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FFmpeg/libavcodec/dcadec.c

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  1. /*

  2.  * DCA compatible decoder

  3.  * Copyright (C) 2004 Gildas Bazin

  4.  * Copyright (C) 2004 Benjamin Zores

  5.  * Copyright (C) 2006 Benjamin Larsson

  6.  * Copyright (C) 2007 Konstantin Shishkov

  7.  *

  8.  * This file is part of Libav.

  9.  *

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

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

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

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

  14.  *

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

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

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

  18.  * Lesser General Public License for more details.

  19.  *

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

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

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

  23.  */

  24. 

  25. #include <math.h>

  26. #include <stddef.h>

  27. #include <stdio.h>

  28. 

  29. #include "libavutil/channel_layout.h"

  30. #include "libavutil/common.h"

  31. #include "libavutil/float_dsp.h"

  32. #include "libavutil/internal.h"

  33. #include "libavutil/intreadwrite.h"

  34. #include "libavutil/mathematics.h"

  35. #include "libavutil/samplefmt.h"

  36. #include "avcodec.h"

  37. #include "fft.h"

  38. #include "get_bits.h"

  39. #include "put_bits.h"

  40. #include "dcadata.h"

  41. #include "dcahuff.h"

  42. #include "dca.h"

  43. #include "mathops.h"

  44. #include "synth_filter.h"

  45. #include "dcadsp.h"

  46. #include "fmtconvert.h"

  47. #include "internal.h"

  48. 

  49. #if ARCH_ARM

  50. #   include "arm/dca.h"

  51. #endif

  52. 

  53. //#define TRACE

  54. 

  55. #define DCA_PRIM_CHANNELS_MAX  (7)

  56. #define DCA_SUBBANDS          (32)

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

  58. #define DCA_SUBSUBFRAMES_MAX   (4)

  59. #define DCA_SUBFRAMES_MAX     (16)

  60. #define DCA_BLOCKS_MAX        (16)

  61. #define DCA_LFE_MAX            (3)

  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.     /* Frame header */

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

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

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

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

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

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

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

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

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

  297. 

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

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

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

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

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

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

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

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

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

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

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

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

  310.     int copy_history;           ///< copy history

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

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

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

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

  315. 

  316.     /* Primary audio coding header */

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

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

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

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

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

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

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

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

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

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

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

  328. 

  329.     /* Primary audio coding side information */

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

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

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

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

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

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

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

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

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

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

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

  341. 

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

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

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

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

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

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

  348. 

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

  350. 

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

  352.     int lfe_scale_factor;

  353. 

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

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

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

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

  358.     int hist_index[DCA_PRIM_CHANNELS_MAX];

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

  360. 

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

  362. 

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

  364.     float *samples_chanptr[DCA_PRIM_CHANNELS_MAX + 1];

  365.     float *extra_channels[DCA_PRIM_CHANNELS_MAX + 1];

  366.     uint8_t *extra_channels_buffer;

  367.     unsigned int extra_channels_buffer_size;

  368. 

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

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

  371. 

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

  373.     GetBitContext gb;

  374.     /* Current position in DCA frame */

  375.     int current_subframe;

  376.     int current_subsubframe;

  377. 

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

  379. 

  380.     /* XCh extension information */

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

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

  383. 

  384.     /* ExSS header parser */

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

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

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

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

  389. 

  390.     int profile;

  391. 

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

  393.     AVFloatDSPContext fdsp;

  394.     FFTContext imdct;

  395.     SynthFilterContext synth;

  396.     DCADSPContext dcadsp;

  397.     FmtConvertContext fmt_conv;

  398. } DCAContext;

  399. 

  400. static const uint16_t dca_vlc_offs[] = {

  401.         0,   512,   640,   768,  1282,  1794,  2436,  3080,  3770,  4454,  5364,

  402.      5372,  5380,  5388,  5392,  5396,  5412,  5420,  5428,  5460,  5492,  5508,

  403.      5572,  5604,  5668,  5796,  5860,  5892,  6412,  6668,  6796,  7308,  7564,

  404.      7820,  8076,  8620,  9132,  9388,  9910, 10166, 10680, 11196, 11726, 12240,

  405.     12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264,

  406.     18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622,

  407. };

  408. 

  409. static av_cold void dca_init_vlcs(void)

  410. {

  411.     static int vlcs_initialized = 0;

  412.     int i, j, c = 14;

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

  414. 

  415.     if (vlcs_initialized)

  416.         return;

  417. 

  418.     dca_bitalloc_index.offset = 1;

  419.     dca_bitalloc_index.wrap = 2;

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

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

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

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

  424.                  bitalloc_12_bits[i], 1, 1,

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

  426.     }

  427.     dca_scalefactor.offset = -64;

  428.     dca_scalefactor.wrap = 2;

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

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

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

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

  433.                  scales_bits[i], 1, 1,

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

  435.     }

  436.     dca_tmode.offset = 0;

  437.     dca_tmode.wrap = 1;

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

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

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

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

  442.                  tmode_bits[i], 1, 1,

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

  444.     }

  445. 

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

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

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

  449.                 break;

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

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

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

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

  454. 

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

  456.                      bitalloc_sizes[i],

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

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

  459.             c++;

  460.         }

  461.     vlcs_initialized = 1;

  462. }

  463. 

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

  465. {

  466.     while (len--)

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

  468. }

  469. 

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

  471. {

  472.     int i, j;

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

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

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

  476. 

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

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

  479. 

  480.     if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)

  481.         s->prim_channels = DCA_PRIM_CHANNELS_MAX;

  482. 

  483. 

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

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

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

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

  488.     }

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

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

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

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

  493.     }

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

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

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

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

  498. 

  499.     /* Get codebooks quantization indexes */

  500.     if (!base_channel)

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

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

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

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

  505. 

  506.     /* Get scale factor adjustment */

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

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

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

  510. 

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

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

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

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

  515. 

  516.     if (s->crc_present) {

  517.         /* Audio header CRC check */

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

  519.     }

  520. 

  521.     s->current_subframe    = 0;

  522.     s->current_subsubframe = 0;

  523. 

  524. #ifdef TRACE

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

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

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

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

  529.                s->subband_activity[i]);

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

  531.                s->vq_start_subband[i]);

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

  533.                s->joint_intensity[i]);

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

  535.                s->transient_huffman[i]);

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

  537.                s->scalefactor_huffman[i]);

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

  539.                s->bitalloc_huffman[i]);

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

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

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

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

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

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

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

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

  548.     }

  549. #endif

  550. 

  551.     return 0;

  552. }

  553. 

  554. static int dca_parse_frame_header(DCAContext *s)

  555. {

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

  557. 

  558.     /* Sync code */

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

  560. 

  561.     /* Frame header */

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

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

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

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

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

  567.     if (s->frame_size < 95)

  568.         return AVERROR_INVALIDDATA;

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

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

  571.     if (!s->sample_rate)

  572.         return AVERROR_INVALIDDATA;

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

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

  575.     if (!s->bit_rate)

  576.         return AVERROR_INVALIDDATA;

  577. 

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

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

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

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

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

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

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

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

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

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

  588. 

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

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

  591.         return AVERROR_INVALIDDATA;

  592.     }

  593. 

  594.     /* TODO: check CRC */

  595.     if (s->crc_present)

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

  597. 

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

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

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

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

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

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

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

  605. 

  606.     /* FIXME: channels mixing levels */

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

  608.     if (s->lfe)

  609.         s->output |= DCA_LFE;

  610. 

  611. #ifdef TRACE

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

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

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

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

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

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

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

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

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

  621.            s->sample_rate);

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

  623.            s->bit_rate);

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

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

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

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

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

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

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

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

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

  633.            s->predictor_history);

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

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

  636.            s->multirate_inter);

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

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

  639.     av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

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

  646. #endif

  647. 

  648.     /* Primary audio coding header */

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

  650. 

  651.     return dca_parse_audio_coding_header(s, 0);

  652. }

  653. 

  654. 

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

  656. {

  657.     if (level < 5) {

  658.         /* huffman encoded */

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

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

  661.     } else if (level < 8) {

  662.         if (level + 1 > log2range) {

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

  664.             value = get_bits(gb, log2range);

  665.         } else {

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

  667.         }

  668.     }

  669.     return value;

  670. }

  671. 

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

  673. {

  674.     /* Primary audio coding side information */

  675.     int j, k;

  676. 

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

  678.         return AVERROR_INVALIDDATA;

  679. 

  680.     if (!base_channel) {

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

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

  683.     }

  684. 

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

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

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

  688.     }

  689. 

  690.     /* Get prediction codebook */

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

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

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

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

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

  696.             }

  697.         }

  698.     }

  699. 

  700.     /* Bit allocation index */

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

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

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

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

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

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

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

  708.                 av_log(s->avctx, AV_LOG_ERROR,

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

  710.                 return AVERROR_INVALIDDATA;

  711.             } else {

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

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

  714.             }

  715. 

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

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

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

  719.                 return AVERROR_INVALIDDATA;

  720.             }

  721.         }

  722.     }

  723. 

  724.     /* Transition mode */

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

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

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

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

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

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

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

  732.             }

  733.         }

  734.     }

  735. 

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

  737.         return AVERROR_INVALIDDATA;

  738. 

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

  740.         const uint32_t *scale_table;

  741.         int scale_sum, log_size;

  742. 

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

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

  745. 

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

  747.             scale_table = scale_factor_quant7;

  748.             log_size = 7;

  749.         } else {

  750.             scale_table = scale_factor_quant6;

  751.             log_size = 6;

  752.         }

  753. 

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

  755.         scale_sum = 0;

  756. 

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

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

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

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

  761.             }

  762. 

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

  764.                 /* Get second scale factor */

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

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

  767.             }

  768.         }

  769.     }

  770. 

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

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

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

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

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

  776.     }

  777. 

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

  779.         return AVERROR_INVALIDDATA;

  780. 

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

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

  783.         int source_channel;

  784. 

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

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

  787.             int scale = 0;

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

  789. 

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

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

  792. 

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

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

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

  796.             }

  797. 

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

  799.                 av_log(s->avctx, AV_LOG_DEBUG,

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

  801.                 s->debug_flag |= 0x02;

  802.             }

  803.         }

  804.     }

  805. 

  806.     /* Dynamic range coefficient */

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

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

  809. 

  810.     /* Side information CRC check word */

  811.     if (s->crc_present) {

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

  813.     }

  814. 

  815.     /*

  816.      * Primary audio data arrays

  817.      */

  818. 

  819.     /* VQ encoded high frequency subbands */

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

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

  822.             /* 1 vector -> 32 samples */

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

  824. 

  825.     /* Low frequency effect data */

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

  827.         /* LFE samples */

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

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

  830.         float lfe_scale;

  831. 

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

  833.             /* Signed 8 bits int */

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

  835.         }

  836. 

  837.         /* Scale factor index */

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

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

  840. 

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

  842.         lfe_scale = 0.035 * s->lfe_scale_factor;

  843. 

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

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

  846.     }

  847. 

  848. #ifdef TRACE

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

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

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

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

  853. 

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

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

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

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

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

  859.     }

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

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

  862.             av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

  868.     }

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

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

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

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

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

  874.     }

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

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

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

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

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

  880.     }

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

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

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

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

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

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

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

  888.         }

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

  890.     }

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

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

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

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

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

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

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

  898.         }

  899.     }

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

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

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

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

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

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

  906. 

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

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

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

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

  911.     }

  912. #endif

  913. 

  914.     return 0;

  915. }

  916. 

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

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

  919.                             float scale)

  920. {

  921.     const float *prCoeff;

  922. 

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

  924. 

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

  926. 

  927.     /* Select filter */

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

  929.         prCoeff = fir_32bands_nonperfect;

  930.     else                        /* Perfect reconstruction */

  931.         prCoeff = fir_32bands_perfect;

  932. 

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

  934.                               s->subband_fir_hist[chans],

  935.                               &s->hist_index[chans],

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

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

  938. }

  939. 

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

  941.                                   int num_deci_sample, float *samples_in,

  942.                                   float *samples_out, float scale)

  943. {

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

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

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

  947.      *   from last subframe as history.

  948.      *

  949.      * samples_out: An array holding interpolated samples

  950.      */

  951. 

  952.     int decifactor;

  953.     const float *prCoeff;

  954.     int deciindex;

  955. 

  956.     /* Select decimation filter */

  957.     if (decimation_select == 1) {

  958.         decifactor = 64;

  959.         prCoeff = lfe_fir_128;

  960.     } else {

  961.         decifactor = 32;

  962.         prCoeff = lfe_fir_64;

  963.     }

  964.     /* Interpolation */

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

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

  967.         samples_in++;

  968.         samples_out += 2 * decifactor;

  969.     }

  970. }

  971. 

  972. /* downmixing routines */

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

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

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

  976. 

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

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

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

  980. 

  981. #define MIX_FRONT3(samples, coef)                                      \

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

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

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

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

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

  987. 

  988. #define DOWNMIX_TO_STEREO(op1, op2)             \

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

  990.         op1                                     \

  991.         op2                                     \

  992.     }

  993. 

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

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

  996.                         const int8_t *channel_mapping)

  997. {

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

  999.     int i;

  1000.     float t, u, v;

  1001. 

  1002.     switch (srcfmt) {

  1003.     case DCA_MONO:

  1004.     case DCA_CHANNEL:

  1005.     case DCA_STEREO_TOTAL:

  1006.     case DCA_STEREO_SUMDIFF:

  1007.     case DCA_4F2R:

  1008.         av_log(NULL, 0, "Not implemented!\n");

  1009.         break;

  1010.     case DCA_STEREO:

  1011.         break;

  1012.     case DCA_3F:

  1013.         c = channel_mapping[0];

  1014.         l = channel_mapping[1];

  1015.         r = channel_mapping[2];

  1016.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );

  1017.         break;

  1018.     case DCA_2F1R:

  1019.         s = channel_mapping[2];

  1020.         DOWNMIX_TO_STEREO(MIX_REAR1(samples, s, 2, coef), );

  1021.         break;

  1022.     case DCA_3F1R:

  1023.         c = channel_mapping[0];

  1024.         l = channel_mapping[1];

  1025.         r = channel_mapping[2];

  1026.         s = channel_mapping[3];

  1027.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1028.                           MIX_REAR1(samples, s, 3, coef));

  1029.         break;

  1030.     case DCA_2F2R:

  1031.         sl = channel_mapping[2];

  1032.         sr = channel_mapping[3];

  1033.         DOWNMIX_TO_STEREO(MIX_REAR2(samples, sl, sr, 2, coef), );

  1034.         break;

  1035.     case DCA_3F2R:

  1036.         c  = channel_mapping[0];

  1037.         l  = channel_mapping[1];

  1038.         r  = channel_mapping[2];

  1039.         sl = channel_mapping[3];

  1040.         sr = channel_mapping[4];

  1041.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1042.                           MIX_REAR2(samples, sl, sr, 3, coef));

  1043.         break;

  1044.     }

  1045.     if (lfe_present) {

  1046.         int lf_buf = dca_lfe_index[srcfmt];

  1047.         int lf_idx = dca_channels [srcfmt];

  1048.         for (i = 0; i < 256; i++) {

  1049.             samples[0][i] += samples[lf_buf][i] * coef[lf_idx][0];

  1050.             samples[1][i] += samples[lf_buf][i] * coef[lf_idx][1];

  1051.         }

  1052.     }

  1053. }

  1054. 

  1055. 

  1056. #ifndef decode_blockcodes

  1057. /* Very compact version of the block code decoder that does not use table

  1058.  * look-up but is slightly slower */

  1059. static int decode_blockcode(int code, int levels, int32_t *values)

  1060. {

  1061.     int i;

  1062.     int offset = (levels - 1) >> 1;

  1063. 

  1064.     for (i = 0; i < 4; i++) {

  1065.         int div = FASTDIV(code, levels);

  1066.         values[i] = code - offset - div * levels;

  1067.         code = div;

  1068.     }

  1069. 

  1070.     return code;

  1071. }

  1072. 

  1073. static int decode_blockcodes(int code1, int code2, int levels, int32_t *values)

  1074. {

  1075.     return decode_blockcode(code1, levels, values) |

  1076.            decode_blockcode(code2, levels, values + 4);

  1077. }

  1078. #endif

  1079. 

  1080. static const uint8_t abits_sizes[7]  = { 7, 10, 12, 13, 15, 17, 19 };

  1081. static const uint8_t abits_levels[7] = { 3,  5,  7,  9, 13, 17, 25 };

  1082. 

  1083. #ifndef int8x8_fmul_int32

  1084. static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)

  1085. {

  1086.     float fscale = scale / 16.0;

  1087.     int i;

  1088.     for (i = 0; i < 8; i++)

  1089.         dst[i] = src[i] * fscale;

  1090. }

  1091. #endif

  1092. 

  1093. static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)

  1094. {

  1095.     int k, l;

  1096.     int subsubframe = s->current_subsubframe;

  1097. 

  1098.     const float *quant_step_table;

  1099. 

  1100.     /* FIXME */

  1101.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1102.     LOCAL_ALIGNED_16(int32_t, block, [8 * DCA_SUBBANDS]);

  1103. 

  1104.     /*

  1105.      * Audio data

  1106.      */

  1107. 

  1108.     /* Select quantization step size table */

  1109.     if (s->bit_rate_index == 0x1f)

  1110.         quant_step_table = lossless_quant_d;

  1111.     else

  1112.         quant_step_table = lossy_quant_d;

  1113. 

  1114.     for (k = base_channel; k < s->prim_channels; k++) {

  1115.         float rscale[DCA_SUBBANDS];

  1116. 

  1117.         if (get_bits_left(&s->gb) < 0)

  1118.             return AVERROR_INVALIDDATA;

  1119. 

  1120.         for (l = 0; l < s->vq_start_subband[k]; l++) {

  1121.             int m;

  1122. 

  1123.             /* Select the mid-tread linear quantizer */

  1124.             int abits = s->bitalloc[k][l];

  1125. 

  1126.             float quant_step_size = quant_step_table[abits];

  1127. 

  1128.             /*

  1129.              * Determine quantization index code book and its type

  1130.              */

  1131. 

  1132.             /* Select quantization index code book */

  1133.             int sel = s->quant_index_huffman[k][abits];

  1134. 

  1135.             /*

  1136.              * Extract bits from the bit stream

  1137.              */

  1138.             if (!abits) {

  1139.                 rscale[l] = 0;

  1140.                 memset(block + 8 * l, 0, 8 * sizeof(block[0]));

  1141.             } else {

  1142.                 /* Deal with transients */

  1143.                 int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];

  1144.                 rscale[l] = quant_step_size * s->scale_factor[k][l][sfi] *

  1145.                                s->scalefactor_adj[k][sel];

  1146. 

  1147.                 if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {

  1148.                     if (abits <= 7) {

  1149.                         /* Block code */

  1150.                         int block_code1, block_code2, size, levels, err;

  1151. 

  1152.                         size   = abits_sizes[abits - 1];

  1153.                         levels = abits_levels[abits - 1];

  1154. 

  1155.                         block_code1 = get_bits(&s->gb, size);

  1156.                         block_code2 = get_bits(&s->gb, size);

  1157.                         err = decode_blockcodes(block_code1, block_code2,

  1158.                                                 levels, block + 8 * l);

  1159.                         if (err) {

  1160.                             av_log(s->avctx, AV_LOG_ERROR,

  1161.                                    "ERROR: block code look-up failed\n");

  1162.                             return AVERROR_INVALIDDATA;

  1163.                         }

  1164.                     } else {

  1165.                         /* no coding */

  1166.                         for (m = 0; m < 8; m++)

  1167.                             block[8 * l + m] = get_sbits(&s->gb, abits - 3);

  1168.                     }

  1169.                 } else {

  1170.                     /* Huffman coded */

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

  1172.                         block[8 * l + m] = get_bitalloc(&s->gb,

  1173.                                                 &dca_smpl_bitalloc[abits], sel);

  1174.                 }

  1175. 

  1176.             }

  1177.         }

  1178. 

  1179.         s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[k][0],

  1180.                                                block, rscale, 8 * s->vq_start_subband[k]);

  1181. 

  1182.         for (l = 0; l < s->vq_start_subband[k]; l++) {

  1183.             int m;

  1184.             /*

  1185.              * Inverse ADPCM if in prediction mode

  1186.              */

  1187.             if (s->prediction_mode[k][l]) {

  1188.                 int n;

  1189.                 for (m = 0; m < 8; m++) {

  1190.                     for (n = 1; n <= 4; n++)

  1191.                         if (m >= n)

  1192.                             subband_samples[k][l][m] +=

  1193.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1194.                                  subband_samples[k][l][m - n] / 8192);

  1195.                         else if (s->predictor_history)

  1196.                             subband_samples[k][l][m] +=

  1197.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1198.                                  s->subband_samples_hist[k][l][m - n + 4] / 8192);

  1199.                 }

  1200.             }

  1201.         }

  1202. 

  1203.         /*

  1204.          * Decode VQ encoded high frequencies

  1205.          */

  1206.         for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {

  1207.             /* 1 vector -> 32 samples but we only need the 8 samples

  1208.              * for this subsubframe. */

  1209.             int hfvq = s->high_freq_vq[k][l];

  1210. 

  1211.             if (!s->debug_flag & 0x01) {

  1212.                 av_log(s->avctx, AV_LOG_DEBUG,

  1213.                        "Stream with high frequencies VQ coding\n");

  1214.                 s->debug_flag |= 0x01;

  1215.             }

  1216. 

  1217.             int8x8_fmul_int32(subband_samples[k][l],

  1218.                               &high_freq_vq[hfvq][subsubframe * 8],

  1219.                               s->scale_factor[k][l][0]);

  1220.         }

  1221.     }

  1222. 

  1223.     /* Check for DSYNC after subsubframe */

  1224.     if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {

  1225.         if (0xFFFF == get_bits(&s->gb, 16)) {   /* 0xFFFF */

  1226. #ifdef TRACE

  1227.             av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");

  1228. #endif

  1229.         } else {

  1230.             av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");

  1231.             return AVERROR_INVALIDDATA;

  1232.         }

  1233.     }

  1234. 

  1235.     /* Backup predictor history for adpcm */

  1236.     for (k = base_channel; k < s->prim_channels; k++)

  1237.         for (l = 0; l < s->vq_start_subband[k]; l++)

  1238.             memcpy(s->subband_samples_hist[k][l],

  1239.                    &subband_samples[k][l][4],

  1240.                    4 * sizeof(subband_samples[0][0][0]));

  1241. 

  1242.     return 0;

  1243. }

  1244. 

  1245. static int dca_filter_channels(DCAContext *s, int block_index)

  1246. {

  1247.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1248.     int k;

  1249. 

  1250.     /* 32 subbands QMF */

  1251.     for (k = 0; k < s->prim_channels; k++) {

  1252. /*        static float pcm_to_double[8] = { 32768.0, 32768.0, 524288.0, 524288.0,

  1253.                                             0, 8388608.0, 8388608.0 };*/

  1254.         if (s->channel_order_tab[k] >= 0)

  1255.             qmf_32_subbands(s, k, subband_samples[k],

  1256.                             s->samples_chanptr[s->channel_order_tab[k]],

  1257.                             M_SQRT1_2 / 32768.0 /* pcm_to_double[s->source_pcm_res] */);

  1258.     }

  1259. 

  1260.     /* Generate LFE samples for this subsubframe FIXME!!! */

  1261.     if (s->output & DCA_LFE) {

  1262.         lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,

  1263.                               s->lfe_data + 2 * s->lfe * (block_index + 4),

  1264.                               s->samples_chanptr[dca_lfe_index[s->amode]],

  1265.                               1.0 / (256.0 * 32768.0));

  1266.         /* Outputs 20bits pcm samples */

  1267.     }

  1268. 

  1269.     /* Downmixing to Stereo */

  1270.     if (s->prim_channels + !!s->lfe > 2 &&

  1271.         s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {

  1272.         dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef,

  1273.                     s->channel_order_tab);

  1274.     }

  1275. 

  1276.     return 0;

  1277. }

  1278. 

  1279. 

  1280. static int dca_subframe_footer(DCAContext *s, int base_channel)

  1281. {

  1282.     int in, out, aux_data_count, aux_data_end, reserved;

  1283. 

  1284.     /*

  1285.      * Unpack optional information

  1286.      */

  1287. 

  1288.     /* presumably optional information only appears in the core? */

  1289.     if (!base_channel) {

  1290.         if (s->timestamp)

  1291.             skip_bits_long(&s->gb, 32);

  1292. 

  1293.         if (s->aux_data) {

  1294.             aux_data_count = get_bits(&s->gb, 6);

  1295. 

  1296.             // align (32-bit)

  1297.             skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1298. 

  1299.             aux_data_end = 8 * aux_data_count + get_bits_count(&s->gb);

  1300. 

  1301.             if (get_bits_long(&s->gb, 32) != 0x9A1105A0) // nSYNCAUX

  1302.                 return AVERROR_INVALIDDATA;

  1303. 

  1304.             if (get_bits1(&s->gb)) { // bAUXTimeStampFlag

  1305.                 avpriv_request_sample(s->avctx,

  1306.                                       "Auxiliary Decode Time Stamp Flag");

  1307.                 // align (4-bit)

  1308.                 skip_bits(&s->gb, (-get_bits_count(&s->gb)) & 4);

  1309.                 // 44 bits: nMSByte (8), nMarker (4), nLSByte (28), nMarker (4)

  1310.                 skip_bits_long(&s->gb, 44);

  1311.             }

  1312. 

  1313.             if ((s->core_downmix = get_bits1(&s->gb))) {

  1314.                 switch (get_bits(&s->gb, 3)) {

  1315.                 case 0:

  1316.                     s->core_downmix_amode = DCA_MONO;

  1317.                     break;

  1318.                 case 1:

  1319.                     s->core_downmix_amode = DCA_STEREO;

  1320.                     break;

  1321.                 case 2:

  1322.                     s->core_downmix_amode = DCA_STEREO_TOTAL;

  1323.                     break;

  1324.                 case 3:

  1325.                     s->core_downmix_amode = DCA_3F;

  1326.                     break;

  1327.                 case 4:

  1328.                     s->core_downmix_amode = DCA_2F1R;

  1329.                     break;

  1330.                 case 5:

  1331.                     s->core_downmix_amode = DCA_2F2R;

  1332.                     break;

  1333.                 case 6:

  1334.                     s->core_downmix_amode = DCA_3F1R;

  1335.                     break;

  1336.                 default:

  1337.                     return AVERROR_INVALIDDATA;

  1338.                 }

  1339.                 for (out = 0; out < dca_channels[s->core_downmix_amode]; out++) {

  1340.                     for (in = 0; in < s->prim_channels + !!s->lfe; in++) {

  1341.                         uint16_t tmp = get_bits(&s->gb, 9);

  1342.                         if ((tmp & 0xFF) > 241)

  1343.                             return AVERROR_INVALIDDATA;

  1344.                         s->core_downmix_codes[in][out] = tmp;

  1345.                     }

  1346.                 }

  1347.             }

  1348. 

  1349.             align_get_bits(&s->gb); // byte align

  1350.             skip_bits(&s->gb, 16);  // nAUXCRC16

  1351. 

  1352.             // additional data (reserved, cf. ETSI TS 102 114 V1.4.1)

  1353.             if ((reserved = (aux_data_end - get_bits_count(&s->gb))) < 0)

  1354.                 return AVERROR_INVALIDDATA;

  1355.             else if (reserved) {

  1356.                 avpriv_request_sample(s->avctx,

  1357.                                       "Core auxiliary data reserved content");

  1358.                 skip_bits_long(&s->gb, reserved);

  1359.             }

  1360.         }

  1361. 

  1362.         if (s->crc_present && s->dynrange)

  1363.             get_bits(&s->gb, 16);

  1364.     }

  1365. 

  1366.     return 0;

  1367. }

  1368. 

  1369. /**

  1370.  * Decode a dca frame block

  1371.  *

  1372.  * @param s     pointer to the DCAContext

  1373.  */

  1374. 

  1375. static int dca_decode_block(DCAContext *s, int base_channel, int block_index)

  1376. {

  1377.     int ret;

  1378. 

  1379.     /* Sanity check */

  1380.     if (s->current_subframe >= s->subframes) {

  1381.         av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",

  1382.                s->current_subframe, s->subframes);

  1383.         return AVERROR_INVALIDDATA;

  1384.     }

  1385. 

  1386.     if (!s->current_subsubframe) {

  1387. #ifdef TRACE

  1388.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");

  1389. #endif

  1390.         /* Read subframe header */

  1391.         if ((ret = dca_subframe_header(s, base_channel, block_index)))

  1392.             return ret;

  1393.     }

  1394. 

  1395.     /* Read subsubframe */

  1396. #ifdef TRACE

  1397.     av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");

  1398. #endif

  1399.     if ((ret = dca_subsubframe(s, base_channel, block_index)))

  1400.         return ret;

  1401. 

  1402.     /* Update state */

  1403.     s->current_subsubframe++;

  1404.     if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {

  1405.         s->current_subsubframe = 0;

  1406.         s->current_subframe++;

  1407.     }

  1408.     if (s->current_subframe >= s->subframes) {

  1409. #ifdef TRACE

  1410.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");

  1411. #endif

  1412.         /* Read subframe footer */

  1413.         if ((ret = dca_subframe_footer(s, base_channel)))

  1414.             return ret;

  1415.     }

  1416. 

  1417.     return 0;

  1418. }

  1419. 

  1420. /**

  1421.  * Return the number of channels in an ExSS speaker mask (HD)

  1422.  */

  1423. static int dca_exss_mask2count(int mask)

  1424. {

  1425.     /* count bits that mean speaker pairs twice */

  1426.     return av_popcount(mask) +

  1427.            av_popcount(mask & (DCA_EXSS_CENTER_LEFT_RIGHT      |

  1428.                                DCA_EXSS_FRONT_LEFT_RIGHT       |

  1429.                                DCA_EXSS_FRONT_HIGH_LEFT_RIGHT  |

  1430.                                DCA_EXSS_WIDE_LEFT_RIGHT        |

  1431.                                DCA_EXSS_SIDE_LEFT_RIGHT        |

  1432.                                DCA_EXSS_SIDE_HIGH_LEFT_RIGHT   |

  1433.                                DCA_EXSS_SIDE_REAR_LEFT_RIGHT   |

  1434.                                DCA_EXSS_REAR_LEFT_RIGHT        |

  1435.                                DCA_EXSS_REAR_HIGH_LEFT_RIGHT));

  1436. }

  1437. 

  1438. /**

  1439.  * Skip mixing coefficients of a single mix out configuration (HD)

  1440.  */

  1441. static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch)

  1442. {

  1443.     int i;

  1444. 

  1445.     for (i = 0; i < channels; i++) {

  1446.         int mix_map_mask = get_bits(gb, out_ch);

  1447.         int num_coeffs = av_popcount(mix_map_mask);

  1448.         skip_bits_long(gb, num_coeffs * 6);

  1449.     }

  1450. }

  1451. 

  1452. /**

  1453.  * Parse extension substream asset header (HD)

  1454.  */

  1455. static int dca_exss_parse_asset_header(DCAContext *s)

  1456. {

  1457.     int header_pos = get_bits_count(&s->gb);

  1458.     int header_size;

  1459.     int channels;

  1460.     int embedded_stereo = 0;

  1461.     int embedded_6ch    = 0;

  1462.     int drc_code_present;

  1463.     int extensions_mask;

  1464.     int i, j;

  1465. 

  1466.     if (get_bits_left(&s->gb) < 16)

  1467.         return -1;

  1468. 

  1469.     /* We will parse just enough to get to the extensions bitmask with which

  1470.      * we can set the profile value. */

  1471. 

  1472.     header_size = get_bits(&s->gb, 9) + 1;

  1473.     skip_bits(&s->gb, 3); // asset index

  1474. 

  1475.     if (s->static_fields) {

  1476.         if (get_bits1(&s->gb))

  1477.             skip_bits(&s->gb, 4); // asset type descriptor

  1478.         if (get_bits1(&s->gb))

  1479.             skip_bits_long(&s->gb, 24); // language descriptor

  1480. 

  1481.         if (get_bits1(&s->gb)) {

  1482.             /* How can one fit 1024 bytes of text here if the maximum value

  1483.              * for the asset header size field above was 512 bytes? */

  1484.             int text_length = get_bits(&s->gb, 10) + 1;

  1485.             if (get_bits_left(&s->gb) < text_length * 8)

  1486.                 return -1;

  1487.             skip_bits_long(&s->gb, text_length * 8); // info text

  1488.         }

  1489. 

  1490.         skip_bits(&s->gb, 5); // bit resolution - 1

  1491.         skip_bits(&s->gb, 4); // max sample rate code

  1492.         channels = get_bits(&s->gb, 8) + 1;

  1493. 

  1494.         if (get_bits1(&s->gb)) { // 1-to-1 channels to speakers

  1495.             int spkr_remap_sets;

  1496.             int spkr_mask_size = 16;

  1497.             int num_spkrs[7];

  1498. 

  1499.             if (channels > 2)

  1500.                 embedded_stereo = get_bits1(&s->gb);

  1501.             if (channels > 6)

  1502.                 embedded_6ch = get_bits1(&s->gb);

  1503. 

  1504.             if (get_bits1(&s->gb)) {

  1505.                 spkr_mask_size = (get_bits(&s->gb, 2) + 1) << 2;

  1506.                 skip_bits(&s->gb, spkr_mask_size); // spkr activity mask

  1507.             }

  1508. 

  1509.             spkr_remap_sets = get_bits(&s->gb, 3);

  1510. 

  1511.             for (i = 0; i < spkr_remap_sets; i++) {

  1512.                 /* std layout mask for each remap set */

  1513.                 num_spkrs[i] = dca_exss_mask2count(get_bits(&s->gb, spkr_mask_size));

  1514.             }

  1515. 

  1516.             for (i = 0; i < spkr_remap_sets; i++) {

  1517.                 int num_dec_ch_remaps = get_bits(&s->gb, 5) + 1;

  1518.                 if (get_bits_left(&s->gb) < 0)

  1519.                     return -1;

  1520. 

  1521.                 for (j = 0; j < num_spkrs[i]; j++) {

  1522.                     int remap_dec_ch_mask = get_bits_long(&s->gb, num_dec_ch_remaps);

  1523.                     int num_dec_ch = av_popcount(remap_dec_ch_mask);

  1524.                     skip_bits_long(&s->gb, num_dec_ch * 5); // remap codes

  1525.                 }

  1526.             }

  1527. 

  1528.         } else {

  1529.             skip_bits(&s->gb, 3); // representation type

  1530.         }

  1531.     }

  1532. 

  1533.     drc_code_present = get_bits1(&s->gb);

  1534.     if (drc_code_present)

  1535.         get_bits(&s->gb, 8); // drc code

  1536. 

  1537.     if (get_bits1(&s->gb))

  1538.         skip_bits(&s->gb, 5); // dialog normalization code

  1539. 

  1540.     if (drc_code_present && embedded_stereo)

  1541.         get_bits(&s->gb, 8); // drc stereo code

  1542. 

  1543.     if (s->mix_metadata && get_bits1(&s->gb)) {

  1544.         skip_bits(&s->gb, 1); // external mix

  1545.         skip_bits(&s->gb, 6); // post mix gain code

  1546. 

  1547.         if (get_bits(&s->gb, 2) != 3) // mixer drc code

  1548.             skip_bits(&s->gb, 3); // drc limit

  1549.         else

  1550.             skip_bits(&s->gb, 8); // custom drc code

  1551. 

  1552.         if (get_bits1(&s->gb)) // channel specific scaling

  1553.             for (i = 0; i < s->num_mix_configs; i++)

  1554.                 skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6); // scale codes

  1555.         else

  1556.             skip_bits_long(&s->gb, s->num_mix_configs * 6); // scale codes

  1557. 

  1558.         for (i = 0; i < s->num_mix_configs; i++) {

  1559.             if (get_bits_left(&s->gb) < 0)

  1560.                 return -1;

  1561.             dca_exss_skip_mix_coeffs(&s->gb, channels, s->mix_config_num_ch[i]);

  1562.             if (embedded_6ch)

  1563.                 dca_exss_skip_mix_coeffs(&s->gb, 6, s->mix_config_num_ch[i]);

  1564.             if (embedded_stereo)

  1565.                 dca_exss_skip_mix_coeffs(&s->gb, 2, s->mix_config_num_ch[i]);

  1566.         }

  1567.     }

  1568. 

  1569.     switch (get_bits(&s->gb, 2)) {

  1570.     case 0: extensions_mask = get_bits(&s->gb, 12); break;

  1571.     case 1: extensions_mask = DCA_EXT_EXSS_XLL;     break;

  1572.     case 2: extensions_mask = DCA_EXT_EXSS_LBR;     break;

  1573.     case 3: extensions_mask = 0; /* aux coding */   break;

  1574.     }

  1575. 

  1576.     /* not parsed further, we were only interested in the extensions mask */

  1577. 

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

  1579.         return -1;

  1580. 

  1581.     if (get_bits_count(&s->gb) - header_pos > header_size * 8) {

  1582.         av_log(s->avctx, AV_LOG_WARNING, "Asset header size mismatch.\n");

  1583.         return -1;

  1584.     }

  1585.     skip_bits_long(&s->gb, header_pos + header_size * 8 - get_bits_count(&s->gb));

  1586. 

  1587.     if (extensions_mask & DCA_EXT_EXSS_XLL)

  1588.         s->profile = FF_PROFILE_DTS_HD_MA;

  1589.     else if (extensions_mask & (DCA_EXT_EXSS_XBR | DCA_EXT_EXSS_X96 |

  1590.                                 DCA_EXT_EXSS_XXCH))

  1591.         s->profile = FF_PROFILE_DTS_HD_HRA;

  1592. 

  1593.     if (!(extensions_mask & DCA_EXT_CORE))

  1594.         av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n");

  1595.     if ((extensions_mask & DCA_CORE_EXTS) != s->core_ext_mask)

  1596.         av_log(s->avctx, AV_LOG_WARNING,

  1597.                "DTS extensions detection mismatch (%d, %d)\n",

  1598.                extensions_mask & DCA_CORE_EXTS, s->core_ext_mask);

  1599. 

  1600.     return 0;

  1601. }

  1602. 

  1603. /**

  1604.  * Parse extension substream header (HD)

  1605.  */

  1606. static void dca_exss_parse_header(DCAContext *s)

  1607. {

  1608.     int ss_index;

  1609.     int blownup;

  1610.     int num_audiop = 1;

  1611.     int num_assets = 1;

  1612.     int active_ss_mask[8];

  1613.     int i, j;

  1614. 

  1615.     if (get_bits_left(&s->gb) < 52)

  1616.         return;

  1617. 

  1618.     skip_bits(&s->gb, 8); // user data

  1619.     ss_index = get_bits(&s->gb, 2);

  1620. 

  1621.     blownup = get_bits1(&s->gb);

  1622.     skip_bits(&s->gb,  8 + 4 * blownup); // header_size

  1623.     skip_bits(&s->gb, 16 + 4 * blownup); // hd_size

  1624. 

  1625.     s->static_fields = get_bits1(&s->gb);

  1626.     if (s->static_fields) {

  1627.         skip_bits(&s->gb, 2); // reference clock code

  1628.         skip_bits(&s->gb, 3); // frame duration code

  1629. 

  1630.         if (get_bits1(&s->gb))

  1631.             skip_bits_long(&s->gb, 36); // timestamp

  1632. 

  1633.         /* a single stream can contain multiple audio assets that can be

  1634.          * combined to form multiple audio presentations */

  1635. 

  1636.         num_audiop = get_bits(&s->gb, 3) + 1;

  1637.         if (num_audiop > 1) {

  1638.             avpriv_request_sample(s->avctx,

  1639.                                   "Multiple DTS-HD audio presentations");

  1640.             /* ignore such streams for now */

  1641.             return;

  1642.         }

  1643. 

  1644.         num_assets = get_bits(&s->gb, 3) + 1;

  1645.         if (num_assets > 1) {

  1646.             avpriv_request_sample(s->avctx, "Multiple DTS-HD audio assets");

  1647.             /* ignore such streams for now */

  1648.             return;

  1649.         }

  1650. 

  1651.         for (i = 0; i < num_audiop; i++)

  1652.             active_ss_mask[i] = get_bits(&s->gb, ss_index + 1);

  1653. 

  1654.         for (i = 0; i < num_audiop; i++)

  1655.             for (j = 0; j <= ss_index; j++)

  1656.                 if (active_ss_mask[i] & (1 << j))

  1657.                     skip_bits(&s->gb, 8); // active asset mask

  1658. 

  1659.         s->mix_metadata = get_bits1(&s->gb);

  1660.         if (s->mix_metadata) {

  1661.             int mix_out_mask_size;

  1662. 

  1663.             skip_bits(&s->gb, 2); // adjustment level

  1664.             mix_out_mask_size  = (get_bits(&s->gb, 2) + 1) << 2;

  1665.             s->num_mix_configs =  get_bits(&s->gb, 2) + 1;

  1666. 

  1667.             for (i = 0; i < s->num_mix_configs; i++) {

  1668.                 int mix_out_mask        = get_bits(&s->gb, mix_out_mask_size);

  1669.                 s->mix_config_num_ch[i] = dca_exss_mask2count(mix_out_mask);

  1670.             }

  1671.         }

  1672.     }

  1673. 

  1674.     for (i = 0; i < num_assets; i++)

  1675.         skip_bits_long(&s->gb, 16 + 4 * blownup);  // asset size

  1676. 

  1677.     for (i = 0; i < num_assets; i++) {

  1678.         if (dca_exss_parse_asset_header(s))

  1679.             return;

  1680.     }

  1681. 

  1682.     /* not parsed further, we were only interested in the extensions mask

  1683.      * from the asset header */

  1684. }

  1685. 

  1686. /**

  1687.  * Main frame decoding function

  1688.  * FIXME add arguments

  1689.  */

  1690. static int dca_decode_frame(AVCodecContext *avctx, void *data,

  1691.                             int *got_frame_ptr, AVPacket *avpkt)

  1692. {

  1693.     AVFrame *frame     = data;

  1694.     const uint8_t *buf = avpkt->data;

  1695.     int buf_size = avpkt->size;

  1696. 

  1697.     int lfe_samples;

  1698.     int num_core_channels = 0;

  1699.     int i, ret;

  1700.     float  **samples_flt;

  1701.     DCAContext *s = avctx->priv_data;

  1702.     int channels, full_channels;

  1703.     int core_ss_end;

  1704. 

  1705. 

  1706.     s->xch_present = 0;

  1707. 

  1708.     s->dca_buffer_size = ff_dca_convert_bitstream(buf, buf_size, s->dca_buffer,

  1709.                                                   DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);

  1710.     if (s->dca_buffer_size == AVERROR_INVALIDDATA) {

  1711.         av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");

  1712.         return AVERROR_INVALIDDATA;

  1713.     }

  1714. 

  1715.     init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);

  1716.     if ((ret = dca_parse_frame_header(s)) < 0) {

  1717.         //seems like the frame is corrupt, try with the next one

  1718.         return ret;

  1719.     }

  1720.     //set AVCodec values with parsed data

  1721.     avctx->sample_rate = s->sample_rate;

  1722.     avctx->bit_rate    = s->bit_rate;

  1723. 

  1724.     s->profile = FF_PROFILE_DTS;

  1725. 

  1726.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1727.         if ((ret = dca_decode_block(s, 0, i))) {

  1728.             av_log(avctx, AV_LOG_ERROR, "error decoding block\n");

  1729.             return ret;

  1730.         }

  1731.     }

  1732. 

  1733.     /* record number of core channels incase less than max channels are requested */

  1734.     num_core_channels = s->prim_channels;

  1735. 

  1736.     if (s->ext_coding)

  1737.         s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];

  1738.     else

  1739.         s->core_ext_mask = 0;

  1740. 

  1741.     core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;

  1742. 

  1743.     /* only scan for extensions if ext_descr was unknown or indicated a

  1744.      * supported XCh extension */

  1745.     if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {

  1746. 

  1747.         /* if ext_descr was unknown, clear s->core_ext_mask so that the

  1748.          * extensions scan can fill it up */

  1749.         s->core_ext_mask = FFMAX(s->core_ext_mask, 0);

  1750. 

  1751.         /* extensions start at 32-bit boundaries into bitstream */

  1752.         skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1753. 

  1754.         while (core_ss_end - get_bits_count(&s->gb) >= 32) {

  1755.             uint32_t bits = get_bits_long(&s->gb, 32);

  1756. 

  1757.             switch (bits) {

  1758.             case 0x5a5a5a5a: {

  1759.                 int ext_amode, xch_fsize;

  1760. 

  1761.                 s->xch_base_channel = s->prim_channels;

  1762. 

  1763.                 /* validate sync word using XCHFSIZE field */

  1764.                 xch_fsize = show_bits(&s->gb, 10);

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

  1766.                     (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))

  1767.                     continue;

  1768. 

  1769.                 /* skip length-to-end-of-frame field for the moment */

  1770.                 skip_bits(&s->gb, 10);

  1771. 

  1772.                 s->core_ext_mask |= DCA_EXT_XCH;

  1773. 

  1774.                 /* extension amode(number of channels in extension) should be 1 */

  1775.                 /* AFAIK XCh is not used for more channels */

  1776.                 if ((ext_amode = get_bits(&s->gb, 4)) != 1) {

  1777.                     av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"

  1778.                            " supported!\n", ext_amode);

  1779.                     continue;

  1780.                 }

  1781. 

  1782.                 /* much like core primary audio coding header */

  1783.                 dca_parse_audio_coding_header(s, s->xch_base_channel);

  1784. 

  1785.                 for (i = 0; i < (s->sample_blocks / 8); i++)

  1786.                     if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {

  1787.                         av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");

  1788.                         continue;

  1789.                     }

  1790. 

  1791.                 s->xch_present = 1;

  1792.                 break;

  1793.             }

  1794.             case 0x47004a03:

  1795.                 /* XXCh: extended channels */

  1796.                 /* usually found either in core or HD part in DTS-HD HRA streams,

  1797.                  * but not in DTS-ES which contains XCh extensions instead */

  1798.                 s->core_ext_mask |= DCA_EXT_XXCH;

  1799.                 break;

  1800. 

  1801.             case 0x1d95f262: {

  1802.                 int fsize96 = show_bits(&s->gb, 12) + 1;

  1803.                 if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)

  1804.                     continue;

  1805. 

  1806.                 av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n",

  1807.                        get_bits_count(&s->gb));

  1808.                 skip_bits(&s->gb, 12);

  1809.                 av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);

  1810.                 av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));

  1811. 

  1812.                 s->core_ext_mask |= DCA_EXT_X96;

  1813.                 break;

  1814.             }

  1815.             }

  1816. 

  1817.             skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1818.         }

  1819.     } else {

  1820.         /* no supported extensions, skip the rest of the core substream */

  1821.         skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));

  1822.     }

  1823. 

  1824.     if (s->core_ext_mask & DCA_EXT_X96)

  1825.         s->profile = FF_PROFILE_DTS_96_24;

  1826.     else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))

  1827.         s->profile = FF_PROFILE_DTS_ES;

  1828. 

  1829.     /* check for ExSS (HD part) */

  1830.     if (s->dca_buffer_size - s->frame_size > 32 &&

  1831.         get_bits_long(&s->gb, 32) == DCA_HD_MARKER)

  1832.         dca_exss_parse_header(s);

  1833. 

  1834.     avctx->profile = s->profile;

  1835. 

  1836.     full_channels = channels = s->prim_channels + !!s->lfe;

  1837. 

  1838.     if (s->amode < 16) {

  1839.         avctx->channel_layout = dca_core_channel_layout[s->amode];

  1840. 

  1841. #if FF_API_REQUEST_CHANNELS

  1842. FF_DISABLE_DEPRECATION_WARNINGS

  1843.         if (s->xch_present && (!avctx->request_channels ||

  1844.                                avctx->request_channels > num_core_channels + !!s->lfe)) {

  1845. FF_ENABLE_DEPRECATION_WARNINGS

  1846. #else

  1847.         if (s->xch_present) {

  1848. #endif

  1849.             avctx->channel_layout |= AV_CH_BACK_CENTER;

  1850.             if (s->lfe) {

  1851.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1852.                 s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode];

  1853.             } else {

  1854.                 s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];

  1855.             }

  1856.         } else {

  1857.             channels = num_core_channels + !!s->lfe;

  1858.             s->xch_present = 0; /* disable further xch processing */

  1859.             if (s->lfe) {

  1860.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1861.                 s->channel_order_tab = dca_channel_reorder_lfe[s->amode];

  1862.             } else

  1863.                 s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];

  1864.         }

  1865. 

  1866.         if (channels > !!s->lfe &&

  1867.             s->channel_order_tab[channels - 1 - !!s->lfe] < 0)

  1868.             return AVERROR_INVALIDDATA;

  1869. 

  1870.         if (s->prim_channels + !!s->lfe > 2 &&

  1871.             avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {

  1872.             channels = 2;

  1873.             s->output = DCA_STEREO;

  1874.             avctx->channel_layout = AV_CH_LAYOUT_STEREO;

  1875. 

  1876.             /* Stereo downmix coefficients

  1877.              *

  1878.              * The decoder can only downmix to 2-channel, so we need to ensure

  1879.              * embedded downmix coefficients are actually targeting 2-channel.

  1880.              */

  1881.             if (s->core_downmix && (s->core_downmix_amode == DCA_STEREO ||

  1882.                                     s->core_downmix_amode == DCA_STEREO_TOTAL)) {

  1883.                 int sign, code;

  1884.                 for (i = 0; i < s->prim_channels + !!s->lfe; i++) {

  1885.                     sign = s->core_downmix_codes[i][0] & 0x100 ? 1 : -1;

  1886.                     code = s->core_downmix_codes[i][0] & 0x0FF;

  1887.                     s->downmix_coef[i][0] = (!code ? 0.0f :

  1888.                                              sign * dca_dmixtable[code - 1]);

  1889.                     sign = s->core_downmix_codes[i][1] & 0x100 ? 1 : -1;

  1890.                     code = s->core_downmix_codes[i][1] & 0x0FF;

  1891.                     s->downmix_coef[i][1] = (!code ? 0.0f :

  1892.                                              sign * dca_dmixtable[code - 1]);

  1893.                 }

  1894.             } else {

  1895.                 int am = s->amode & DCA_CHANNEL_MASK;

  1896.                 if (am >= FF_ARRAY_ELEMS(dca_default_coeffs)) {

  1897.                     av_log(s->avctx, AV_LOG_ERROR,

  1898.                            "Invalid channel mode %d\n", am);

  1899.                     return AVERROR_INVALIDDATA;

  1900.                 }

  1901.                 if (s->prim_channels + !!s->lfe >

  1902.                     FF_ARRAY_ELEMS(dca_default_coeffs[0])) {

  1903.                     avpriv_request_sample(s->avctx, "Downmixing %d channels",

  1904.                                           s->prim_channels + !!s->lfe);

  1905.                     return AVERROR_PATCHWELCOME;

  1906.                 }

  1907.                 for (i = 0; i < s->prim_channels + !!s->lfe; i++) {

  1908.                     s->downmix_coef[i][0] = dca_default_coeffs[am][i][0];

  1909.                     s->downmix_coef[i][1] = dca_default_coeffs[am][i][1];

  1910.                 }

  1911.             }

  1912.             av_dlog(s->avctx, "Stereo downmix coeffs:\n");

  1913.             for (i = 0; i < s->prim_channels + !!s->lfe; i++) {

  1914.                 av_dlog(s->avctx, "L, input channel %d = %f\n", i,

  1915.                         s->downmix_coef[i][0]);

  1916.                 av_dlog(s->avctx, "R, input channel %d = %f\n", i,

  1917.                         s->downmix_coef[i][1]);

  1918.             }

  1919.             av_dlog(s->avctx, "\n");

  1920.         }

  1921.     } else {

  1922.         av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);

  1923.         return AVERROR_INVALIDDATA;

  1924.     }

  1925.     avctx->channels = channels;

  1926. 

  1927.     /* get output buffer */

  1928.     frame->nb_samples = 256 * (s->sample_blocks / 8);

  1929.     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {

  1930.         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");

  1931.         return ret;

  1932.     }

  1933.     samples_flt = (float **)frame->extended_data;

  1934. 

  1935.     /* allocate buffer for extra channels if downmixing */

  1936.     if (avctx->channels < full_channels) {

  1937.         ret = av_samples_get_buffer_size(NULL, full_channels - channels,

  1938.                                          frame->nb_samples,

  1939.                                          avctx->sample_fmt, 0);

  1940.         if (ret < 0)

  1941.             return ret;

  1942. 

  1943.         av_fast_malloc(&s->extra_channels_buffer,

  1944.                        &s->extra_channels_buffer_size, ret);

  1945.         if (!s->extra_channels_buffer)

  1946.             return AVERROR(ENOMEM);

  1947. 

  1948.         ret = av_samples_fill_arrays((uint8_t **)s->extra_channels, NULL,

  1949.                                      s->extra_channels_buffer,

  1950.                                      full_channels - channels,

  1951.                                      frame->nb_samples, avctx->sample_fmt, 0);

  1952.         if (ret < 0)

  1953.             return ret;

  1954.     }

  1955. 

  1956.     /* filter to get final output */

  1957.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1958.         int ch;

  1959. 

  1960.         for (ch = 0; ch < channels; ch++)

  1961.             s->samples_chanptr[ch] = samples_flt[ch] + i * 256;

  1962.         for (; ch < full_channels; ch++)

  1963.             s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * 256;

  1964. 

  1965.         dca_filter_channels(s, i);

  1966. 

  1967.         /* If this was marked as a DTS-ES stream we need to subtract back- */

  1968.         /* channel from SL & SR to remove matrixed back-channel signal */

  1969.         if ((s->source_pcm_res & 1) && s->xch_present) {

  1970.             float *back_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel]];

  1971.             float *lt_chan   = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 2]];

  1972.             float *rt_chan   = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 1]];

  1973.             s->fdsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);

  1974.             s->fdsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);

  1975.         }

  1976.     }

  1977. 

  1978.     /* update lfe history */

  1979.     lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);

  1980.     for (i = 0; i < 2 * s->lfe * 4; i++)

  1981.         s->lfe_data[i] = s->lfe_data[i + lfe_samples];

  1982. 

  1983.     *got_frame_ptr = 1;

  1984. 

  1985.     return buf_size;

  1986. }

  1987. 

  1988. 

  1989. 

  1990. /**

  1991.  * DCA initialization

  1992.  *

  1993.  * @param avctx     pointer to the AVCodecContext

  1994.  */

  1995. 

  1996. static av_cold int dca_decode_init(AVCodecContext *avctx)

  1997. {

  1998.     DCAContext *s = avctx->priv_data;

  1999. 

  2000.     s->avctx = avctx;

  2001.     dca_init_vlcs();

  2002. 

  2003.     avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);

  2004.     ff_mdct_init(&s->imdct, 6, 1, 1.0);

  2005.     ff_synth_filter_init(&s->synth);

  2006.     ff_dcadsp_init(&s->dcadsp);

  2007.     ff_fmt_convert_init(&s->fmt_conv, avctx);

  2008. 

  2009.     avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

  2010. 

  2011.     /* allow downmixing to stereo */

  2012. #if FF_API_REQUEST_CHANNELS

  2013. FF_DISABLE_DEPRECATION_WARNINGS

  2014.     if (avctx->request_channels == 2)

  2015.         avctx->request_channel_layout = AV_CH_LAYOUT_STEREO;

  2016. FF_ENABLE_DEPRECATION_WARNINGS

  2017. #endif

  2018.     if (avctx->channels > 2 &&

  2019.         avctx->request_channel_layout == AV_CH_LAYOUT_STEREO)

  2020.         avctx->channels = 2;

  2021. 

  2022.     return 0;

  2023. }

  2024. 

  2025. static av_cold int dca_decode_end(AVCodecContext *avctx)

  2026. {

  2027.     DCAContext *s = avctx->priv_data;

  2028.     ff_mdct_end(&s->imdct);

  2029.     av_freep(&s->extra_channels_buffer);

  2030.     return 0;

  2031. }

  2032. 

  2033. static const AVProfile profiles[] = {

  2034.     { FF_PROFILE_DTS,        "DTS"        },

  2035.     { FF_PROFILE_DTS_ES,     "DTS-ES"     },

  2036.     { FF_PROFILE_DTS_96_24,  "DTS 96/24"  },

  2037.     { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },

  2038.     { FF_PROFILE_DTS_HD_MA,  "DTS-HD MA"  },

  2039.     { FF_PROFILE_UNKNOWN },

  2040. };

  2041. 

  2042. AVCodec ff_dca_decoder = {

  2043.     .name            = "dca",

  2044.     .long_name       = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),

  2045.     .type            = AVMEDIA_TYPE_AUDIO,

  2046.     .id              = AV_CODEC_ID_DTS,

  2047.     .priv_data_size  = sizeof(DCAContext),

  2048.     .init            = dca_decode_init,

  2049.     .decode          = dca_decode_frame,

  2050.     .close           = dca_decode_end,

  2051.     .capabilities    = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,

  2052.     .sample_fmts     = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,

  2053.                                                        AV_SAMPLE_FMT_NONE },

  2054.     .profiles        = NULL_IF_CONFIG_SMALL(profiles),

  2055. };