falkTX
/
FFmpeg
mirror of https://github.com/falkTX/FFmpeg.git
1
0
Fork 0
Code Issues 0 Releases 338 Wiki Activity
You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
38737 Commits
32 Branches
303MB
Tree: 27245b4e1b
Branches Tags
${ item.name }
Create branch ${ searchTerm }
from '27245b4e1b'
${ noResults }
FFmpeg/libavcodec/dcadec.c

1961 lines
72KB
Raw Blame History 

  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][2];                ///< stereo downmix coefficients

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

  341. 

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

  343. 

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

  345.     int lfe_scale_factor;

  346. 

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

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

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

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

  351.     int hist_index[DCA_PRIM_CHANNELS_MAX];

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

  353. 

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

  355. 

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

  357.     float *samples_chanptr[DCA_PRIM_CHANNELS_MAX + 1];

  358.     float *extra_channels[DCA_PRIM_CHANNELS_MAX + 1];

  359.     uint8_t *extra_channels_buffer;

  360.     unsigned int extra_channels_buffer_size;

  361. 

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

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

  364. 

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

  366.     GetBitContext gb;

  367.     /* Current position in DCA frame */

  368.     int current_subframe;

  369.     int current_subsubframe;

  370. 

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

  372. 

  373.     /* XCh extension information */

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

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

  376. 

  377.     /* ExSS header parser */

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

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

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

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

  382. 

  383.     int profile;

  384. 

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

  386.     AVFloatDSPContext fdsp;

  387.     FFTContext imdct;

  388.     SynthFilterContext synth;

  389.     DCADSPContext dcadsp;

  390.     FmtConvertContext fmt_conv;

  391. } DCAContext;

  392. 

  393. static const uint16_t dca_vlc_offs[] = {

  394.         0,   512,   640,   768,  1282,  1794,  2436,  3080,  3770,  4454,  5364,

  395.      5372,  5380,  5388,  5392,  5396,  5412,  5420,  5428,  5460,  5492,  5508,

  396.      5572,  5604,  5668,  5796,  5860,  5892,  6412,  6668,  6796,  7308,  7564,

  397.      7820,  8076,  8620,  9132,  9388,  9910, 10166, 10680, 11196, 11726, 12240,

  398.     12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264,

  399.     18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622,

  400. };

  401. 

  402. static av_cold void dca_init_vlcs(void)

  403. {

  404.     static int vlcs_initialized = 0;

  405.     int i, j, c = 14;

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

  407. 

  408.     if (vlcs_initialized)

  409.         return;

  410. 

  411.     dca_bitalloc_index.offset = 1;

  412.     dca_bitalloc_index.wrap = 2;

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

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

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

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

  417.                  bitalloc_12_bits[i], 1, 1,

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

  419.     }

  420.     dca_scalefactor.offset = -64;

  421.     dca_scalefactor.wrap = 2;

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

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

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

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

  426.                  scales_bits[i], 1, 1,

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

  428.     }

  429.     dca_tmode.offset = 0;

  430.     dca_tmode.wrap = 1;

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

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

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

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

  435.                  tmode_bits[i], 1, 1,

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

  437.     }

  438. 

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

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

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

  442.                 break;

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

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

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

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

  447. 

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

  449.                      bitalloc_sizes[i],

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

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

  452.             c++;

  453.         }

  454.     vlcs_initialized = 1;

  455. }

  456. 

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

  458. {

  459.     while (len--)

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

  461. }

  462. 

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

  464. {

  465.     int i, j;

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

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

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

  469. 

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

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

  472. 

  473.     if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)

  474.         s->prim_channels = DCA_PRIM_CHANNELS_MAX;

  475. 

  476. 

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

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

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

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

  481.     }

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

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

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

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

  486.     }

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

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

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

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

  491. 

  492.     /* Get codebooks quantization indexes */

  493.     if (!base_channel)

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

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

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

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

  498. 

  499.     /* Get scale factor adjustment */

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

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

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

  503. 

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

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

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

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

  508. 

  509.     if (s->crc_present) {

  510.         /* Audio header CRC check */

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

  512.     }

  513. 

  514.     s->current_subframe    = 0;

  515.     s->current_subsubframe = 0;

  516. 

  517. #ifdef TRACE

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

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

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

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

  522.                s->subband_activity[i]);

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

  524.                s->vq_start_subband[i]);

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

  526.                s->joint_intensity[i]);

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

  528.                s->transient_huffman[i]);

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

  530.                s->scalefactor_huffman[i]);

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

  532.                s->bitalloc_huffman[i]);

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

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

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

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

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

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

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

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

  541.     }

  542. #endif

  543. 

  544.     return 0;

  545. }

  546. 

  547. static int dca_parse_frame_header(DCAContext *s)

  548. {

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

  550. 

  551.     /* Sync code */

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

  553. 

  554.     /* Frame header */

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

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

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

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

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

  560.     if (s->frame_size < 95)

  561.         return AVERROR_INVALIDDATA;

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

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

  564.     if (!s->sample_rate)

  565.         return AVERROR_INVALIDDATA;

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

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

  568.     if (!s->bit_rate)

  569.         return AVERROR_INVALIDDATA;

  570. 

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

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

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

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

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

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

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

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

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

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

  581. 

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

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

  584.         return AVERROR_INVALIDDATA;

  585.     }

  586. 

  587.     /* TODO: check CRC */

  588.     if (s->crc_present)

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

  590. 

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

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

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

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

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

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

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

  598. 

  599.     /* FIXME: channels mixing levels */

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

  601.     if (s->lfe)

  602.         s->output |= DCA_LFE;

  603. 

  604. #ifdef TRACE

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

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

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

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

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

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

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

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

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

  614.            s->sample_rate);

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

  616.            s->bit_rate);

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

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

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

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

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

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

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

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

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

  626.            s->predictor_history);

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

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

  629.            s->multirate_inter);

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

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

  632.     av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

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

  639. #endif

  640. 

  641.     /* Primary audio coding header */

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

  643. 

  644.     return dca_parse_audio_coding_header(s, 0);

  645. }

  646. 

  647. 

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

  649. {

  650.     if (level < 5) {

  651.         /* huffman encoded */

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

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

  654.     } else if (level < 8) {

  655.         if (level + 1 > log2range) {

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

  657.             value = get_bits(gb, log2range);

  658.         } else {

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

  660.         }

  661.     }

  662.     return value;

  663. }

  664. 

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

  666. {

  667.     /* Primary audio coding side information */

  668.     int j, k;

  669. 

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

  671.         return AVERROR_INVALIDDATA;

  672. 

  673.     if (!base_channel) {

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

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

  676.     }

  677. 

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

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

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

  681.     }

  682. 

  683.     /* Get prediction codebook */

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

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

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

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

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

  689.             }

  690.         }

  691.     }

  692. 

  693.     /* Bit allocation index */

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

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

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

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

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

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

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

  701.                 av_log(s->avctx, AV_LOG_ERROR,

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

  703.                 return AVERROR_INVALIDDATA;

  704.             } else {

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

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

  707.             }

  708. 

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

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

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

  712.                 return AVERROR_INVALIDDATA;

  713.             }

  714.         }

  715.     }

  716. 

  717.     /* Transition mode */

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

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

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

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

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

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

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

  725.             }

  726.         }

  727.     }

  728. 

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

  730.         return AVERROR_INVALIDDATA;

  731. 

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

  733.         const uint32_t *scale_table;

  734.         int scale_sum, log_size;

  735. 

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

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

  738. 

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

  740.             scale_table = scale_factor_quant7;

  741.             log_size = 7;

  742.         } else {

  743.             scale_table = scale_factor_quant6;

  744.             log_size = 6;

  745.         }

  746. 

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

  748.         scale_sum = 0;

  749. 

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

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

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

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

  754.             }

  755. 

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

  757.                 /* Get second scale factor */

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

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

  760.             }

  761.         }

  762.     }

  763. 

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

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

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

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

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

  769.     }

  770. 

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

  772.         return AVERROR_INVALIDDATA;

  773. 

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

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

  776.         int source_channel;

  777. 

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

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

  780.             int scale = 0;

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

  782. 

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

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

  785. 

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

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

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

  789.             }

  790. 

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

  792.                 av_log(s->avctx, AV_LOG_DEBUG,

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

  794.                 s->debug_flag |= 0x02;

  795.             }

  796.         }

  797.     }

  798. 

  799.     /* Stereo downmix coefficients */

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

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

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

  803.             av_log(s->avctx, AV_LOG_ERROR,

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

  805.             return AVERROR_INVALIDDATA;

  806.         }

  807.         if (s->prim_channels > FF_ARRAY_ELEMS(dca_default_coeffs[0])) {

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

  809.                                   s->prim_channels);

  810.             return AVERROR_PATCHWELCOME;

  811.         }

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

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

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

  815.         }

  816.     }

  817. 

  818.     /* Dynamic range coefficient */

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

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

  821. 

  822.     /* Side information CRC check word */

  823.     if (s->crc_present) {

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

  825.     }

  826. 

  827.     /*

  828.      * Primary audio data arrays

  829.      */

  830. 

  831.     /* VQ encoded high frequency subbands */

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

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

  834.             /* 1 vector -> 32 samples */

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

  836. 

  837.     /* Low frequency effect data */

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

  839.         /* LFE samples */

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

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

  842.         float lfe_scale;

  843. 

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

  845.             /* Signed 8 bits int */

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

  847.         }

  848. 

  849.         /* Scale factor index */

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

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

  852. 

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

  854.         lfe_scale = 0.035 * s->lfe_scale_factor;

  855. 

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

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

  858.     }

  859. 

  860. #ifdef TRACE

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

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

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

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

  865. 

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

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

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

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

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

  871.     }

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

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

  874.             av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

  880.     }

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

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

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

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

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

  886.     }

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

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

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

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

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

  892.     }

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

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

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

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

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

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

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

  900.         }

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

  902.     }

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

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

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

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

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

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

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

  910.         }

  911.     }

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

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

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

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

  916.                    s->downmix_coef[j][0]);

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

  918.                    s->downmix_coef[j][1]);

  919.         }

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

  921.     }

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

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

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

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

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

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

  928. 

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

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

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

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

  933.     }

  934. #endif

  935. 

  936.     return 0;

  937. }

  938. 

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

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

  941.                             float scale)

  942. {

  943.     const float *prCoeff;

  944. 

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

  946. 

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

  948. 

  949.     /* Select filter */

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

  951.         prCoeff = fir_32bands_nonperfect;

  952.     else                        /* Perfect reconstruction */

  953.         prCoeff = fir_32bands_perfect;

  954. 

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

  956.                               s->subband_fir_hist[chans],

  957.                               &s->hist_index[chans],

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

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

  960. }

  961. 

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

  963.                                   int num_deci_sample, float *samples_in,

  964.                                   float *samples_out, float scale)

  965. {

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

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

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

  969.      *   from last subframe as history.

  970.      *

  971.      * samples_out: An array holding interpolated samples

  972.      */

  973. 

  974.     int decifactor;

  975.     const float *prCoeff;

  976.     int deciindex;

  977. 

  978.     /* Select decimation filter */

  979.     if (decimation_select == 1) {

  980.         decifactor = 64;

  981.         prCoeff = lfe_fir_128;

  982.     } else {

  983.         decifactor = 32;

  984.         prCoeff = lfe_fir_64;

  985.     }

  986.     /* Interpolation */

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

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

  989.         samples_in++;

  990.         samples_out += 2 * decifactor;

  991.     }

  992. }

  993. 

  994. /* downmixing routines */

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

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

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

  998. 

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

  1000.     samples[0][i] += samples[s1][i] * coef[rs][0] + samples[s2][i] * coef[rs + 1][0]; \

  1001.     samples[1][i] += samples[s1][i] * coef[rs][1] + samples[s2][i] * coef[rs + 1][1];

  1002. 

  1003. #define MIX_FRONT3(samples, coef)                                      \

  1004.     t = samples[c][i];                                                 \

  1005.     u = samples[l][i];                                                 \

  1006.     v = samples[r][i];                                                 \

  1007.     samples[0][i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0];  \

  1008.     samples[1][i] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];

  1009. 

  1010. #define DOWNMIX_TO_STEREO(op1, op2)             \

  1011.     for (i = 0; i < 256; i++) {                 \

  1012.         op1                                     \

  1013.         op2                                     \

  1014.     }

  1015. 

  1016. static void dca_downmix(float **samples, int srcfmt,

  1017.                         float coef[DCA_PRIM_CHANNELS_MAX][2],

  1018.                         const int8_t *channel_mapping)

  1019. {

  1020.     int c, l, r, sl, sr, s;

  1021.     int i;

  1022.     float t, u, v;

  1023. 

  1024.     switch (srcfmt) {

  1025.     case DCA_MONO:

  1026.     case DCA_CHANNEL:

  1027.     case DCA_STEREO_TOTAL:

  1028.     case DCA_STEREO_SUMDIFF:

  1029.     case DCA_4F2R:

  1030.         av_log(NULL, 0, "Not implemented!\n");

  1031.         break;

  1032.     case DCA_STEREO:

  1033.         break;

  1034.     case DCA_3F:

  1035.         c = channel_mapping[0];

  1036.         l = channel_mapping[1];

  1037.         r = channel_mapping[2];

  1038.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );

  1039.         break;

  1040.     case DCA_2F1R:

  1041.         s = channel_mapping[2];

  1042.         DOWNMIX_TO_STEREO(MIX_REAR1(samples, s, 2, coef), );

  1043.         break;

  1044.     case DCA_3F1R:

  1045.         c = channel_mapping[0];

  1046.         l = channel_mapping[1];

  1047.         r = channel_mapping[2];

  1048.         s = channel_mapping[3];

  1049.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1050.                           MIX_REAR1(samples, s, 3, coef));

  1051.         break;

  1052.     case DCA_2F2R:

  1053.         sl = channel_mapping[2];

  1054.         sr = channel_mapping[3];

  1055.         DOWNMIX_TO_STEREO(MIX_REAR2(samples, sl, sr, 2, coef), );

  1056.         break;

  1057.     case DCA_3F2R:

  1058.         c  = channel_mapping[0];

  1059.         l  = channel_mapping[1];

  1060.         r  = channel_mapping[2];

  1061.         sl = channel_mapping[3];

  1062.         sr = channel_mapping[4];

  1063.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1064.                           MIX_REAR2(samples, sl, sr, 3, coef));

  1065.         break;

  1066.     }

  1067. }

  1068. 

  1069. 

  1070. #ifndef decode_blockcodes

  1071. /* Very compact version of the block code decoder that does not use table

  1072.  * look-up but is slightly slower */

  1073. static int decode_blockcode(int code, int levels, int32_t *values)

  1074. {

  1075.     int i;

  1076.     int offset = (levels - 1) >> 1;

  1077. 

  1078.     for (i = 0; i < 4; i++) {

  1079.         int div = FASTDIV(code, levels);

  1080.         values[i] = code - offset - div * levels;

  1081.         code = div;

  1082.     }

  1083. 

  1084.     return code;

  1085. }

  1086. 

  1087. static int decode_blockcodes(int code1, int code2, int levels, int32_t *values)

  1088. {

  1089.     return decode_blockcode(code1, levels, values) |

  1090.            decode_blockcode(code2, levels, values + 4);

  1091. }

  1092. #endif

  1093. 

  1094. static const uint8_t abits_sizes[7]  = { 7, 10, 12, 13, 15, 17, 19 };

  1095. static const uint8_t abits_levels[7] = { 3,  5,  7,  9, 13, 17, 25 };

  1096. 

  1097. #ifndef int8x8_fmul_int32

  1098. static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)

  1099. {

  1100.     float fscale = scale / 16.0;

  1101.     int i;

  1102.     for (i = 0; i < 8; i++)

  1103.         dst[i] = src[i] * fscale;

  1104. }

  1105. #endif

  1106. 

  1107. static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)

  1108. {

  1109.     int k, l;

  1110.     int subsubframe = s->current_subsubframe;

  1111. 

  1112.     const float *quant_step_table;

  1113. 

  1114.     /* FIXME */

  1115.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1116.     LOCAL_ALIGNED_16(int32_t, block, [8 * DCA_SUBBANDS]);

  1117. 

  1118.     /*

  1119.      * Audio data

  1120.      */

  1121. 

  1122.     /* Select quantization step size table */

  1123.     if (s->bit_rate_index == 0x1f)

  1124.         quant_step_table = lossless_quant_d;

  1125.     else

  1126.         quant_step_table = lossy_quant_d;

  1127. 

  1128.     for (k = base_channel; k < s->prim_channels; k++) {

  1129.         float rscale[DCA_SUBBANDS];

  1130. 

  1131.         if (get_bits_left(&s->gb) < 0)

  1132.             return AVERROR_INVALIDDATA;

  1133. 

  1134.         for (l = 0; l < s->vq_start_subband[k]; l++) {

  1135.             int m;

  1136. 

  1137.             /* Select the mid-tread linear quantizer */

  1138.             int abits = s->bitalloc[k][l];

  1139. 

  1140.             float quant_step_size = quant_step_table[abits];

  1141. 

  1142.             /*

  1143.              * Determine quantization index code book and its type

  1144.              */

  1145. 

  1146.             /* Select quantization index code book */

  1147.             int sel = s->quant_index_huffman[k][abits];

  1148. 

  1149.             /*

  1150.              * Extract bits from the bit stream

  1151.              */

  1152.             if (!abits) {

  1153.                 rscale[l] = 0;

  1154.                 memset(block + 8 * l, 0, 8 * sizeof(block[0]));

  1155.             } else {

  1156.                 /* Deal with transients */

  1157.                 int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];

  1158.                 rscale[l] = quant_step_size * s->scale_factor[k][l][sfi] *

  1159.                                s->scalefactor_adj[k][sel];

  1160. 

  1161.                 if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {

  1162.                     if (abits <= 7) {

  1163.                         /* Block code */

  1164.                         int block_code1, block_code2, size, levels, err;

  1165. 

  1166.                         size   = abits_sizes[abits - 1];

  1167.                         levels = abits_levels[abits - 1];

  1168. 

  1169.                         block_code1 = get_bits(&s->gb, size);

  1170.                         block_code2 = get_bits(&s->gb, size);

  1171.                         err = decode_blockcodes(block_code1, block_code2,

  1172.                                                 levels, block + 8 * l);

  1173.                         if (err) {

  1174.                             av_log(s->avctx, AV_LOG_ERROR,

  1175.                                    "ERROR: block code look-up failed\n");

  1176.                             return AVERROR_INVALIDDATA;

  1177.                         }

  1178.                     } else {

  1179.                         /* no coding */

  1180.                         for (m = 0; m < 8; m++)

  1181.                             block[8 * l + m] = get_sbits(&s->gb, abits - 3);

  1182.                     }

  1183.                 } else {

  1184.                     /* Huffman coded */

  1185.                     for (m = 0; m < 8; m++)

  1186.                         block[8 * l + m] = get_bitalloc(&s->gb,

  1187.                                                 &dca_smpl_bitalloc[abits], sel);

  1188.                 }

  1189. 

  1190.             }

  1191.         }

  1192. 

  1193.         s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[k][0],

  1194.                                                block, rscale, 8 * s->vq_start_subband[k]);

  1195. 

  1196.         for (l = 0; l < s->vq_start_subband[k]; l++) {

  1197.             int m;

  1198.             /*

  1199.              * Inverse ADPCM if in prediction mode

  1200.              */

  1201.             if (s->prediction_mode[k][l]) {

  1202.                 int n;

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

  1204.                     for (n = 1; n <= 4; n++)

  1205.                         if (m >= n)

  1206.                             subband_samples[k][l][m] +=

  1207.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1208.                                  subband_samples[k][l][m - n] / 8192);

  1209.                         else if (s->predictor_history)

  1210.                             subband_samples[k][l][m] +=

  1211.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1212.                                  s->subband_samples_hist[k][l][m - n + 4] / 8192);

  1213.                 }

  1214.             }

  1215.         }

  1216. 

  1217.         /*

  1218.          * Decode VQ encoded high frequencies

  1219.          */

  1220.         for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {

  1221.             /* 1 vector -> 32 samples but we only need the 8 samples

  1222.              * for this subsubframe. */

  1223.             int hfvq = s->high_freq_vq[k][l];

  1224. 

  1225.             if (!s->debug_flag & 0x01) {

  1226.                 av_log(s->avctx, AV_LOG_DEBUG,

  1227.                        "Stream with high frequencies VQ coding\n");

  1228.                 s->debug_flag |= 0x01;

  1229.             }

  1230. 

  1231.             int8x8_fmul_int32(subband_samples[k][l],

  1232.                               &high_freq_vq[hfvq][subsubframe * 8],

  1233.                               s->scale_factor[k][l][0]);

  1234.         }

  1235.     }

  1236. 

  1237.     /* Check for DSYNC after subsubframe */

  1238.     if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {

  1239.         if (0xFFFF == get_bits(&s->gb, 16)) {   /* 0xFFFF */

  1240. #ifdef TRACE

  1241.             av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");

  1242. #endif

  1243.         } else {

  1244.             av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");

  1245.             return AVERROR_INVALIDDATA;

  1246.         }

  1247.     }

  1248. 

  1249.     /* Backup predictor history for adpcm */

  1250.     for (k = base_channel; k < s->prim_channels; k++)

  1251.         for (l = 0; l < s->vq_start_subband[k]; l++)

  1252.             memcpy(s->subband_samples_hist[k][l],

  1253.                    &subband_samples[k][l][4],

  1254.                    4 * sizeof(subband_samples[0][0][0]));

  1255. 

  1256.     return 0;

  1257. }

  1258. 

  1259. static int dca_filter_channels(DCAContext *s, int block_index)

  1260. {

  1261.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1262.     int k;

  1263. 

  1264.     /* 32 subbands QMF */

  1265.     for (k = 0; k < s->prim_channels; k++) {

  1266. /*        static float pcm_to_double[8] = { 32768.0, 32768.0, 524288.0, 524288.0,

  1267.                                             0, 8388608.0, 8388608.0 };*/

  1268.         if (s->channel_order_tab[k] >= 0)

  1269.             qmf_32_subbands(s, k, subband_samples[k],

  1270.                             s->samples_chanptr[s->channel_order_tab[k]],

  1271.                             M_SQRT1_2 / 32768.0 /* pcm_to_double[s->source_pcm_res] */);

  1272.     }

  1273. 

  1274.     /* Down mixing */

  1275.     if (s->prim_channels > 2 &&

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

  1277.         dca_downmix(s->samples_chanptr, s->amode, s->downmix_coef, s->channel_order_tab);

  1278.     }

  1279. 

  1280.     /* Generate LFE samples for this subsubframe FIXME!!! */

  1281.     if (s->output & DCA_LFE) {

  1282.         lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,

  1283.                               s->lfe_data + 2 * s->lfe * (block_index + 4),

  1284.                               s->samples_chanptr[dca_lfe_index[s->amode]],

  1285.                               1.0 / (256.0 * 32768.0));

  1286.         /* Outputs 20bits pcm samples */

  1287.     }

  1288. 

  1289.     return 0;

  1290. }

  1291. 

  1292. 

  1293. static int dca_subframe_footer(DCAContext *s, int base_channel)

  1294. {

  1295.     int aux_data_count = 0, i;

  1296. 

  1297.     /*

  1298.      * Unpack optional information

  1299.      */

  1300. 

  1301.     /* presumably optional information only appears in the core? */

  1302.     if (!base_channel) {

  1303.         if (s->timestamp)

  1304.             skip_bits_long(&s->gb, 32);

  1305. 

  1306.         if (s->aux_data)

  1307.             aux_data_count = get_bits(&s->gb, 6);

  1308. 

  1309.         for (i = 0; i < aux_data_count; i++)

  1310.             get_bits(&s->gb, 8);

  1311. 

  1312.         if (s->crc_present && s->dynrange)

  1313.             get_bits(&s->gb, 16);

  1314.     }

  1315. 

  1316.     return 0;

  1317. }

  1318. 

  1319. /**

  1320.  * Decode a dca frame block

  1321.  *

  1322.  * @param s     pointer to the DCAContext

  1323.  */

  1324. 

  1325. static int dca_decode_block(DCAContext *s, int base_channel, int block_index)

  1326. {

  1327.     int ret;

  1328. 

  1329.     /* Sanity check */

  1330.     if (s->current_subframe >= s->subframes) {

  1331.         av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",

  1332.                s->current_subframe, s->subframes);

  1333.         return AVERROR_INVALIDDATA;

  1334.     }

  1335. 

  1336.     if (!s->current_subsubframe) {

  1337. #ifdef TRACE

  1338.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");

  1339. #endif

  1340.         /* Read subframe header */

  1341.         if ((ret = dca_subframe_header(s, base_channel, block_index)))

  1342.             return ret;

  1343.     }

  1344. 

  1345.     /* Read subsubframe */

  1346. #ifdef TRACE

  1347.     av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");

  1348. #endif

  1349.     if ((ret = dca_subsubframe(s, base_channel, block_index)))

  1350.         return ret;

  1351. 

  1352.     /* Update state */

  1353.     s->current_subsubframe++;

  1354.     if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {

  1355.         s->current_subsubframe = 0;

  1356.         s->current_subframe++;

  1357.     }

  1358.     if (s->current_subframe >= s->subframes) {

  1359. #ifdef TRACE

  1360.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");

  1361. #endif

  1362.         /* Read subframe footer */

  1363.         if ((ret = dca_subframe_footer(s, base_channel)))

  1364.             return ret;

  1365.     }

  1366. 

  1367.     return 0;

  1368. }

  1369. 

  1370. /**

  1371.  * Return the number of channels in an ExSS speaker mask (HD)

  1372.  */

  1373. static int dca_exss_mask2count(int mask)

  1374. {

  1375.     /* count bits that mean speaker pairs twice */

  1376.     return av_popcount(mask) +

  1377.            av_popcount(mask & (DCA_EXSS_CENTER_LEFT_RIGHT      |

  1378.                                DCA_EXSS_FRONT_LEFT_RIGHT       |

  1379.                                DCA_EXSS_FRONT_HIGH_LEFT_RIGHT  |

  1380.                                DCA_EXSS_WIDE_LEFT_RIGHT        |

  1381.                                DCA_EXSS_SIDE_LEFT_RIGHT        |

  1382.                                DCA_EXSS_SIDE_HIGH_LEFT_RIGHT   |

  1383.                                DCA_EXSS_SIDE_REAR_LEFT_RIGHT   |

  1384.                                DCA_EXSS_REAR_LEFT_RIGHT        |

  1385.                                DCA_EXSS_REAR_HIGH_LEFT_RIGHT));

  1386. }

  1387. 

  1388. /**

  1389.  * Skip mixing coefficients of a single mix out configuration (HD)

  1390.  */

  1391. static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch)

  1392. {

  1393.     int i;

  1394. 

  1395.     for (i = 0; i < channels; i++) {

  1396.         int mix_map_mask = get_bits(gb, out_ch);

  1397.         int num_coeffs = av_popcount(mix_map_mask);

  1398.         skip_bits_long(gb, num_coeffs * 6);

  1399.     }

  1400. }

  1401. 

  1402. /**

  1403.  * Parse extension substream asset header (HD)

  1404.  */

  1405. static int dca_exss_parse_asset_header(DCAContext *s)

  1406. {

  1407.     int header_pos = get_bits_count(&s->gb);

  1408.     int header_size;

  1409.     int channels;

  1410.     int embedded_stereo = 0;

  1411.     int embedded_6ch    = 0;

  1412.     int drc_code_present;

  1413.     int extensions_mask;

  1414.     int i, j;

  1415. 

  1416.     if (get_bits_left(&s->gb) < 16)

  1417.         return -1;

  1418. 

  1419.     /* We will parse just enough to get to the extensions bitmask with which

  1420.      * we can set the profile value. */

  1421. 

  1422.     header_size = get_bits(&s->gb, 9) + 1;

  1423.     skip_bits(&s->gb, 3); // asset index

  1424. 

  1425.     if (s->static_fields) {

  1426.         if (get_bits1(&s->gb))

  1427.             skip_bits(&s->gb, 4); // asset type descriptor

  1428.         if (get_bits1(&s->gb))

  1429.             skip_bits_long(&s->gb, 24); // language descriptor

  1430. 

  1431.         if (get_bits1(&s->gb)) {

  1432.             /* How can one fit 1024 bytes of text here if the maximum value

  1433.              * for the asset header size field above was 512 bytes? */

  1434.             int text_length = get_bits(&s->gb, 10) + 1;

  1435.             if (get_bits_left(&s->gb) < text_length * 8)

  1436.                 return -1;

  1437.             skip_bits_long(&s->gb, text_length * 8); // info text

  1438.         }

  1439. 

  1440.         skip_bits(&s->gb, 5); // bit resolution - 1

  1441.         skip_bits(&s->gb, 4); // max sample rate code

  1442.         channels = get_bits(&s->gb, 8) + 1;

  1443. 

  1444.         if (get_bits1(&s->gb)) { // 1-to-1 channels to speakers

  1445.             int spkr_remap_sets;

  1446.             int spkr_mask_size = 16;

  1447.             int num_spkrs[7];

  1448. 

  1449.             if (channels > 2)

  1450.                 embedded_stereo = get_bits1(&s->gb);

  1451.             if (channels > 6)

  1452.                 embedded_6ch = get_bits1(&s->gb);

  1453. 

  1454.             if (get_bits1(&s->gb)) {

  1455.                 spkr_mask_size = (get_bits(&s->gb, 2) + 1) << 2;

  1456.                 skip_bits(&s->gb, spkr_mask_size); // spkr activity mask

  1457.             }

  1458. 

  1459.             spkr_remap_sets = get_bits(&s->gb, 3);

  1460. 

  1461.             for (i = 0; i < spkr_remap_sets; i++) {

  1462.                 /* std layout mask for each remap set */

  1463.                 num_spkrs[i] = dca_exss_mask2count(get_bits(&s->gb, spkr_mask_size));

  1464.             }

  1465. 

  1466.             for (i = 0; i < spkr_remap_sets; i++) {

  1467.                 int num_dec_ch_remaps = get_bits(&s->gb, 5) + 1;

  1468.                 if (get_bits_left(&s->gb) < 0)

  1469.                     return -1;

  1470. 

  1471.                 for (j = 0; j < num_spkrs[i]; j++) {

  1472.                     int remap_dec_ch_mask = get_bits_long(&s->gb, num_dec_ch_remaps);

  1473.                     int num_dec_ch = av_popcount(remap_dec_ch_mask);

  1474.                     skip_bits_long(&s->gb, num_dec_ch * 5); // remap codes

  1475.                 }

  1476.             }

  1477. 

  1478.         } else {

  1479.             skip_bits(&s->gb, 3); // representation type

  1480.         }

  1481.     }

  1482. 

  1483.     drc_code_present = get_bits1(&s->gb);

  1484.     if (drc_code_present)

  1485.         get_bits(&s->gb, 8); // drc code

  1486. 

  1487.     if (get_bits1(&s->gb))

  1488.         skip_bits(&s->gb, 5); // dialog normalization code

  1489. 

  1490.     if (drc_code_present && embedded_stereo)

  1491.         get_bits(&s->gb, 8); // drc stereo code

  1492. 

  1493.     if (s->mix_metadata && get_bits1(&s->gb)) {

  1494.         skip_bits(&s->gb, 1); // external mix

  1495.         skip_bits(&s->gb, 6); // post mix gain code

  1496. 

  1497.         if (get_bits(&s->gb, 2) != 3) // mixer drc code

  1498.             skip_bits(&s->gb, 3); // drc limit

  1499.         else

  1500.             skip_bits(&s->gb, 8); // custom drc code

  1501. 

  1502.         if (get_bits1(&s->gb)) // channel specific scaling

  1503.             for (i = 0; i < s->num_mix_configs; i++)

  1504.                 skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6); // scale codes

  1505.         else

  1506.             skip_bits_long(&s->gb, s->num_mix_configs * 6); // scale codes

  1507. 

  1508.         for (i = 0; i < s->num_mix_configs; i++) {

  1509.             if (get_bits_left(&s->gb) < 0)

  1510.                 return -1;

  1511.             dca_exss_skip_mix_coeffs(&s->gb, channels, s->mix_config_num_ch[i]);

  1512.             if (embedded_6ch)

  1513.                 dca_exss_skip_mix_coeffs(&s->gb, 6, s->mix_config_num_ch[i]);

  1514.             if (embedded_stereo)

  1515.                 dca_exss_skip_mix_coeffs(&s->gb, 2, s->mix_config_num_ch[i]);

  1516.         }

  1517.     }

  1518. 

  1519.     switch (get_bits(&s->gb, 2)) {

  1520.     case 0: extensions_mask = get_bits(&s->gb, 12); break;

  1521.     case 1: extensions_mask = DCA_EXT_EXSS_XLL;     break;

  1522.     case 2: extensions_mask = DCA_EXT_EXSS_LBR;     break;

  1523.     case 3: extensions_mask = 0; /* aux coding */   break;

  1524.     }

  1525. 

  1526.     /* not parsed further, we were only interested in the extensions mask */

  1527. 

  1528.     if (get_bits_left(&s->gb) < 0)

  1529.         return -1;

  1530. 

  1531.     if (get_bits_count(&s->gb) - header_pos > header_size * 8) {

  1532.         av_log(s->avctx, AV_LOG_WARNING, "Asset header size mismatch.\n");

  1533.         return -1;

  1534.     }

  1535.     skip_bits_long(&s->gb, header_pos + header_size * 8 - get_bits_count(&s->gb));

  1536. 

  1537.     if (extensions_mask & DCA_EXT_EXSS_XLL)

  1538.         s->profile = FF_PROFILE_DTS_HD_MA;

  1539.     else if (extensions_mask & (DCA_EXT_EXSS_XBR | DCA_EXT_EXSS_X96 |

  1540.                                 DCA_EXT_EXSS_XXCH))

  1541.         s->profile = FF_PROFILE_DTS_HD_HRA;

  1542. 

  1543.     if (!(extensions_mask & DCA_EXT_CORE))

  1544.         av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n");

  1545.     if ((extensions_mask & DCA_CORE_EXTS) != s->core_ext_mask)

  1546.         av_log(s->avctx, AV_LOG_WARNING,

  1547.                "DTS extensions detection mismatch (%d, %d)\n",

  1548.                extensions_mask & DCA_CORE_EXTS, s->core_ext_mask);

  1549. 

  1550.     return 0;

  1551. }

  1552. 

  1553. /**

  1554.  * Parse extension substream header (HD)

  1555.  */

  1556. static void dca_exss_parse_header(DCAContext *s)

  1557. {

  1558.     int ss_index;

  1559.     int blownup;

  1560.     int num_audiop = 1;

  1561.     int num_assets = 1;

  1562.     int active_ss_mask[8];

  1563.     int i, j;

  1564. 

  1565.     if (get_bits_left(&s->gb) < 52)

  1566.         return;

  1567. 

  1568.     skip_bits(&s->gb, 8); // user data

  1569.     ss_index = get_bits(&s->gb, 2);

  1570. 

  1571.     blownup = get_bits1(&s->gb);

  1572.     skip_bits(&s->gb,  8 + 4 * blownup); // header_size

  1573.     skip_bits(&s->gb, 16 + 4 * blownup); // hd_size

  1574. 

  1575.     s->static_fields = get_bits1(&s->gb);

  1576.     if (s->static_fields) {

  1577.         skip_bits(&s->gb, 2); // reference clock code

  1578.         skip_bits(&s->gb, 3); // frame duration code

  1579. 

  1580.         if (get_bits1(&s->gb))

  1581.             skip_bits_long(&s->gb, 36); // timestamp

  1582. 

  1583.         /* a single stream can contain multiple audio assets that can be

  1584.          * combined to form multiple audio presentations */

  1585. 

  1586.         num_audiop = get_bits(&s->gb, 3) + 1;

  1587.         if (num_audiop > 1) {

  1588.             avpriv_request_sample(s->avctx,

  1589.                                   "Multiple DTS-HD audio presentations");

  1590.             /* ignore such streams for now */

  1591.             return;

  1592.         }

  1593. 

  1594.         num_assets = get_bits(&s->gb, 3) + 1;

  1595.         if (num_assets > 1) {

  1596.             avpriv_request_sample(s->avctx, "Multiple DTS-HD audio assets");

  1597.             /* ignore such streams for now */

  1598.             return;

  1599.         }

  1600. 

  1601.         for (i = 0; i < num_audiop; i++)

  1602.             active_ss_mask[i] = get_bits(&s->gb, ss_index + 1);

  1603. 

  1604.         for (i = 0; i < num_audiop; i++)

  1605.             for (j = 0; j <= ss_index; j++)

  1606.                 if (active_ss_mask[i] & (1 << j))

  1607.                     skip_bits(&s->gb, 8); // active asset mask

  1608. 

  1609.         s->mix_metadata = get_bits1(&s->gb);

  1610.         if (s->mix_metadata) {

  1611.             int mix_out_mask_size;

  1612. 

  1613.             skip_bits(&s->gb, 2); // adjustment level

  1614.             mix_out_mask_size  = (get_bits(&s->gb, 2) + 1) << 2;

  1615.             s->num_mix_configs =  get_bits(&s->gb, 2) + 1;

  1616. 

  1617.             for (i = 0; i < s->num_mix_configs; i++) {

  1618.                 int mix_out_mask        = get_bits(&s->gb, mix_out_mask_size);

  1619.                 s->mix_config_num_ch[i] = dca_exss_mask2count(mix_out_mask);

  1620.             }

  1621.         }

  1622.     }

  1623. 

  1624.     for (i = 0; i < num_assets; i++)

  1625.         skip_bits_long(&s->gb, 16 + 4 * blownup);  // asset size

  1626. 

  1627.     for (i = 0; i < num_assets; i++) {

  1628.         if (dca_exss_parse_asset_header(s))

  1629.             return;

  1630.     }

  1631. 

  1632.     /* not parsed further, we were only interested in the extensions mask

  1633.      * from the asset header */

  1634. }

  1635. 

  1636. /**

  1637.  * Main frame decoding function

  1638.  * FIXME add arguments

  1639.  */

  1640. static int dca_decode_frame(AVCodecContext *avctx, void *data,

  1641.                             int *got_frame_ptr, AVPacket *avpkt)

  1642. {

  1643.     AVFrame *frame     = data;

  1644.     const uint8_t *buf = avpkt->data;

  1645.     int buf_size = avpkt->size;

  1646. 

  1647.     int lfe_samples;

  1648.     int num_core_channels = 0;

  1649.     int i, ret;

  1650.     float  **samples_flt;

  1651.     DCAContext *s = avctx->priv_data;

  1652.     int channels, full_channels;

  1653.     int core_ss_end;

  1654. 

  1655. 

  1656.     s->xch_present = 0;

  1657. 

  1658.     s->dca_buffer_size = ff_dca_convert_bitstream(buf, buf_size, s->dca_buffer,

  1659.                                                   DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);

  1660.     if (s->dca_buffer_size == AVERROR_INVALIDDATA) {

  1661.         av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");

  1662.         return AVERROR_INVALIDDATA;

  1663.     }

  1664. 

  1665.     init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);

  1666.     if ((ret = dca_parse_frame_header(s)) < 0) {

  1667.         //seems like the frame is corrupt, try with the next one

  1668.         return ret;

  1669.     }

  1670.     //set AVCodec values with parsed data

  1671.     avctx->sample_rate = s->sample_rate;

  1672.     avctx->bit_rate    = s->bit_rate;

  1673. 

  1674.     s->profile = FF_PROFILE_DTS;

  1675. 

  1676.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1677.         if ((ret = dca_decode_block(s, 0, i))) {

  1678.             av_log(avctx, AV_LOG_ERROR, "error decoding block\n");

  1679.             return ret;

  1680.         }

  1681.     }

  1682. 

  1683.     /* record number of core channels incase less than max channels are requested */

  1684.     num_core_channels = s->prim_channels;

  1685. 

  1686.     if (s->ext_coding)

  1687.         s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];

  1688.     else

  1689.         s->core_ext_mask = 0;

  1690. 

  1691.     core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;

  1692. 

  1693.     /* only scan for extensions if ext_descr was unknown or indicated a

  1694.      * supported XCh extension */

  1695.     if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {

  1696. 

  1697.         /* if ext_descr was unknown, clear s->core_ext_mask so that the

  1698.          * extensions scan can fill it up */

  1699.         s->core_ext_mask = FFMAX(s->core_ext_mask, 0);

  1700. 

  1701.         /* extensions start at 32-bit boundaries into bitstream */

  1702.         skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1703. 

  1704.         while (core_ss_end - get_bits_count(&s->gb) >= 32) {

  1705.             uint32_t bits = get_bits_long(&s->gb, 32);

  1706. 

  1707.             switch (bits) {

  1708.             case 0x5a5a5a5a: {

  1709.                 int ext_amode, xch_fsize;

  1710. 

  1711.                 s->xch_base_channel = s->prim_channels;

  1712. 

  1713.                 /* validate sync word using XCHFSIZE field */

  1714.                 xch_fsize = show_bits(&s->gb, 10);

  1715.                 if ((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&

  1716.                     (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))

  1717.                     continue;

  1718. 

  1719.                 /* skip length-to-end-of-frame field for the moment */

  1720.                 skip_bits(&s->gb, 10);

  1721. 

  1722.                 s->core_ext_mask |= DCA_EXT_XCH;

  1723. 

  1724.                 /* extension amode(number of channels in extension) should be 1 */

  1725.                 /* AFAIK XCh is not used for more channels */

  1726.                 if ((ext_amode = get_bits(&s->gb, 4)) != 1) {

  1727.                     av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"

  1728.                            " supported!\n", ext_amode);

  1729.                     continue;

  1730.                 }

  1731. 

  1732.                 /* much like core primary audio coding header */

  1733.                 dca_parse_audio_coding_header(s, s->xch_base_channel);

  1734. 

  1735.                 for (i = 0; i < (s->sample_blocks / 8); i++)

  1736.                     if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {

  1737.                         av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");

  1738.                         continue;

  1739.                     }

  1740. 

  1741.                 s->xch_present = 1;

  1742.                 break;

  1743.             }

  1744.             case 0x47004a03:

  1745.                 /* XXCh: extended channels */

  1746.                 /* usually found either in core or HD part in DTS-HD HRA streams,

  1747.                  * but not in DTS-ES which contains XCh extensions instead */

  1748.                 s->core_ext_mask |= DCA_EXT_XXCH;

  1749.                 break;

  1750. 

  1751.             case 0x1d95f262: {

  1752.                 int fsize96 = show_bits(&s->gb, 12) + 1;

  1753.                 if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)

  1754.                     continue;

  1755. 

  1756.                 av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n",

  1757.                        get_bits_count(&s->gb));

  1758.                 skip_bits(&s->gb, 12);

  1759.                 av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);

  1760.                 av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));

  1761. 

  1762.                 s->core_ext_mask |= DCA_EXT_X96;

  1763.                 break;

  1764.             }

  1765.             }

  1766. 

  1767.             skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1768.         }

  1769.     } else {

  1770.         /* no supported extensions, skip the rest of the core substream */

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

  1772.     }

  1773. 

  1774.     if (s->core_ext_mask & DCA_EXT_X96)

  1775.         s->profile = FF_PROFILE_DTS_96_24;

  1776.     else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))

  1777.         s->profile = FF_PROFILE_DTS_ES;

  1778. 

  1779.     /* check for ExSS (HD part) */

  1780.     if (s->dca_buffer_size - s->frame_size > 32 &&

  1781.         get_bits_long(&s->gb, 32) == DCA_HD_MARKER)

  1782.         dca_exss_parse_header(s);

  1783. 

  1784.     avctx->profile = s->profile;

  1785. 

  1786.     full_channels = channels = s->prim_channels + !!s->lfe;

  1787. 

  1788.     if (s->amode < 16) {

  1789.         avctx->channel_layout = dca_core_channel_layout[s->amode];

  1790. 

  1791. #if FF_API_REQUEST_CHANNELS

  1792. FF_DISABLE_DEPRECATION_WARNINGS

  1793.         if (s->xch_present && (!avctx->request_channels ||

  1794.                                avctx->request_channels > num_core_channels + !!s->lfe)) {

  1795. FF_ENABLE_DEPRECATION_WARNINGS

  1796. #else

  1797.         if (s->xch_present) {

  1798. #endif

  1799.             avctx->channel_layout |= AV_CH_BACK_CENTER;

  1800.             if (s->lfe) {

  1801.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1802.                 s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode];

  1803.             } else {

  1804.                 s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];

  1805.             }

  1806.         } else {

  1807.             channels = num_core_channels + !!s->lfe;

  1808.             s->xch_present = 0; /* disable further xch processing */

  1809.             if (s->lfe) {

  1810.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1811.                 s->channel_order_tab = dca_channel_reorder_lfe[s->amode];

  1812.             } else

  1813.                 s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];

  1814.         }

  1815. 

  1816.         if (channels > !!s->lfe &&

  1817.             s->channel_order_tab[channels - 1 - !!s->lfe] < 0)

  1818.             return AVERROR_INVALIDDATA;

  1819. 

  1820.         if (s->prim_channels > 2 &&

  1821.             avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {

  1822.             channels = 2;

  1823.             s->output = DCA_STEREO;

  1824.             avctx->channel_layout = AV_CH_LAYOUT_STEREO;

  1825.         }

  1826.     } else {

  1827.         av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);

  1828.         return AVERROR_INVALIDDATA;

  1829.     }

  1830.     avctx->channels = channels;

  1831. 

  1832.     /* get output buffer */

  1833.     frame->nb_samples = 256 * (s->sample_blocks / 8);

  1834.     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {

  1835.         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");

  1836.         return ret;

  1837.     }

  1838.     samples_flt = (float **)frame->extended_data;

  1839. 

  1840.     /* allocate buffer for extra channels if downmixing */

  1841.     if (avctx->channels < full_channels) {

  1842.         ret = av_samples_get_buffer_size(NULL, full_channels - channels,

  1843.                                          frame->nb_samples,

  1844.                                          avctx->sample_fmt, 0);

  1845.         if (ret < 0)

  1846.             return ret;

  1847. 

  1848.         av_fast_malloc(&s->extra_channels_buffer,

  1849.                        &s->extra_channels_buffer_size, ret);

  1850.         if (!s->extra_channels_buffer)

  1851.             return AVERROR(ENOMEM);

  1852. 

  1853.         ret = av_samples_fill_arrays((uint8_t **)s->extra_channels, NULL,

  1854.                                      s->extra_channels_buffer,

  1855.                                      full_channels - channels,

  1856.                                      frame->nb_samples, avctx->sample_fmt, 0);

  1857.         if (ret < 0)

  1858.             return ret;

  1859.     }

  1860. 

  1861.     /* filter to get final output */

  1862.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1863.         int ch;

  1864. 

  1865.         for (ch = 0; ch < channels; ch++)

  1866.             s->samples_chanptr[ch] = samples_flt[ch] + i * 256;

  1867.         for (; ch < full_channels; ch++)

  1868.             s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * 256;

  1869. 

  1870.         dca_filter_channels(s, i);

  1871. 

  1872.         /* If this was marked as a DTS-ES stream we need to subtract back- */

  1873.         /* channel from SL & SR to remove matrixed back-channel signal */

  1874.         if ((s->source_pcm_res & 1) && s->xch_present) {

  1875.             float *back_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel]];

  1876.             float *lt_chan   = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 2]];

  1877.             float *rt_chan   = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 1]];

  1878.             s->fdsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);

  1879.             s->fdsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);

  1880.         }

  1881.     }

  1882. 

  1883.     /* update lfe history */

  1884.     lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);

  1885.     for (i = 0; i < 2 * s->lfe * 4; i++)

  1886.         s->lfe_data[i] = s->lfe_data[i + lfe_samples];

  1887. 

  1888.     *got_frame_ptr = 1;

  1889. 

  1890.     return buf_size;

  1891. }

  1892. 

  1893. 

  1894. 

  1895. /**

  1896.  * DCA initialization

  1897.  *

  1898.  * @param avctx     pointer to the AVCodecContext

  1899.  */

  1900. 

  1901. static av_cold int dca_decode_init(AVCodecContext *avctx)

  1902. {

  1903.     DCAContext *s = avctx->priv_data;

  1904. 

  1905.     s->avctx = avctx;

  1906.     dca_init_vlcs();

  1907. 

  1908.     avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);

  1909.     ff_mdct_init(&s->imdct, 6, 1, 1.0);

  1910.     ff_synth_filter_init(&s->synth);

  1911.     ff_dcadsp_init(&s->dcadsp);

  1912.     ff_fmt_convert_init(&s->fmt_conv, avctx);

  1913. 

  1914.     avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

  1915. 

  1916.     /* allow downmixing to stereo */

  1917. #if FF_API_REQUEST_CHANNELS

  1918. FF_DISABLE_DEPRECATION_WARNINGS

  1919.     if (avctx->request_channels == 2)

  1920.         avctx->request_channel_layout = AV_CH_LAYOUT_STEREO;

  1921. FF_ENABLE_DEPRECATION_WARNINGS

  1922. #endif

  1923.     if (avctx->channels > 2 &&

  1924.         avctx->request_channel_layout == AV_CH_LAYOUT_STEREO)

  1925.         avctx->channels = 2;

  1926. 

  1927.     return 0;

  1928. }

  1929. 

  1930. static av_cold int dca_decode_end(AVCodecContext *avctx)

  1931. {

  1932.     DCAContext *s = avctx->priv_data;

  1933.     ff_mdct_end(&s->imdct);

  1934.     av_freep(&s->extra_channels_buffer);

  1935.     return 0;

  1936. }

  1937. 

  1938. static const AVProfile profiles[] = {

  1939.     { FF_PROFILE_DTS,        "DTS"        },

  1940.     { FF_PROFILE_DTS_ES,     "DTS-ES"     },

  1941.     { FF_PROFILE_DTS_96_24,  "DTS 96/24"  },

  1942.     { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },

  1943.     { FF_PROFILE_DTS_HD_MA,  "DTS-HD MA"  },

  1944.     { FF_PROFILE_UNKNOWN },

  1945. };

  1946. 

  1947. AVCodec ff_dca_decoder = {

  1948.     .name            = "dca",

  1949.     .long_name       = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),

  1950.     .type            = AVMEDIA_TYPE_AUDIO,

  1951.     .id              = AV_CODEC_ID_DTS,

  1952.     .priv_data_size  = sizeof(DCAContext),

  1953.     .init            = dca_decode_init,

  1954.     .decode          = dca_decode_frame,

  1955.     .close           = dca_decode_end,

  1956.     .capabilities    = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,

  1957.     .sample_fmts     = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,

  1958.                                                        AV_SAMPLE_FMT_NONE },

  1959.     .profiles        = NULL_IF_CONFIG_SMALL(profiles),

  1960. };