact version of the block code decoder that does not use table

  1098.  * look-up but is slightly slower */

  1099. static int decode_blockcode(int code, int levels, int *values)

  1100. {

  1101.     int i;

  1102.     int offset = (levels - 1) >> 1;

  1103. 

  1104.     for (i = 0; i < 4; i++) {

  1105.         int div = FASTDIV(code, levels);

  1106.         values[i] = code - offset - div * levels;

  1107.         code = div;

  1108.     }

  1109. 

  1110.     return code;

  1111. }

  1112. 

  1113. static int decode_blockcodes(int code1, int code2, int levels, int *values)

  1114. {

  1115.     return decode_blockcode(code1, levels, values) |

  1116.            decode_blockcode(code2, levels, values + 4);

  1117. }

  1118. #endif

  1119. 

  1120. static const uint8_t abits_sizes[7]  = { 7, 10, 12, 13, 15, 17, 19 };

  1121. static const uint8_t abits_levels[7] = { 3,  5,  7,  9, 13, 17, 25 };

  1122. 

  1123. #ifndef int8x8_fmul_int32

  1124. static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)

  1125. {

  1126.     float fscale = scale / 16.0;

  1127.     int i;

  1128.     for (i = 0; i < 8; i++)

  1129.         dst[i] = src[i] * fscale;

  1130. }

  1131. #endif

  1132. 

  1133. static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)

  1134. {

  1135.     int k, l;

  1136.     int subsubframe = s->current_subsubframe;

  1137. 

  1138.     const float *quant_step_table;

  1139. 

  1140.     /* FIXME */

  1141.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1142.     LOCAL_ALIGNED_16(int, block, [8]);

  1143. 

  1144.     /*

  1145.      * Audio data

  1146.      */

  1147. 

  1148.     /* Select quantization step size table */

  1149.     if (s->bit_rate_index == 0x1f)

  1150.         quant_step_table = lossless_quant_d;

  1151.     else

  1152.         quant_step_table = lossy_quant_d;

  1153. 

  1154.     for (k = base_channel; k < s->prim_channels; k++) {

  1155.         if (get_bits_left(&s->gb) < 0)

  1156.             return AVERROR_INVALIDDATA;

  1157. 

  1158.         for (l = 0; l < s->vq_start_subband[k]; l++) {

  1159.             int m;

  1160. 

  1161.             /* Select the mid-tread linear quantizer */

  1162.             int abits = s->bitalloc[k][l];

  1163. 

  1164.             float quant_step_size = quant_step_table[abits];

  1165. 

  1166.             /*

  1167.              * Determine quantization index code book and its type

  1168.              */

  1169. 

  1170.             /* Select quantization index code book */

  1171.             int sel = s->quant_index_huffman[k][abits];

  1172. 

  1173.             /*

  1174.              * Extract bits from the bit stream

  1175.              */

  1176.             if (!abits) {

  1177.                 memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));

  1178.             } else {

  1179.                 /* Deal with transients */

  1180.                 int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];

  1181.                 float rscale = quant_step_size * s->scale_factor[k][l][sfi] *

  1182.                                s->scalefactor_adj[k][sel];

  1183. 

  1184.                 if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {

  1185.                     if (abits <= 7) {

  1186.                         /* Block code */

  1187.                         int block_code1, block_code2, size, levels, err;

  1188. 

  1189.                         size   = abits_sizes[abits - 1];

  1190.                         levels = abits_levels[abits - 1];

  1191. 

  1192.                         block_code1 = get_bits(&s->gb, size);

  1193.                         block_code2 = get_bits(&s->gb, size);

  1194.                         err = decode_blockcodes(block_code1, block_code2,

  1195.                                                 levels, block);

  1196.                         if (err) {

  1197.                             av_log(s->avctx, AV_LOG_ERROR,

  1198.                                    "ERROR: block code look-up failed\n");

  1199.                             return AVERROR_INVALIDDATA;

  1200.                         }

  1201.                     } else {

  1202.                         /* no coding */

  1203.                         for (m = 0; m < 8; m++)

  1204.                             block[m] = get_sbits(&s->gb, abits - 3);

  1205.                     }

  1206.                 } else {

  1207.                     /* Huffman coded */

  1208.                     for (m = 0; m < 8; m++)

  1209.                         block[m] = get_bitalloc(&s->gb,

  1210.                                                 &dca_smpl_bitalloc[abits], sel);

  1211.                 }

  1212. 

  1213.                 s->fmt_conv.int32_to_float_fmul_scalar(subband_samples[k][l],

  1214.                                                        block, rscale, 8);

  1215.             }

  1216. 

  1217.             /*

  1218.              * Inverse ADPCM if in prediction mode

  1219.              */

  1220.             if (s->prediction_mode[k][l]) {

  1221.                 int n;

  1222.                 for (m = 0; m < 8; m++) {

  1223.                     for (n = 1; n <= 4; n++)

  1224.                         if (m >= n)

  1225.                             subband_samples[k][l][m] +=

  1226.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1227.                                  subband_samples[k][l][m - n] / 8192);

  1228.                         else if (s->predictor_history)

  1229.                             subband_samples[k][l][m] +=

  1230.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1231.                                  s->subband_samples_hist[k][l][m - n + 4] / 8192);

  1232.                 }

  1233.             }

  1234.         }

  1235. 

  1236.         /*

  1237.          * Decode VQ encoded high frequencies

  1238.          */

  1239.         for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {

  1240.             /* 1 vector -> 32 samples but we only need the 8 samples

  1241.              * for this subsubframe. */

  1242.             int hfvq = s->high_freq_vq[k][l];

  1243. 

  1244.             if (!s->debug_flag & 0x01) {

  1245.                 av_log(s->avctx, AV_LOG_DEBUG,

  1246.                        "Stream with high frequencies VQ coding\n");

  1247.                 s->debug_flag |= 0x01;

  1248.             }

  1249. 

  1250.             int8x8_fmul_int32(subband_samples[k][l],

  1251.                               &high_freq_vq[hfvq][subsubframe * 8],

  1252.                               s->scale_factor[k][l][0]);

  1253.         }

  1254.     }

  1255. 

  1256.     /* Check for DSYNC after subsubframe */

  1257.     if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {

  1258.         if (0xFFFF == get_bits(&s->gb, 16)) {   /* 0xFFFF */

  1259. #ifdef TRACE

  1260.             av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");

  1261. #endif

  1262.         } else {

  1263.             av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");

  1264.         }

  1265.     }

  1266. 

  1267.     /* Backup predictor history for adpcm */

  1268.     for (k = base_channel; k < s->prim_channels; k++)

  1269.         for (l = 0; l < s->vq_start_subband[k]; l++)

  1270.             memcpy(s->subband_samples_hist[k][l],

  1271.                    &subband_samples[k][l][4],

  1272.                    4 * sizeof(subband_samples[0][0][0]));

  1273. 

  1274.     return 0;

  1275. }

  1276. 

  1277. static int dca_filter_channels(DCAContext *s, int block_index)

  1278. {

  1279.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1280.     int k;

  1281. 

  1282.     /* 32 subbands QMF */

  1283.     for (k = 0; k < s->prim_channels; k++) {

  1284. /*        static float pcm_to_double[8] = { 32768.0, 32768.0, 524288.0, 524288.0,

  1285.                                             0, 8388608.0, 8388608.0 };*/

  1286.         qmf_32_subbands(s, k, subband_samples[k],

  1287.                         &s->samples[256 * s->channel_order_tab[k]],

  1288.                         M_SQRT1_2 * s->scale_bias /* pcm_to_double[s->source_pcm_res] */);

  1289.     }

  1290. 

  1291.     /* Down mixing */

  1292.     if (s->avctx->request_channels == 2 && s->prim_channels > 2) {

  1293.         dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab);

  1294.     }

  1295. 

  1296.     /* Generate LFE samples for this subsubframe FIXME!!! */

  1297.     if (s->output & DCA_LFE) {

  1298.         lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,

  1299.                               s->lfe_data + 2 * s->lfe * (block_index + 4),

  1300.                               &s->samples[256 * dca_lfe_index[s->amode]],

  1301.                               (1.0 / 256.0) * s->scale_bias);

  1302.         /* Outputs 20bits pcm samples */

  1303.     }

  1304. 

  1305.     return 0;

  1306. }

  1307. 

  1308. 

  1309. static int dca_subframe_footer(DCAContext *s, int base_channel)

  1310. {

  1311.     int aux_data_count = 0, i;

  1312. 

  1313.     /*

  1314.      * Unpack optional information

  1315.      */

  1316. 

  1317.     /* presumably optional information only appears in the core? */

  1318.     if (!base_channel) {

  1319.         if (s->timestamp)

  1320.             skip_bits_long(&s->gb, 32);

  1321. 

  1322.         if (s->aux_data)

  1323.             aux_data_count = get_bits(&s->gb, 6);

  1324. 

  1325.         for (i = 0; i < aux_data_count; i++)

  1326.             get_bits(&s->gb, 8);

  1327. 

  1328.         if (s->crc_present && (s->downmix || s->dynrange))

  1329.             get_bits(&s->gb, 16);

  1330.     }

  1331. 

  1332.     return 0;

  1333. }

  1334. 

  1335. /**

  1336.  * Decode a dca frame block

  1337.  *

  1338.  * @param s     pointer to the DCAContext

  1339.  */

  1340. 

  1341. static int dca_decode_block(DCAContext *s, int base_channel, int block_index)

  1342. {

  1343.     int ret;

  1344. 

  1345.     /* Sanity check */

  1346.     if (s->current_subframe >= s->subframes) {

  1347.         av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",

  1348.                s->current_subframe, s->subframes);

  1349.         return AVERROR_INVALIDDATA;

  1350.     }

  1351. 

  1352.     if (!s->current_subsubframe) {

  1353. #ifdef TRACE

  1354.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");

  1355. #endif

  1356.         /* Read subframe header */

  1357.         if ((ret = dca_subframe_header(s, base_channel, block_index)))

  1358.             return ret;

  1359.     }

  1360. 

  1361.     /* Read subsubframe */

  1362. #ifdef TRACE

  1363.     av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");

  1364. #endif

  1365.     if ((ret = dca_subsubframe(s, base_channel, block_index)))

  1366.         return ret;

  1367. 

  1368.     /* Update state */

  1369.     s->current_subsubframe++;

  1370.     if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {

  1371.         s->current_subsubframe = 0;

  1372.         s->current_subframe++;

  1373.     }

  1374.     if (s->current_subframe >= s->subframes) {

  1375. #ifdef TRACE

  1376.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");

  1377. #endif

  1378.         /* Read subframe footer */

  1379.         if ((ret = dca_subframe_footer(s, base_channel)))

  1380.             return ret;

  1381.     }

  1382. 

  1383.     return 0;

  1384. }

  1385. 

  1386. /**

  1387.  * Return the number of channels in an ExSS speaker mask (HD)

  1388.  */

  1389. static int dca_exss_mask2count(int mask)

  1390. {

  1391.     /* count bits that mean speaker pairs twice */

  1392.     return av_popcount(mask) +

  1393.            av_popcount(mask & (DCA_EXSS_CENTER_LEFT_RIGHT      |

  1394.                                DCA_EXSS_FRONT_LEFT_RIGHT       |

  1395.                                DCA_EXSS_FRONT_HIGH_LEFT_RIGHT  |

  1396.                                DCA_EXSS_WIDE_LEFT_RIGHT        |

  1397.                                DCA_EXSS_SIDE_LEFT_RIGHT        |

  1398.                                DCA_EXSS_SIDE_HIGH_LEFT_RIGHT   |

  1399.                                DCA_EXSS_SIDE_REAR_LEFT_RIGHT   |

  1400.                                DCA_EXSS_REAR_LEFT_RIGHT        |

  1401.                                DCA_EXSS_REAR_HIGH_LEFT_RIGHT));

  1402. }

  1403. 

  1404. /**

  1405.  * Skip mixing coefficients of a single mix out configuration (HD)

  1406.  */

  1407. static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch)

  1408. {

  1409.     int i;

  1410. 

  1411.     for (i = 0; i < channels; i++) {

  1412.         int mix_map_mask = get_bits(gb, out_ch);

  1413.         int num_coeffs = av_popcount(mix_map_mask);

  1414.         skip_bits_long(gb, num_coeffs * 6);

  1415.     }

  1416. }

  1417. 

  1418. /**

  1419.  * Parse extension substream asset header (HD)

  1420.  */

  1421. static int dca_exss_parse_asset_header(DCAContext *s)

  1422. {

  1423.     int header_pos = get_bits_count(&s->gb);

  1424.     int header_size;

  1425.     int channels = 0;

  1426.     int embedded_stereo = 0;

  1427.     int embedded_6ch    = 0;

  1428.     int drc_code_present;

  1429.     int av_uninit(extensions_mask);

  1430.     int i, j;

  1431. 

  1432.     if (get_bits_left(&s->gb) < 16)

  1433.         return -1;

  1434. 

  1435.     /* We will parse just enough to get to the extensions bitmask with which

  1436.      * we can set the profile value. */

  1437. 

  1438.     header_size = get_bits(&s->gb, 9) + 1;

  1439.     skip_bits(&s->gb, 3); // asset index

  1440. 

  1441.     if (s->static_fields) {

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

  1443.             skip_bits(&s->gb, 4); // asset type descriptor

  1444.         if (get_bits1(&s->gb))

  1445.             skip_bits_long(&s->gb, 24); // language descriptor

  1446. 

  1447.         if (get_bits1(&s->gb)) {

  1448.             /* How can one fit 1024 bytes of text here if the maximum value

  1449.              * for the asset header size field above was 512 bytes? */

  1450.             int text_length = get_bits(&s->gb, 10) + 1;

  1451.             if (get_bits_left(&s->gb) < text_length * 8)

  1452.                 return -1;

  1453.             skip_bits_long(&s->gb, text_length * 8); // info text

  1454.         }

  1455. 

  1456.         skip_bits(&s->gb, 5); // bit resolution - 1

  1457.         skip_bits(&s->gb, 4); // max sample rate code

  1458.         channels = get_bits(&s->gb, 8) + 1;

  1459. 

  1460.         if (get_bits1(&s->gb)) { // 1-to-1 channels to speakers

  1461.             int spkr_remap_sets;

  1462.             int spkr_mask_size = 16;

  1463.             int num_spkrs[7];

  1464. 

  1465.             if (channels > 2)

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

  1467.             if (channels > 6)

  1468.                 embedded_6ch = get_bits1(&s->gb);

  1469. 

  1470.             if (get_bits1(&s->gb)) {

  1471.                 spkr_mask_size = (get_bits(&s->gb, 2) + 1) << 2;

  1472.                 skip_bits(&s->gb, spkr_mask_size); // spkr activity mask

  1473.             }

  1474. 

  1475.             spkr_remap_sets = get_bits(&s->gb, 3);

  1476. 

  1477.             for (i = 0; i < spkr_remap_sets; i++) {

  1478.                 /* std layout mask for each remap set */

  1479.                 num_spkrs[i] = dca_exss_mask2count(get_bits(&s->gb, spkr_mask_size));

  1480.             }

  1481. 

  1482.             for (i = 0; i < spkr_remap_sets; i++) {

  1483.                 int num_dec_ch_remaps = get_bits(&s->gb, 5) + 1;

  1484.                 if (get_bits_left(&s->gb) < 0)

  1485.                     return -1;

  1486. 

  1487.                 for (j = 0; j < num_spkrs[i]; j++) {

  1488.                     int remap_dec_ch_mask = get_bits_long(&s->gb, num_dec_ch_remaps);

  1489.                     int num_dec_ch = av_popcount(remap_dec_ch_mask);

  1490.                     skip_bits_long(&s->gb, num_dec_ch * 5); // remap codes

  1491.                 }

  1492.             }

  1493. 

  1494.         } else {

  1495.             skip_bits(&s->gb, 3); // representation type

  1496.         }

  1497.     }

  1498. 

  1499.     drc_code_present = get_bits1(&s->gb);

  1500.     if (drc_code_present)

  1501.         get_bits(&s->gb, 8); // drc code

  1502. 

  1503.     if (get_bits1(&s->gb))

  1504.         skip_bits(&s->gb, 5); // dialog normalization code

  1505. 

  1506.     if (drc_code_present && embedded_stereo)

  1507.         get_bits(&s->gb, 8); // drc stereo code

  1508. 

  1509.     if (s->mix_metadata && get_bits1(&s->gb)) {

  1510.         skip_bits(&s->gb, 1); // external mix

  1511.         skip_bits(&s->gb, 6); // post mix gain code

  1512. 

  1513.         if (get_bits(&s->gb, 2) != 3) // mixer drc code

  1514.             skip_bits(&s->gb, 3); // drc limit

  1515.         else

  1516.             skip_bits(&s->gb, 8); // custom drc code

  1517. 

  1518.         if (get_bits1(&s->gb)) // channel specific scaling

  1519.             for (i = 0; i < s->num_mix_configs; i++)

  1520.                 skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6); // scale codes

  1521.         else

  1522.             skip_bits_long(&s->gb, s->num_mix_configs * 6); // scale codes

  1523. 

  1524.         for (i = 0; i < s->num_mix_configs; i++) {

  1525.             if (get_bits_left(&s->gb) < 0)

  1526.                 return -1;

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

  1528.             if (embedded_6ch)

  1529.                 dca_exss_skip_mix_coeffs(&s->gb, 6, s->mix_config_num_ch[i]);

  1530.             if (embedded_stereo)

  1531.                 dca_exss_skip_mix_coeffs(&s->gb, 2, s->mix_config_num_ch[i]);

  1532.         }

  1533.     }

  1534. 

  1535.     switch (get_bits(&s->gb, 2)) {

  1536.     case 0: extensions_mask = get_bits(&s->gb, 12); break;

  1537.     case 1: extensions_mask = DCA_EXT_EXSS_XLL;     break;

  1538.     case 2: extensions_mask = DCA_EXT_EXSS_LBR;     break;

  1539.     case 3: extensions_mask = 0; /* aux coding */   break;

  1540.     }

  1541. 

  1542.     /* not parsed further, we were only interested in the extensions mask */

  1543. 

  1544.     if (get_bits_left(&s->gb) < 0)

  1545.         return -1;

  1546. 

  1547.     if (get_bits_count(&s->gb) - header_pos > header_size * 8) {

  1548.         av_log(s->avctx, AV_LOG_WARNING, "Asset header size mismatch.\n");

  1549.         return -1;

  1550.     }

  1551.     skip_bits_long(&s->gb, header_pos + header_size * 8 - get_bits_count(&s->gb));

  1552. 

  1553.     if (extensions_mask & DCA_EXT_EXSS_XLL)

  1554.         s->profile = FF_PROFILE_DTS_HD_MA;

  1555.     else if (extensions_mask & (DCA_EXT_EXSS_XBR | DCA_EXT_EXSS_X96 |

  1556.                                 DCA_EXT_EXSS_XXCH))

  1557.         s->profile = FF_PROFILE_DTS_HD_HRA;

  1558. 

  1559.     if (!(extensions_mask & DCA_EXT_CORE))

  1560.         av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n");

  1561.     if ((extensions_mask & DCA_CORE_EXTS) != s->core_ext_mask)

  1562.         av_log(s->avctx, AV_LOG_WARNING,

  1563.                "DTS extensions detection mismatch (%d, %d)\n",

  1564.                extensions_mask & DCA_CORE_EXTS, s->core_ext_mask);

  1565. 

  1566.     return 0;

  1567. }

  1568. 

  1569. static int dca_xbr_parse_frame(DCAContext *s)

  1570. {

  1571.     int scale_table_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS][2];

  1572.     int active_bands[DCA_CHSETS_MAX][DCA_CHSET_CHANS_MAX];

  1573.     int abits_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS];

  1574.     int anctemp[DCA_CHSET_CHANS_MAX];

  1575.     int chset_fsize[DCA_CHSETS_MAX];

  1576.     int n_xbr_ch[DCA_CHSETS_MAX];

  1577.     int hdr_size, num_chsets, xbr_tmode, hdr_pos;

  1578.     int i, j, k, l, chset, chan_base;

  1579. 

  1580.     av_log(s->avctx, AV_LOG_DEBUG, "DTS-XBR: decoding XBR extension\n");

  1581. 

  1582.     /* get bit position of sync header */

  1583.     hdr_pos = get_bits_count(&s->gb) - 32;

  1584. 

  1585.     hdr_size = get_bits(&s->gb, 6) + 1;

  1586.     num_chsets = get_bits(&s->gb, 2) + 1;

  1587. 

  1588.     for(i = 0; i < num_chsets; i++)

  1589.         chset_fsize[i] = get_bits(&s->gb, 14) + 1;

  1590. 

  1591.     xbr_tmode = get_bits1(&s->gb);

  1592. 

  1593.     for(i = 0; i < num_chsets; i++) {

  1594.         n_xbr_ch[i] = get_bits(&s->gb, 3) + 1;

  1595.         k = get_bits(&s->gb, 2) + 5;

  1596.         for(j = 0; j < n_xbr_ch[i]; j++)

  1597.             active_bands[i][j] = get_bits(&s->gb, k) + 1;

  1598.     }

  1599. 

  1600.     /* skip to the end of the header */

  1601.     i = get_bits_count(&s->gb);

  1602.     if(hdr_pos + hdr_size * 8 > i)

  1603.         skip_bits_long(&s->gb, hdr_pos + hdr_size * 8 - i);

  1604. 

  1605.     /* loop over the channel data sets */

  1606.     /* only decode as many channels as we've decoded base data for */

  1607.     for(chset = 0, chan_base = 0;

  1608.         chset < num_chsets && chan_base + n_xbr_ch[chset] <= s->prim_channels;

  1609.         chan_base += n_xbr_ch[chset++]) {

  1610.         int start_posn = get_bits_count(&s->gb);

  1611.         int subsubframe = 0;

  1612.         int subframe = 0;

  1613. 

  1614.         /* loop over subframes */

  1615.         for (k = 0; k < (s->sample_blocks / 8); k++) {

  1616.             /* parse header if we're on first subsubframe of a block */

  1617.             if(subsubframe == 0) {

  1618.                 /* Parse subframe header */

  1619.                 for(i = 0; i < n_xbr_ch[chset]; i++) {

  1620.                     anctemp[i] = get_bits(&s->gb, 2) + 2;

  1621.                 }

  1622. 

  1623.                 for(i = 0; i < n_xbr_ch[chset]; i++) {

  1624.                     get_array(&s->gb, abits_high[i], active_bands[chset][i], anctemp[i]);

  1625.                 }

  1626. 

  1627.                 for(i = 0; i < n_xbr_ch[chset]; i++) {

  1628.                     anctemp[i] = get_bits(&s->gb, 3);

  1629.                     if(anctemp[i] < 1) {

  1630.                         av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: SYNC ERROR\n");

  1631.                         return AVERROR_INVALIDDATA;

  1632.                     }

  1633.                 }

  1634. 

  1635.                 /* generate scale factors */

  1636.                 for(i = 0; i < n_xbr_ch[chset]; i++) {

  1637.                     const uint32_t *scale_table;

  1638.                     int nbits;

  1639. 

  1640.                     if (s->scalefactor_huffman[chan_base+i] == 6) {

  1641.                         scale_table = scale_factor_quant7;

  1642.                     } else {

  1643.                         scale_table = scale_factor_quant6;

  1644.                     }

  1645. 

  1646.                     nbits = anctemp[i];

  1647. 

  1648.                     for(j = 0; j < active_bands[chset][i]; j++) {

  1649.                         if(abits_high[i][j] > 0) {

  1650.                             scale_table_high[i][j][0] =

  1651.                                 scale_table[get_bits(&s->gb, nbits)];

  1652. 

  1653.                             if(xbr_tmode && s->transition_mode[i][j]) {

  1654.                                 scale_table_high[i][j][1] =

  1655.                                     scale_table[get_bits(&s->gb, nbits)];

  1656.                             }

  1657.                         }

  1658.                     }

  1659.                 }

  1660.             }

  1661. 

  1662.             /* decode audio array for this block */

  1663.             for(i = 0; i < n_xbr_ch[chset]; i++) {

  1664.                 for(j = 0; j < active_bands[chset][i]; j++) {

  1665.                     const int xbr_abits = abits_high[i][j];

  1666.                     const float quant_step_size = lossless_quant_d[xbr_abits];

  1667.                     const int sfi = xbr_tmode && s->transition_mode[i][j] && subsubframe >= s->transition_mode[i][j];

  1668.                     const float rscale = quant_step_size * scale_table_high[i][j][sfi];

  1669.                     float *subband_samples = s->subband_samples[k][chan_base+i][j];

  1670.                     int block[8];

  1671. 

  1672.                     if(xbr_abits <= 0)

  1673.                         continue;

  1674. 

  1675.                     if(xbr_abits > 7) {

  1676.                         get_array(&s->gb, block, 8, xbr_abits - 3);

  1677.                     } else {

  1678.                         int block_code1, block_code2, size, levels, err;

  1679. 

  1680.                         size   = abits_sizes[xbr_abits - 1];

  1681.                         levels = abits_levels[xbr_abits - 1];

  1682. 

  1683.                         block_code1 = get_bits(&s->gb, size);

  1684.                         block_code2 = get_bits(&s->gb, size);

  1685.                         err = decode_blockcodes(block_code1, block_code2,

  1686.                                                 levels, block);

  1687.                         if (err) {

  1688.                             av_log(s->avctx, AV_LOG_ERROR,

  1689.                                    "ERROR: DTS-XBR: block code look-up failed\n");

  1690.                             return AVERROR_INVALIDDATA;

  1691.                         }

  1692.                     }

  1693. 

  1694.                     /* scale & sum into subband */

  1695.                     for(l = 0; l < 8; l++)

  1696.                         subband_samples[l] += (float)block[l] * rscale;

  1697.                 }

  1698.             }

  1699. 

  1700.             /* check DSYNC marker */

  1701.             if(s->aspf || subsubframe == s->subsubframes[subframe] - 1) {

  1702.                 if(get_bits(&s->gb, 16) != 0xffff) {

  1703.                     av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: Didn't get subframe DSYNC\n");

  1704.                     return AVERROR_INVALIDDATA;

  1705.                 }

  1706.             }

  1707. 

  1708.             /* advance sub-sub-frame index */

  1709.             if(++subsubframe >= s->subsubframes[subframe]) {

  1710.                 subsubframe = 0;

  1711.                 subframe++;

  1712.             }

  1713.         }

  1714. 

  1715.         /* skip to next channel set */

  1716.         i = get_bits_count(&s->gb);

  1717.         if(start_posn + chset_fsize[chset] * 8 != i) {

  1718.             j = start_posn + chset_fsize[chset] * 8 - i;

  1719.             if(j < 0 || j >= 8)

  1720.                 av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: end of channel set,"

  1721.                        " skipping further than expected (%d bits)\n", j);

  1722.             skip_bits_long(&s->gb, j);

  1723.         }

  1724.     }

  1725. 

  1726.     return 0;

  1727. }

  1728. 

  1729. /**

  1730.  * Parse extension substream header (HD)

  1731.  */

  1732. static void dca_exss_parse_header(DCAContext *s)

  1733. {

  1734.     int ss_index;

  1735.     int blownup;

  1736.     int num_audiop = 1;

  1737.     int num_assets = 1;

  1738.     int active_ss_mask[8];

  1739.     int i, j;

  1740.     int start_posn;

  1741.     int hdrsize;

  1742.     uint32_t mkr;

  1743. 

  1744.     if (get_bits_left(&s->gb) < 52)

  1745.         return;

  1746. 

  1747.     start_posn = get_bits_count(&s->gb) - 32;

  1748. 

  1749.     skip_bits(&s->gb, 8); // user data

  1750.     ss_index = get_bits(&s->gb, 2);

  1751. 

  1752.     blownup = get_bits1(&s->gb);

  1753.     hdrsize = get_bits(&s->gb,  8 + 4 * blownup) + 1; // header_size

  1754.     skip_bits(&s->gb, 16 + 4 * blownup); // hd_size

  1755. 

  1756.     s->static_fields = get_bits1(&s->gb);

  1757.     if (s->static_fields) {

  1758.         skip_bits(&s->gb, 2); // reference clock code

  1759.         skip_bits(&s->gb, 3); // frame duration code

  1760. 

  1761.         if (get_bits1(&s->gb))

  1762.             skip_bits_long(&s->gb, 36); // timestamp

  1763. 

  1764.         /* a single stream can contain multiple audio assets that can be

  1765.          * combined to form multiple audio presentations */

  1766. 

  1767.         num_audiop = get_bits(&s->gb, 3) + 1;

  1768.         if (num_audiop > 1) {

  1769.             av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio presentations.");

  1770.             /* ignore such streams for now */

  1771.             return;

  1772.         }

  1773. 

  1774.         num_assets = get_bits(&s->gb, 3) + 1;

  1775.         if (num_assets > 1) {

  1776.             av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio assets.");

  1777.             /* ignore such streams for now */

  1778.             return;

  1779.         }

  1780. 

  1781.         for (i = 0; i < num_audiop; i++)

  1782.             active_ss_mask[i] = get_bits(&s->gb, ss_index + 1);

  1783. 

  1784.         for (i = 0; i < num_audiop; i++)

  1785.             for (j = 0; j <= ss_index; j++)

  1786.                 if (active_ss_mask[i] & (1 << j))

  1787.                     skip_bits(&s->gb, 8); // active asset mask

  1788. 

  1789.         s->mix_metadata = get_bits1(&s->gb);

  1790.         if (s->mix_metadata) {

  1791.             int mix_out_mask_size;

  1792. 

  1793.             skip_bits(&s->gb, 2); // adjustment level

  1794.             mix_out_mask_size  = (get_bits(&s->gb, 2) + 1) << 2;

  1795.             s->num_mix_configs =  get_bits(&s->gb, 2) + 1;

  1796. 

  1797.             for (i = 0; i < s->num_mix_configs; i++) {

  1798.                 int mix_out_mask        = get_bits(&s->gb, mix_out_mask_size);

  1799.                 s->mix_config_num_ch[i] = dca_exss_mask2count(mix_out_mask);

  1800.             }

  1801.         }

  1802.     }

  1803. 

  1804.     for (i = 0; i < num_assets; i++)

  1805.         skip_bits_long(&s->gb, 16 + 4 * blownup);  // asset size

  1806. 

  1807.     for (i = 0; i < num_assets; i++) {

  1808.         if (dca_exss_parse_asset_header(s))

  1809.             return;

  1810.     }

  1811. 

  1812.     /* not parsed further, we were only interested in the extensions mask

  1813.      * from the asset header */

  1814. 

  1815.     if(num_assets > 0) {

  1816.         j = get_bits_count(&s->gb);

  1817.         if(start_posn + hdrsize * 8 > j)

  1818.             skip_bits_long(&s->gb, start_posn + hdrsize * 8 - j);

  1819. 

  1820.         /* check first asset for XBR -- should also check extension mask! */

  1821.         mkr = get_bits_long(&s->gb, 32);

  1822. 

  1823.         if(mkr == 0x655e315e)

  1824.             dca_xbr_parse_frame(s);

  1825.     }

  1826. }

  1827. 

  1828. /**

  1829.  * Main frame decoding function

  1830.  * FIXME add arguments

  1831.  */

  1832. static int dca_decode_frame(AVCodecContext *avctx, void *data,

  1833.                             int *got_frame_ptr, AVPacket *avpkt)

  1834. {

  1835.     const uint8_t *buf = avpkt->data;

  1836.     int buf_size = avpkt->size;

  1837. 

  1838.     int lfe_samples;

  1839.     int num_core_channels = 0;

  1840.     int i, ret;

  1841.     float   *samples_flt;

  1842.     int16_t *samples_s16;

  1843.     DCAContext *s = avctx->priv_data;

  1844.     int channels;

  1845.     int core_ss_end;

  1846. 

  1847. 

  1848.     s->xch_present = 0;

  1849. 

  1850.     s->dca_buffer_size = ff_dca_convert_bitstream(buf, buf_size, s->dca_buffer,

  1851.                                                   DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);

  1852.     if (s->dca_buffer_size == AVERROR_INVALIDDATA) {

  1853.         av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");

  1854.         return AVERROR_INVALIDDATA;

  1855.     }

  1856. 

  1857.     init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);

  1858.     if ((ret = dca_parse_frame_header(s)) < 0) {

  1859.         //seems like the frame is corrupt, try with the next one

  1860.         return ret;

  1861.     }

  1862.     //set AVCodec values with parsed data

  1863.     avctx->sample_rate = s->sample_rate;

  1864.     avctx->bit_rate    = s->bit_rate;

  1865. 

  1866.     s->profile = FF_PROFILE_DTS;

  1867. 

  1868.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1869.         if ((ret = dca_decode_block(s, 0, i))) {

  1870.             av_log(avctx, AV_LOG_ERROR, "error decoding block\n");

  1871.             return ret;

  1872.         }

  1873.     }

  1874. 

  1875.     /* record number of core channels incase less than max channels are requested */

  1876.     num_core_channels = s->prim_channels;

  1877. 

  1878.     if (s->ext_coding)

  1879.         s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];

  1880.     else

  1881.         s->core_ext_mask = 0;

  1882. 

  1883.     core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;

  1884. 

  1885.     /* only scan for extensions if ext_descr was unknown or indicated a

  1886.      * supported XCh extension */

  1887.     if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {

  1888. 

  1889.         /* if ext_descr was unknown, clear s->core_ext_mask so that the

  1890.          * extensions scan can fill it up */

  1891.         s->core_ext_mask = FFMAX(s->core_ext_mask, 0);

  1892. 

  1893.         /* extensions start at 32-bit boundaries into bitstream */

  1894.         skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1895. 

  1896.         while (core_ss_end - get_bits_count(&s->gb) >= 32) {

  1897.             uint32_t bits = get_bits_long(&s->gb, 32);

  1898. 

  1899.             switch (bits) {

  1900.             case 0x5a5a5a5a: {

  1901.                 int ext_amode, xch_fsize;

  1902. 

  1903.                 s->xch_base_channel = s->prim_channels;

  1904. 

  1905.                 /* validate sync word using XCHFSIZE field */

  1906.                 xch_fsize = show_bits(&s->gb, 10);

  1907.                 if ((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&

  1908.                     (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))

  1909.                     continue;

  1910. 

  1911.                 /* skip length-to-end-of-frame field for the moment */

  1912.                 skip_bits(&s->gb, 10);

  1913. 

  1914.                 s->core_ext_mask |= DCA_EXT_XCH;

  1915. 

  1916.                 /* extension amode(number of channels in extension) should be 1 */

  1917.                 /* AFAIK XCh is not used for more channels */

  1918.                 if ((ext_amode = get_bits(&s->gb, 4)) != 1) {

  1919.                     av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"

  1920.                            " supported!\n", ext_amode);

  1921.                     continue;

  1922.                 }

  1923. 

  1924.                 /* much like core primary audio coding header */

  1925.                 dca_parse_audio_coding_header(s, s->xch_base_channel);

  1926. 

  1927.                 for (i = 0; i < (s->sample_blocks / 8); i++)

  1928.                     if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {

  1929.                         av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");

  1930.                         continue;

  1931.                     }

  1932. 

  1933.                 s->xch_present = 1;

  1934.                 break;

  1935.             }

  1936.             case 0x47004a03:

  1937.                 /* XXCh: extended channels */

  1938.                 /* usually found either in core or HD part in DTS-HD HRA streams,

  1939.                  * but not in DTS-ES which contains XCh extensions instead */

  1940.                 s->core_ext_mask |= DCA_EXT_XXCH;

  1941.                 break;

  1942. 

  1943.             case 0x1d95f262: {

  1944.                 int fsize96 = show_bits(&s->gb, 12) + 1;

  1945.                 if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)

  1946.                     continue;

  1947. 

  1948.                 av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n",

  1949.                        get_bits_count(&s->gb));

  1950.                 skip_bits(&s->gb, 12);

  1951.                 av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);

  1952.                 av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));

  1953. 

  1954.                 s->core_ext_mask |= DCA_EXT_X96;

  1955.                 break;

  1956.             }

  1957.             }

  1958. 

  1959.             skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1960.         }

  1961.     } else {

  1962.         /* no supported extensions, skip the rest of the core substream */

  1963.         skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));

  1964.     }

  1965. 

  1966.     if (s->core_ext_mask & DCA_EXT_X96)

  1967.         s->profile = FF_PROFILE_DTS_96_24;

  1968.     else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))

  1969.         s->profile = FF_PROFILE_DTS_ES;

  1970. 

  1971.     /* check for ExSS (HD part) */

  1972.     if (s->dca_buffer_size - s->frame_size > 32 &&

  1973.         get_bits_long(&s->gb, 32) == DCA_HD_MARKER)

  1974.         dca_exss_parse_header(s);

  1975. 

  1976.     avctx->profile = s->profile;

  1977. 

  1978.     channels = s->prim_channels + !!s->lfe;

  1979. 

  1980.     if (s->amode < 16) {

  1981.         avctx->channel_layout = dca_core_channel_layout[s->amode];

  1982. 

  1983.         if (s->xch_present && (!avctx->request_channels ||

  1984.                                avctx->request_channels > num_core_channels + !!s->lfe)) {

  1985.             avctx->channel_layout |= AV_CH_BACK_CENTER;

  1986.             if (s->lfe) {

  1987.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1988.                 s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode];

  1989.             } else {

  1990.                 s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];

  1991.             }

  1992.         } else {

  1993.             channels = num_core_channels + !!s->lfe;

  1994.             s->xch_present = 0; /* disable further xch processing */

  1995.             if (s->lfe) {

  1996.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1997.                 s->channel_order_tab = dca_channel_reorder_lfe[s->amode];

  1998.             } else

  1999.                 s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];

  2000.         }

  2001. 

  2002.         if (channels > !!s->lfe &&

  2003.             s->channel_order_tab[channels - 1 - !!s->lfe] < 0)

  2004.             return AVERROR_INVALIDDATA;

  2005. 

  2006.         if (avctx->request_channels == 2 && s->prim_channels > 2) {

  2007.             channels = 2;

  2008.             s->output = DCA_STEREO;

  2009.             avctx->channel_layout = AV_CH_LAYOUT_STEREO;

  2010.         }

  2011.         else if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) {

  2012.             static const int8_t dca_channel_order_native[9] = { 0, 1, 2, 3, 4, 5, 6, 7, 8 };

  2013.             s->channel_order_tab = dca_channel_order_native;

  2014.         }

  2015.     } else {

  2016.         av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);

  2017.         return AVERROR_INVALIDDATA;

  2018.     }

  2019. 

  2020.     if (avctx->channels != channels) {

  2021.         if (avctx->channels)

  2022.             av_log(avctx, AV_LOG_INFO, "Number of channels changed in DCA decoder (%d -> %d)\n", avctx->channels, channels);

  2023.         avctx->channels = channels;

  2024.     }

  2025. 

  2026.     /* get output buffer */

  2027.     s->frame.nb_samples = 256 * (s->sample_blocks / 8);

  2028.     if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {

  2029.         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");

  2030.         return ret;

  2031.     }

  2032.     samples_flt = (float *)   s->frame.data[0];

  2033.     samples_s16 = (int16_t *) s->frame.data[0];

  2034. 

  2035.     /* filter to get final output */

  2036.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  2037.         dca_filter_channels(s, i);

  2038. 

  2039.         /* If this was marked as a DTS-ES stream we need to subtract back- */

  2040.         /* channel from SL & SR to remove matrixed back-channel signal */

  2041.         if ((s->source_pcm_res & 1) && s->xch_present) {

  2042.             float *back_chan = s->samples + s->channel_order_tab[s->xch_base_channel]     * 256;

  2043.             float *lt_chan   = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256;

  2044.             float *rt_chan   = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;

  2045.             s->dsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);

  2046.             s->dsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);

  2047.         }

  2048. 

  2049.         if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {

  2050.             s->fmt_conv.float_interleave(samples_flt, s->samples_chanptr, 256,

  2051.                                          channels);

  2052.             samples_flt += 256 * channels;

  2053.         } else {

  2054.             s->fmt_conv.float_to_int16_interleave(samples_s16,

  2055.                                                   s->samples_chanptr, 256,

  2056.                                                   channels);

  2057.             samples_s16 += 256 * channels;

  2058.         }

  2059.     }

  2060. 

  2061.     /* update lfe history */

  2062.     lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);

  2063.     for (i = 0; i < 2 * s->lfe * 4; i++)

  2064.         s->lfe_data[i] = s->lfe_data[i + lfe_samples];

  2065. 

  2066.     *got_frame_ptr    = 1;

  2067.     *(AVFrame *) data = s->frame;

  2068. 

  2069.     return buf_size;

  2070. }

  2071. 

  2072. 

  2073. 

  2074. /**

  2075.  * DCA initialization

  2076.  *

  2077.  * @param avctx     pointer to the AVCodecContext

  2078.  */

  2079. 

  2080. static av_cold int dca_decode_init(AVCodecContext *avctx)

  2081. {

  2082.     DCAContext *s = avctx->priv_data;

  2083.     int i;

  2084. 

  2085.     s->avctx = avctx;

  2086.     dca_init_vlcs();

  2087. 

  2088.     ff_dsputil_init(&s->dsp, avctx);

  2089.     ff_mdct_init(&s->imdct, 6, 1, 1.0);

  2090.     ff_synth_filter_init(&s->synth);

  2091.     ff_dcadsp_init(&s->dcadsp);

  2092.     ff_fmt_convert_init(&s->fmt_conv, avctx);

  2093. 

  2094.     for (i = 0; i < DCA_PRIM_CHANNELS_MAX + 1; i++)

  2095.         s->samples_chanptr[i] = s->samples + i * 256;

  2096. 

  2097.     if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {

  2098.         avctx->sample_fmt = AV_SAMPLE_FMT_FLT;

  2099.         s->scale_bias     = 1.0 / 32768.0;

  2100.     } else {

  2101.         avctx->sample_fmt = AV_SAMPLE_FMT_S16;

  2102.         s->scale_bias     = 1.0;

  2103.     }

  2104. 

  2105.     /* allow downmixing to stereo */

  2106.     if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&

  2107.         avctx->request_channels == 2) {

  2108.         avctx->channels = avctx->request_channels;

  2109.     }

  2110. 

  2111.     avcodec_get_frame_defaults(&s->frame);

  2112.     avctx->coded_frame = &s->frame;

  2113. 

  2114.     return 0;

  2115. }

  2116. 

  2117. static av_cold int dca_decode_end(AVCodecContext *avctx)

  2118. {

  2119.     DCAContext *s = avctx->priv_data;

  2120.     ff_mdct_end(&s->imdct);

  2121.     return 0;

  2122. }

  2123. 

  2124. static const AVProfile profiles[] = {

  2125.     { FF_PROFILE_DTS,        "DTS"        },

  2126.     { FF_PROFILE_DTS_ES,     "DTS-ES"     },

  2127.     { FF_PROFILE_DTS_96_24,  "DTS 96/24"  },

  2128.     { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },

  2129.     { FF_PROFILE_DTS_HD_MA,  "DTS-HD MA"  },

  2130.     { FF_PROFILE_UNKNOWN },

  2131. };

  2132. 

  2133. AVCodec ff_dca_decoder = {

  2134.     .name            = "dca",

  2135.     .type            = AVMEDIA_TYPE_AUDIO,

  2136.     .id              = CODEC_ID_DTS,

  2137.     .priv_data_size  = sizeof(DCAContext),

  2138.     .init            = dca_decode_init,

  2139.     .decode          = dca_decode_frame,

  2140.     .close           = dca_decode_end,

  2141.     .long_name       = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),

  2142.     .capabilities    = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,

  2143.     .sample_fmts     = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,

  2144.                                                        AV_SAMPLE_FMT_S16,

  2145.                                                        AV_SAMPLE_FMT_NONE },

  2146.     .profiles        = NULL_IF_CONFIG_SMALL(profiles),

  2147. };