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FFmpeg/libavcodec/dca.c

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  1. /*

  2.  * DCA compatible decoder

  3.  * Copyright (C) 2004 Gildas Bazin

  4.  * Copyright (C) 2004 Benjamin Zores

  5.  * Copyright (C) 2006 Benjamin Larsson

  6.  * Copyright (C) 2007 Konstantin Shishkov

  7.  *

  8.  * This file is part of FFmpeg.

  9.  *

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

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

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

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

  14.  *

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

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

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

  18.  * Lesser General Public License for more details.

  19.  *

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

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

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

  23.  */

  24. 

  25. #include <math.h>

  26. #include <stddef.h>

  27. #include <stdio.h>

  28. 

  29. #include "libavutil/common.h"

  30. #include "libavutil/intmath.h"

  31. #include "libavutil/intreadwrite.h"

  32. #include "libavutil/audioconvert.h"

  33. #include "avcodec.h"

  34. #include "dsputil.h"

  35. #include "fft.h"

  36. #include "get_bits.h"

  37. #include "put_bits.h"

  38. #include "dcadata.h"

  39. #include "dcahuff.h"

  40. #include "dca.h"

  41. #include "dca_parser.h"

  42. #include "synth_filter.h"

  43. #include "dcadsp.h"

  44. #include "fmtconvert.h"

  45. 

  46. #if ARCH_ARM

  47. #   include "arm/dca.h"

  48. #endif

  49. 

  50. //#define TRACE

  51. 

  52. #define DCA_PRIM_CHANNELS_MAX  (7)

  53. #define DCA_SUBBANDS          (32)

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

  55. #define DCA_SUBSUBFRAMES_MAX   (4)

  56. #define DCA_SUBFRAMES_MAX     (16)

  57. #define DCA_BLOCKS_MAX        (16)

  58. #define DCA_LFE_MAX            (3)

  59. 

  60. enum DCAMode {

  61.     DCA_MONO = 0,

  62.     DCA_CHANNEL,

  63.     DCA_STEREO,

  64.     DCA_STEREO_SUMDIFF,

  65.     DCA_STEREO_TOTAL,

  66.     DCA_3F,

  67.     DCA_2F1R,

  68.     DCA_3F1R,

  69.     DCA_2F2R,

  70.     DCA_3F2R,

  71.     DCA_4F2R

  72. };

  73. 

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

  75. enum DCAExSSSpeakerMask {

  76.     DCA_EXSS_FRONT_CENTER          = 0x0001,

  77.     DCA_EXSS_FRONT_LEFT_RIGHT      = 0x0002,

  78.     DCA_EXSS_SIDE_REAR_LEFT_RIGHT  = 0x0004,

  79.     DCA_EXSS_LFE                   = 0x0008,

  80.     DCA_EXSS_REAR_CENTER           = 0x0010,

  81.     DCA_EXSS_FRONT_HIGH_LEFT_RIGHT = 0x0020,

  82.     DCA_EXSS_REAR_LEFT_RIGHT       = 0x0040,

  83.     DCA_EXSS_FRONT_HIGH_CENTER     = 0x0080,

  84.     DCA_EXSS_OVERHEAD              = 0x0100,

  85.     DCA_EXSS_CENTER_LEFT_RIGHT     = 0x0200,

  86.     DCA_EXSS_WIDE_LEFT_RIGHT       = 0x0400,

  87.     DCA_EXSS_SIDE_LEFT_RIGHT       = 0x0800,

  88.     DCA_EXSS_LFE2                  = 0x1000,

  89.     DCA_EXSS_SIDE_HIGH_LEFT_RIGHT  = 0x2000,

  90.     DCA_EXSS_REAR_HIGH_CENTER      = 0x4000,

  91.     DCA_EXSS_REAR_HIGH_LEFT_RIGHT  = 0x8000,

  92. };

  93. 

  94. enum DCAExtensionMask {

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

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

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

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

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

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

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

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

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

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

  105. };

  106. 

  107. /* -1 are reserved or unknown */

  108. static const int dca_ext_audio_descr_mask[] = {

  109.     DCA_EXT_XCH,

  110.     -1,

  111.     DCA_EXT_X96,

  112.     DCA_EXT_XCH | DCA_EXT_X96,

  113.     -1,

  114.     -1,

  115.     DCA_EXT_XXCH,

  116.     -1,

  117. };

  118. 

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

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

  121. 

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

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

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

  125.  *

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

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

  128.  * OV -> center back

  129.  * All 2 channel configurations -> AV_CH_LAYOUT_STEREO

  130.  */

  131. static const uint64_t dca_core_channel_layout[] = {

  132.     AV_CH_FRONT_CENTER,                                                     ///< 1, A

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

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

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

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

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

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

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

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

  141. 

  142.     AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_SIDE_LEFT |

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

  144. 

  145.     AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |

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

  147. 

  148.     AV_CH_LAYOUT_STEREO | AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT |

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

  150. 

  151.     AV_CH_FRONT_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |

  152.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_BACK_CENTER   |

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

  154. 

  155.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER   |

  156.     AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |

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

  158. 

  159.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |

  160.     AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |

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

  162. 

  163.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER   |

  164.     AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |

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

  166. };

  167. 

  168. static const int8_t dca_lfe_index[] = {

  169.     1, 2, 2, 2, 2, 3, 2, 3, 2, 3, 2, 3, 1, 3, 2, 3

  170. };

  171. 

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

  173.     { 0, -1, -1, -1, -1, -1, -1, -1, -1},

  174.     { 0,  1, -1, -1, -1, -1, -1, -1, -1},

  175.     { 0,  1, -1, -1, -1, -1, -1, -1, -1},

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

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

  178.     { 2,  0,  1, -1, -1, -1, -1, -1, -1},

  179.     { 0,  1,  3, -1, -1, -1, -1, -1, -1},

  180.     { 2,  0,  1,  4, -1, -1, -1, -1, -1},

  181.     { 0,  1,  3,  4, -1, -1, -1, -1, -1},

  182.     { 2,  0,  1,  4,  5, -1, -1, -1, -1},

  183.     { 3,  4,  0,  1,  5,  6, -1, -1, -1},

  184.     { 2,  0,  1,  4,  5,  6, -1, -1, -1},

  185.     { 0,  6,  4,  5,  2,  3, -1, -1, -1},

  186.     { 4,  2,  5,  0,  1,  6,  7, -1, -1},

  187.     { 5,  6,  0,  1,  7,  3,  8,  4, -1},

  188.     { 4,  2,  5,  0,  1,  6,  8,  7, -1},

  189. };

  190. 

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

  192.     { 0,  2, -1, -1, -1, -1, -1, -1, -1},

  193.     { 0,  1,  3, -1, -1, -1, -1, -1, -1},

  194.     { 0,  1,  3, -1, -1, -1, -1, -1, -1},

  195.     { 0,  1,  3, -1, -1, -1, -1, -1, -1},

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

  197.     { 2,  0,  1,  4, -1, -1, -1, -1, -1},

  198.     { 0,  1,  3,  4, -1, -1, -1, -1, -1},

  199.     { 2,  0,  1,  4,  5, -1, -1, -1, -1},

  200.     { 0,  1,  4,  5,  3, -1, -1, -1, -1},

  201.     { 2,  0,  1,  5,  6,  4, -1, -1, -1},

  202.     { 3,  4,  0,  1,  6,  7,  5, -1, -1},

  203.     { 2,  0,  1,  4,  5,  6,  7, -1, -1},

  204.     { 0,  6,  4,  5,  2,  3,  7, -1, -1},

  205.     { 4,  2,  5,  0,  1,  7,  8,  6, -1},

  206.     { 5,  6,  0,  1,  8,  3,  9,  4,  7},

  207.     { 4,  2,  5,  0,  1,  6,  9,  8,  7},

  208. };

  209. 

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

  211.     { 0, -1, -1, -1, -1, -1, -1, -1, -1},

  212.     { 0,  1, -1, -1, -1, -1, -1, -1, -1},

  213.     { 0,  1, -1, -1, -1, -1, -1, -1, -1},

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

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

  216.     { 2,  0,  1, -1, -1, -1, -1, -1, -1},

  217.     { 0,  1,  2, -1, -1, -1, -1, -1, -1},

  218.     { 2,  0,  1,  3, -1, -1, -1, -1, -1},

  219.     { 0,  1,  2,  3, -1, -1, -1, -1, -1},

  220.     { 2,  0,  1,  3,  4, -1, -1, -1, -1},

  221.     { 2,  3,  0,  1,  4,  5, -1, -1, -1},

  222.     { 2,  0,  1,  3,  4,  5, -1, -1, -1},

  223.     { 0,  5,  3,  4,  1,  2, -1, -1, -1},

  224.     { 3,  2,  4,  0,  1,  5,  6, -1, -1},

  225.     { 4,  5,  0,  1,  6,  2,  7,  3, -1},

  226.     { 3,  2,  4,  0,  1,  5,  7,  6, -1},

  227. };

  228. 

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

  230.     { 0,  1, -1, -1, -1, -1, -1, -1, -1},

  231.     { 0,  1,  2, -1, -1, -1, -1, -1, -1},

  232.     { 0,  1,  2, -1, -1, -1, -1, -1, -1},

  233.     { 0,  1,  2, -1, -1, -1, -1, -1, -1},

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

  235.     { 2,  0,  1,  3, -1, -1, -1, -1, -1},

  236.     { 0,  1,  2,  3, -1, -1, -1, -1, -1},

  237.     { 2,  0,  1,  3,  4, -1, -1, -1, -1},

  238.     { 0,  1,  3,  4,  2, -1, -1, -1, -1},

  239.     { 2,  0,  1,  4,  5,  3, -1, -1, -1},

  240.     { 2,  3,  0,  1,  5,  6,  4, -1, -1},

  241.     { 2,  0,  1,  3,  4,  5,  6, -1, -1},

  242.     { 0,  5,  3,  4,  1,  2,  6, -1, -1},

  243.     { 3,  2,  4,  0,  1,  6,  7,  5, -1},

  244.     { 4,  5,  0,  1,  7,  2,  8,  3,  6},

  245.     { 3,  2,  4,  0,  1,  5,  8,  7,  6},

  246. };

  247. 

  248. #define DCA_DOLBY                  101           /* FIXME */

  249. 

  250. #define DCA_CHANNEL_BITS             6

  251. #define DCA_CHANNEL_MASK          0x3F

  252. 

  253. #define DCA_LFE                   0x80

  254. 

  255. #define HEADER_SIZE                 14

  256. 

  257. #define DCA_MAX_FRAME_SIZE       16384

  258. #define DCA_MAX_EXSS_HEADER_SIZE  4096

  259. 

  260. #define DCA_BUFFER_PADDING_SIZE   1024

  261. 

  262. /** Bit allocation */

  263. typedef struct {

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

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

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

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

  268. } BitAlloc;

  269. 

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

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

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

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

  274. 

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

  276.                                          int idx)

  277. {

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

  279.            ba->offset;

  280. }

  281. 

  282. typedef struct {

  283.     AVCodecContext *avctx;

  284.     AVFrame frame;

  285.     /* Frame header */

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

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

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

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

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

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

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

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

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

  295. 

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

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

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

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

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

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

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

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

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

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

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

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

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

  309.     int copy_history;           ///< copy history

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

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

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

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

  314. 

  315.     /* Primary audio coding header */

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

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

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

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

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

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

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

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

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

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

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

  327. 

  328.     /* Primary audio coding side information */

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

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

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

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

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

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

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

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

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

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

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

  340. 

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

  342. 

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

  344.     int lfe_scale_factor;

  345. 

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

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

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

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

  350.     int hist_index[DCA_PRIM_CHANNELS_MAX];

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

  352. 

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

  354.     float scale_bias;           ///< output scale

  355. 

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

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

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

  359. 

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

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

  362. 

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

  364.     GetBitContext gb;

  365.     /* Current position in DCA frame */

  366.     int current_subframe;

  367.     int current_subsubframe;

  368. 

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

  370. 

  371.     /* XCh extension information */

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

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

  374. 

  375.     /* ExSS header parser */

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

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

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

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

  380. 

  381.     int profile;

  382. 

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

  384.     DSPContext dsp;

  385.     FFTContext imdct;

  386.     SynthFilterContext synth;

  387.     DCADSPContext dcadsp;

  388.     FmtConvertContext fmt_conv;

  389. } DCAContext;

  390. 

  391. static const uint16_t dca_vlc_offs[] = {

  392.         0,   512,   640,   768,  1282,  1794,  2436,  3080,  3770,  4454,  5364,

  393.      5372,  5380,  5388,  5392,  5396,  5412,  5420,  5428,  5460,  5492,  5508,

  394.      5572,  5604,  5668,  5796,  5860,  5892,  6412,  6668,  6796,  7308,  7564,

  395.      7820,  8076,  8620,  9132,  9388,  9910, 10166, 10680, 11196, 11726, 12240,

  396.     12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264,

  397.     18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622,

  398. };

  399. 

  400. static av_cold void dca_init_vlcs(void)

  401. {

  402.     static int vlcs_initialized = 0;

  403.     int i, j, c = 14;

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

  405. 

  406.     if (vlcs_initialized)

  407.         return;

  408. 

  409.     dca_bitalloc_index.offset = 1;

  410.     dca_bitalloc_index.wrap = 2;

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

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

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

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

  415.                  bitalloc_12_bits[i], 1, 1,

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

  417.     }

  418.     dca_scalefactor.offset = -64;

  419.     dca_scalefactor.wrap = 2;

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

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

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

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

  424.                  scales_bits[i], 1, 1,

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

  426.     }

  427.     dca_tmode.offset = 0;

  428.     dca_tmode.wrap = 1;

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

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

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

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

  433.                  tmode_bits[i], 1, 1,

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

  435.     }

  436. 

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

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

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

  440.                 break;

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

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

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

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

  445. 

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

  447.                      bitalloc_sizes[i],

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

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

  450.             c++;

  451.         }

  452.     vlcs_initialized = 1;

  453. }

  454. 

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

  456. {

  457.     while (len--)

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

  459. }

  460. 

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

  462. {

  463.     int i, j;

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

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

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

  467. 

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

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

  470. 

  471.     if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)

  472.         s->prim_channels = DCA_PRIM_CHANNELS_MAX;

  473. 

  474. 

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

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

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

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

  479.     }

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

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

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

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

  484.     }

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

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

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

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

  489. 

  490.     /* Get codebooks quantization indexes */

  491.     if (!base_channel)

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

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

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

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

  496. 

  497.     /* Get scale factor adjustment */

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

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

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

  501. 

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

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

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

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

  506. 

  507.     if (s->crc_present) {

  508.         /* Audio header CRC check */

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

  510.     }

  511. 

  512.     s->current_subframe    = 0;

  513.     s->current_subsubframe = 0;

  514. 

  515. #ifdef TRACE

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

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

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

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

  520.                s->subband_activity[i]);

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

  522.                s->vq_start_subband[i]);

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

  524.                s->joint_intensity[i]);

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

  526.                s->transient_huffman[i]);

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

  528.                s->scalefactor_huffman[i]);

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

  530.                s->bitalloc_huffman[i]);

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

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

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

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

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

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

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

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

  539.     }

  540. #endif

  541. 

  542.     return 0;

  543. }

  544. 

  545. static int dca_parse_frame_header(DCAContext *s)

  546. {

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

  548. 

  549.     /* Sync code */

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

  551. 

  552.     /* Frame header */

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

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

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

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

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

  558.     if (s->frame_size < 95)

  559.         return AVERROR_INVALIDDATA;

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

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

  562.     if (!s->sample_rate)

  563.         return AVERROR_INVALIDDATA;

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

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

  566.     if (!s->bit_rate)

  567.         return AVERROR_INVALIDDATA;

  568. 

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

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

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

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

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

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

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

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

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

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

  579. 

  580.     /* TODO: check CRC */

  581.     if (s->crc_present)

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

  583. 

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

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

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

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

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

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

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

  591. 

  592.     /* FIXME: channels mixing levels */

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

  594.     if (s->lfe)

  595.         s->output |= DCA_LFE;

  596. 

  597. #ifdef TRACE

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

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

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

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

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

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

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

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

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

  607.            s->sample_rate);

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

  609.            s->bit_rate);

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

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

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

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

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

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

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

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

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

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

  620.            s->predictor_history);

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

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

  623.            s->multirate_inter);

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

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

  626.     av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

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

  633. #endif

  634. 

  635.     /* Primary audio coding header */

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

  637. 

  638.     return dca_parse_audio_coding_header(s, 0);

  639. }

  640. 

  641. 

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

  643. {

  644.     if (level < 5) {

  645.         /* huffman encoded */

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

  647.     } else if (level < 8)

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

  649.     return value;

  650. }

  651. 

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

  653. {

  654.     /* Primary audio coding side information */

  655.     int j, k;

  656. 

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

  658.         return AVERROR_INVALIDDATA;

  659. 

  660.     if (!base_channel) {

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

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

  663.     }

  664. 

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

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

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

  668.     }

  669. 

  670.     /* Get prediction codebook */

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

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

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

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

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

  676.             }

  677.         }

  678.     }

  679. 

  680.     /* Bit allocation index */

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

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

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

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

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

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

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

  688.                 av_log(s->avctx, AV_LOG_ERROR,

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

  690.                 return AVERROR_INVALIDDATA;

  691.             } else {

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

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

  694.             }

  695. 

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

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

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

  699.                 return AVERROR_INVALIDDATA;

  700.             }

  701.         }

  702.     }

  703. 

  704.     /* Transition mode */

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

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

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

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

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

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

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

  712.             }

  713.         }

  714.     }

  715. 

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

  717.         return AVERROR_INVALIDDATA;

  718. 

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

  720.         const uint32_t *scale_table;

  721.         unsigned int scale_max;

  722.         int scale_sum;

  723. 

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

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

  726. 

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

  728.             scale_table = scale_factor_quant7;

  729.             scale_max   = 127;

  730.         } else {

  731.             scale_table = scale_factor_quant6;

  732.             scale_max   = 63;

  733.         }

  734. 

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

  736.         scale_sum = 0;

  737. 

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

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

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

  741.                 if (scale_sum > scale_max) {

  742.                     av_log(s->avctx, AV_LOG_ERROR, "scale_sum out of range\n");

  743.                     return AVERROR_INVALIDDATA;

  744.                 }

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

  746.             }

  747. 

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

  749.                 /* Get second scale factor */

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

  751.                 if (scale_sum > scale_max) {

  752.                     av_log(s->avctx, AV_LOG_ERROR, "scale_sum out of range\n");

  753.                     return AVERROR_INVALIDDATA;

  754.                 }

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

  756.             }

  757.         }

  758.     }

  759. 

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

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

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

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

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

  765.     }

  766. 

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

  768.         return AVERROR_INVALIDDATA;

  769. 

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

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

  772.         int source_channel;

  773. 

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

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

  776.             int scale = 0;

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

  778. 

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

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

  781. 

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

  783.                 scale = get_scale(&s->gb, s->joint_huff[j], 0);

  784.                 scale += 64;    /* bias */

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

  786.             }

  787. 

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

  789.                 av_log(s->avctx, AV_LOG_DEBUG,

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

  791.                 s->debug_flag |= 0x02;

  792.             }

  793.         }

  794.     }

  795. 

  796.     /* Stereo downmix coefficients */

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

  798.         if (s->downmix) {

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

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

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

  802.             }

  803.         } else {

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

  805.             if (am < 16) {

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

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

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

  809.             }

  810.             } else {

  811.                 av_log(s->avctx, AV_LOG_WARNING, "amode > 15 default downmix_coef unsupported\n");

  812.             }

  813.         }

  814.     }

  815. 

  816.     /* Dynamic range coefficient */

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

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

  819. 

  820.     /* Side information CRC check word */

  821.     if (s->crc_present) {

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

  823.     }

  824. 

  825.     /*

  826.      * Primary audio data arrays

  827.      */

  828. 

  829.     /* VQ encoded high frequency subbands */

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

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

  832.             /* 1 vector -> 32 samples */

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

  834. 

  835.     /* Low frequency effect data */

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

  837.         int quant7;

  838.         /* LFE samples */

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

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

  841.         float lfe_scale;

  842. 

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

  844.             /* Signed 8 bits int */

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

  846.         }

  847. 

  848.         /* Scale factor index */

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

  850.         if (quant7 > 127) {

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

  852.             return AVERROR_INVALIDDATA;

  853.         }

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

  855. 

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

  857.         lfe_scale = 0.035 * s->lfe_scale_factor;

  858. 

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

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

  861.     }

  862. 

  863. #ifdef TRACE

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

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

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

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

  868. 

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

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

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

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

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

  874.     }

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

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

  877.             av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

  883.     }

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

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

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

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

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

  889.     }

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

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

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

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

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

  895.     }

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

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

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

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

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

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

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

  903.         }

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

  905.     }

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

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

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

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

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

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

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

  913.         }

  914.     }

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

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

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

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

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

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

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

  922.         }

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

  924.     }

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

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

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

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

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

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

  931. 

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

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

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

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

  936.     }

  937. #endif

  938. 

  939.     return 0;

  940. }

  941. 

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

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

  944.                             float scale)

  945. {

  946.     const float *prCoeff;

  947.     int i;

  948. 

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

  950.     int subindex;

  951. 

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

  953. 

  954.     /* Select filter */

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

  956.         prCoeff = fir_32bands_nonperfect;

  957.     else                        /* Perfect reconstruction */

  958.         prCoeff = fir_32bands_perfect;

  959. 

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

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

  962. 

  963.     /* Reconstructed channel sample index */

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

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

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

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

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

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

  970.         }

  971. 

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

  973.                                     s->subband_fir_hist[chans],

  974.                                     &s->hist_index[chans],

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

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

  977.         samples_out += 32;

  978.     }

  979. }

  980. 

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

  982.                                   int num_deci_sample, float *samples_in,

  983.                                   float *samples_out, float scale)

  984. {

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

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

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

  988.      *   from last subframe as history.

  989.      *

  990.      * samples_out: An array holding interpolated samples

  991.      */

  992. 

  993.     int decifactor;

  994.     const float *prCoeff;

  995.     int deciindex;

  996. 

  997.     /* Select decimation filter */

  998.     if (decimation_select == 1) {

  999.         decifactor = 64;

  1000.         prCoeff = lfe_fir_128;

  1001.     } else {

  1002.         decifactor = 32;

  1003.         prCoeff = lfe_fir_64;

  1004.     }

  1005.     /* Interpolation */

  1006.     for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {

  1007.         s->dcadsp.lfe_fir(samples_out, samples_in, prCoeff, decifactor, scale);

  1008.         samples_in++;

  1009.         samples_out += 2 * decifactor;

  1010.     }

  1011. }

  1012. 

  1013. /* downmixing routines */

  1014. #define MIX_REAR1(samples, si1, rs, coef)           \

  1015.     samples[i]     += samples[si1] * coef[rs][0];   \

  1016.     samples[i+256] += samples[si1] * coef[rs][1];

  1017. 

  1018. #define MIX_REAR2(samples, si1, si2, rs, coef)                                     \

  1019.     samples[i]     += samples[si1] * coef[rs][0] + samples[si2] * coef[rs + 1][0]; \

  1020.     samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs + 1][1];

  1021. 

  1022. #define MIX_FRONT3(samples, coef)                                      \

  1023.     t = samples[i + c];                                                \

  1024.     u = samples[i + l];                                                \

  1025.     v = samples[i + r];                                                \

  1026.     samples[i]     = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \

  1027.     samples[i+256] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];

  1028. 

  1029. #define DOWNMIX_TO_STEREO(op1, op2)             \

  1030.     for (i = 0; i < 256; i++) {                 \

  1031.         op1                                     \

  1032.         op2                                     \

  1033.     }

  1034. 

  1035. static void dca_downmix(float *samples, int srcfmt,

  1036.                         int downmix_coef[DCA_PRIM_CHANNELS_MAX][2],

  1037.                         const int8_t *channel_mapping)

  1038. {

  1039.     int c, l, r, sl, sr, s;

  1040.     int i;

  1041.     float t, u, v;

  1042.     float coef[DCA_PRIM_CHANNELS_MAX][2];

  1043. 

  1044.     for (i = 0; i < DCA_PRIM_CHANNELS_MAX; i++) {

  1045.         coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]];

  1046.         coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]];

  1047.     }

  1048. 

  1049.     switch (srcfmt) {

  1050.     case DCA_MONO:

  1051.     case DCA_CHANNEL:

  1052.     case DCA_STEREO_TOTAL:

  1053.     case DCA_STEREO_SUMDIFF:

  1054.     case DCA_4F2R:

  1055.         av_log(NULL, 0, "Not implemented!\n");

  1056.         break;

  1057.     case DCA_STEREO:

  1058.         break;

  1059.     case DCA_3F:

  1060.         c = channel_mapping[0] * 256;

  1061.         l = channel_mapping[1] * 256;

  1062.         r = channel_mapping[2] * 256;

  1063.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );

  1064.         break;

  1065.     case DCA_2F1R:

  1066.         s = channel_mapping[2] * 256;

  1067.         DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + s, 2, coef), );

  1068.         break;

  1069.     case DCA_3F1R:

  1070.         c = channel_mapping[0] * 256;

  1071.         l = channel_mapping[1] * 256;

  1072.         r = channel_mapping[2] * 256;

  1073.         s = channel_mapping[3] * 256;

  1074.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1075.                           MIX_REAR1(samples, i + s, 3, coef));

  1076.         break;

  1077.     case DCA_2F2R:

  1078.         sl = channel_mapping[2] * 256;

  1079.         sr = channel_mapping[3] * 256;

  1080.         DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + sl, i + sr, 2, coef), );

  1081.         break;

  1082.     case DCA_3F2R:

  1083.         c  = channel_mapping[0] * 256;

  1084.         l  = channel_mapping[1] * 256;

  1085.         r  = channel_mapping[2] * 256;

  1086.         sl = channel_mapping[3] * 256;

  1087.         sr = channel_mapping[4] * 256;

  1088.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1089.                           MIX_REAR2(samples, i + sl, i + sr, 3, coef));

  1090.         break;

  1091.     }

  1092. }

  1093. 

  1094. 

  1095. #ifndef decode_blockcodes

  1096. /* Very compact version of the block code decoder that does not use table

  1097.  * look-up but is slightly slower */

  1098. static int decode_blockcode(int code, int levels, int *values)

  1099. {

  1100.     int i;

  1101.     int offset = (levels - 1) >> 1;

  1102. 

  1103.     for (i = 0; i < 4; i++) {

  1104.         int div = FASTDIV(code, levels);

  1105.         values[i] = code - offset - div * levels;

  1106.         code = div;

  1107.     }

  1108. 

  1109.     return code;

  1110. }

  1111. 

  1112. static int decode_blockcodes(int code1, int code2, int levels, int *values)

  1113. {

  1114.     return decode_blockcode(code1, levels, values) |

  1115.            decode_blockcode(code2, levels, values + 4);

  1116. }

  1117. #endif

  1118. 

  1119. static const uint8_t abits_sizes[7]  = { 7, 10, 12, 13, 15, 17, 19 };

  1120. static const uint8_t abits_levels[7] = { 3,  5,  7,  9, 13, 17, 25 };

  1121. 

  1122. #ifndef int8x8_fmul_int32

  1123. static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)

  1124. {

  1125.     float fscale = scale / 16.0;

  1126.     int i;

  1127.     for (i = 0; i < 8; i++)

  1128.         dst[i] = src[i] * fscale;

  1129. }

  1130. #endif

  1131. 

  1132. static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)

  1133. {

  1134.     int k, l;

  1135.     int subsubframe = s->current_subsubframe;

  1136. 

  1137.     const float *quant_step_table;

  1138. 

  1139.     /* FIXME */

  1140.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1141.     LOCAL_ALIGNED_16(int, block, [8]);

  1142. 

  1143.     /*

  1144.      * Audio data

  1145.      */

  1146. 

  1147.     /* Select quantization step size table */

  1148.     if (s->bit_rate_index == 0x1f)

  1149.         quant_step_table = lossless_quant_d;

  1150.     else

  1151.         quant_step_table = lossy_quant_d;

  1152. 

  1153.     for (k = base_channel; k < s->prim_channels; k++) {

  1154.         if (get_bits_left(&s->gb) < 0)

  1155.             return AVERROR_INVALIDDATA;

  1156. 

  1157.         for (l = 0; l < s->vq_start_subband[k]; l++) {

  1158.             int m;

  1159. 

  1160.             /* Select the mid-tread linear quantizer */

  1161.             int abits = s->bitalloc[k][l];

  1162. 

  1163.             float quant_step_size = quant_step_table[abits];

  1164. 

  1165.             /*

  1166.              * Determine quantization index code book and its type

  1167.              */

  1168. 

  1169.             /* Select quantization index code book */

  1170.             int sel = s->quant_index_huffman[k][abits];

  1171. 

  1172.             /*

  1173.              * Extract bits from the bit stream

  1174.              */

  1175.             if (!abits) {

  1176.                 memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));

  1177.             } else {

  1178.                 /* Deal with transients */

  1179.                 int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];

  1180.                 float rscale = quant_step_size * s->scale_factor[k][l][sfi] *

  1181.                                s->scalefactor_adj[k][sel];

  1182. 

  1183.                 if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {

  1184.                     if (abits <= 7) {

  1185.                         /* Block code */

  1186.                         int block_code1, block_code2, size, levels, err;

  1187. 

  1188.                         size   = abits_sizes[abits - 1];

  1189.                         levels = abits_levels[abits - 1];

  1190. 

  1191.                         block_code1 = get_bits(&s->gb, size);

  1192.                         block_code2 = get_bits(&s->gb, size);

  1193.                         err = decode_blockcodes(block_code1, block_code2,

  1194.                                                 levels, block);

  1195.                         if (err) {

  1196.                             av_log(s->avctx, AV_LOG_ERROR,

  1197.                                    "ERROR: block code look-up failed\n");

  1198.                             return AVERROR_INVALIDDATA;

  1199.                         }

  1200.                     } else {

  1201.                         /* no coding */

  1202.                         for (m = 0; m < 8; m++)

  1203.                             block[m] = get_sbits(&s->gb, abits - 3);

  1204.                     }

  1205.                 } else {

  1206.                     /* Huffman coded */

  1207.                     for (m = 0; m < 8; m++)

  1208.                         block[m] = get_bitalloc(&s->gb,

  1209.                                                 &dca_smpl_bitalloc[abits], sel);

  1210.                 }

  1211. 

  1212.                 s->fmt_conv.int32_to_float_fmul_scalar(subband_samples[k][l],

  1213.                                                        block, rscale, 8);

  1214.             }

  1215. 

  1216.             /*

  1217.              * Inverse ADPCM if in prediction mode

  1218.              */

  1219.             if (s->prediction_mode[k][l]) {

  1220.                 int n;

  1221.                 for (m = 0; m < 8; m++) {

  1222.                     for (n = 1; n <= 4; n++)

  1223.                         if (m >= n)

  1224.                             subband_samples[k][l][m] +=

  1225.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1226.                                  subband_samples[k][l][m - n] / 8192);

  1227.                         else if (s->predictor_history)

  1228.                             subband_samples[k][l][m] +=

  1229.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1230.                                  s->subband_samples_hist[k][l][m - n + 4] / 8192);

  1231.                 }

  1232.             }

  1233.         }

  1234. 

  1235.         /*

  1236.          * Decode VQ encoded high frequencies

  1237.          */

  1238.         for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {

  1239.             /* 1 vector -> 32 samples but we only need the 8 samples

  1240.              * for this subsubframe. */

  1241.             int hfvq = s->high_freq_vq[k][l];

  1242. 

  1243.             if (!s->debug_flag & 0x01) {

  1244.                 av_log(s->avctx, AV_LOG_DEBUG,

  1245.                        "Stream with high frequencies VQ coding\n");

  1246.                 s->debug_flag |= 0x01;

  1247.             }

  1248. 

  1249.             int8x8_fmul_int32(subband_samples[k][l],

  1250.                               &high_freq_vq[hfvq][subsubframe * 8],

  1251.                               s->scale_factor[k][l][0]);

  1252.         }

  1253.     }

  1254. 

  1255.     /* Check for DSYNC after subsubframe */

  1256.     if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {

  1257.         if (0xFFFF == get_bits(&s->gb, 16)) {   /* 0xFFFF */

  1258. #ifdef TRACE

  1259.             av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");

  1260. #endif

  1261.         } else {

  1262.             av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");

  1263.         }

  1264.     }

  1265. 

  1266.     /* Backup predictor history for adpcm */

  1267.     for (k = base_channel; k < s->prim_channels; k++)

  1268.         for (l = 0; l < s->vq_start_subband[k]; l++)

  1269.             memcpy(s->subband_samples_hist[k][l],

  1270.                    &subband_samples[k][l][4],

  1271.                    4 * sizeof(subband_samples[0][0][0]));

  1272. 

  1273.     return 0;

  1274. }

  1275. 

  1276. static int dca_filter_channels(DCAContext *s, int block_index)

  1277. {

  1278.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1279.     int k;

  1280. 

  1281.     /* 32 subbands QMF */

  1282.     for (k = 0; k < s->prim_channels; k++) {

  1283. /*        static float pcm_to_double[8] = { 32768.0, 32768.0, 524288.0, 524288.0,

  1284.                                             0, 8388608.0, 8388608.0 };*/

  1285.         qmf_32_subbands(s, k, subband_samples[k],

  1286.                         &s->samples[256 * s->channel_order_tab[k]],

  1287.                         M_SQRT1_2 * s->scale_bias /* pcm_to_double[s->source_pcm_res] */);

  1288.     }

  1289. 

  1290.     /* Down mixing */

  1291.     if (s->avctx->request_channels == 2 && s->prim_channels > 2) {

  1292.         dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab);

  1293.     }

  1294. 

  1295.     /* Generate LFE samples for this subsubframe FIXME!!! */

  1296.     if (s->output & DCA_LFE) {

  1297.         lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,

  1298.                               s->lfe_data + 2 * s->lfe * (block_index + 4),

  1299.                               &s->samples[256 * dca_lfe_index[s->amode]],

  1300.                               (1.0 / 256.0) * s->scale_bias);

  1301.         /* Outputs 20bits pcm samples */

  1302.     }

  1303. 

  1304.     return 0;

  1305. }

  1306. 

  1307. 

  1308. static int dca_subframe_footer(DCAContext *s, int base_channel)

  1309. {

  1310.     int aux_data_count = 0, i;

  1311. 

  1312.     /*

  1313.      * Unpack optional information

  1314.      */

  1315. 

  1316.     /* presumably optional information only appears in the core? */

  1317.     if (!base_channel) {

  1318.         if (s->timestamp)

  1319.             skip_bits_long(&s->gb, 32);

  1320. 

  1321.         if (s->aux_data)

  1322.             aux_data_count = get_bits(&s->gb, 6);

  1323. 

  1324.         for (i = 0; i < aux_data_count; i++)

  1325.             get_bits(&s->gb, 8);

  1326. 

  1327.         if (s->crc_present && (s->downmix || s->dynrange))

  1328.             get_bits(&s->gb, 16);

  1329.     }

  1330. 

  1331.     return 0;

  1332. }

  1333. 

  1334. /**

  1335.  * Decode a dca frame block

  1336.  *

  1337.  * @param s     pointer to the DCAContext

  1338.  */

  1339. 

  1340. static int dca_decode_block(DCAContext *s, int base_channel, int block_index)

  1341. {

  1342.     int ret;

  1343. 

  1344.     /* Sanity check */

  1345.     if (s->current_subframe >= s->subframes) {

  1346.         av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",

  1347.                s->current_subframe, s->subframes);

  1348.         return AVERROR_INVALIDDATA;

  1349.     }

  1350. 

  1351.     if (!s->current_subsubframe) {

  1352. #ifdef TRACE

  1353.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");

  1354. #endif

  1355.         /* Read subframe header */

  1356.         if ((ret = dca_subframe_header(s, base_channel, block_index)))

  1357.             return ret;

  1358.     }

  1359. 

  1360.     /* Read subsubframe */

  1361. #ifdef TRACE

  1362.     av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");

  1363. #endif

  1364.     if ((ret = dca_subsubframe(s, base_channel, block_index)))

  1365.         return ret;

  1366. 

  1367.     /* Update state */

  1368.     s->current_subsubframe++;

  1369.     if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {

  1370.         s->current_subsubframe = 0;

  1371.         s->current_subframe++;

  1372.     }

  1373.     if (s->current_subframe >= s->subframes) {

  1374. #ifdef TRACE

  1375.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");

  1376. #endif

  1377.         /* Read subframe footer */

  1378.         if ((ret = dca_subframe_footer(s, base_channel)))

  1379.             return ret;

  1380.     }

  1381. 

  1382.     return 0;

  1383. }

  1384. 

  1385. /**

  1386.  * Return the number of channels in an ExSS speaker mask (HD)

  1387.  */

  1388. static int dca_exss_mask2count(int mask)

  1389. {

  1390.     /* count bits that mean speaker pairs twice */

  1391.     return av_popcount(mask) +

  1392.            av_popcount(mask & (DCA_EXSS_CENTER_LEFT_RIGHT      |

  1393.                                DCA_EXSS_FRONT_LEFT_RIGHT       |

  1394.                                DCA_EXSS_FRONT_HIGH_LEFT_RIGHT  |

  1395.                                DCA_EXSS_WIDE_LEFT_RIGHT        |

  1396.                                DCA_EXSS_SIDE_LEFT_RIGHT        |

  1397.                                DCA_EXSS_SIDE_HIGH_LEFT_RIGHT   |

  1398.                                DCA_EXSS_SIDE_REAR_LEFT_RIGHT   |

  1399.                                DCA_EXSS_REAR_LEFT_RIGHT        |

  1400.                                DCA_EXSS_REAR_HIGH_LEFT_RIGHT));

  1401. }

  1402. 

  1403. /**

  1404.  * Skip mixing coefficients of a single mix out configuration (HD)

  1405.  */

  1406. static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch)

  1407. {

  1408.     int i;

  1409. 

  1410.     for (i = 0; i < channels; i++) {

  1411.         int mix_map_mask = get_bits(gb, out_ch);

  1412.         int num_coeffs = av_popcount(mix_map_mask);

  1413.         skip_bits_long(gb, num_coeffs * 6);

  1414.     }

  1415. }

  1416. 

  1417. /**

  1418.  * Parse extension substream asset header (HD)

  1419.  */

  1420. static int dca_exss_parse_asset_header(DCAContext *s)

  1421. {

  1422.     int header_pos = get_bits_count(&s->gb);

  1423.     int header_size;

  1424.     int channels = 0;

  1425.     int embedded_stereo = 0;

  1426.     int embedded_6ch    = 0;

  1427.     int drc_code_present;

  1428.     int av_uninit(extensions_mask);

  1429.     int i, j;

  1430. 

  1431.     if (get_bits_left(&s->gb) < 16)

  1432.         return -1;

  1433. 

  1434.     /* We will parse just enough to get to the extensions bitmask with which

  1435.      * we can set the profile value. */

  1436. 

  1437.     header_size = get_bits(&s->gb, 9) + 1;

  1438.     skip_bits(&s->gb, 3); // asset index

  1439. 

  1440.     if (s->static_fields) {

  1441.         if (get_bits1(&s->gb))

  1442.             skip_bits(&s->gb, 4); // asset type descriptor

  1443.         if (get_bits1(&s->gb))

  1444.             skip_bits_long(&s->gb, 24); // language descriptor

  1445. 

  1446.         if (get_bits1(&s->gb)) {

  1447.             /* How can one fit 1024 bytes of text here if the maximum value

  1448.              * for the asset header size field above was 512 bytes? */

  1449.             int text_length = get_bits(&s->gb, 10) + 1;

  1450.             if (get_bits_left(&s->gb) < text_length * 8)

  1451.                 return -1;

  1452.             skip_bits_long(&s->gb, text_length * 8); // info text

  1453.         }

  1454. 

  1455.         skip_bits(&s->gb, 5); // bit resolution - 1

  1456.         skip_bits(&s->gb, 4); // max sample rate code

  1457.         channels = get_bits(&s->gb, 8) + 1;

  1458. 

  1459.         if (get_bits1(&s->gb)) { // 1-to-1 channels to speakers

  1460.             int spkr_remap_sets;

  1461.             int spkr_mask_size = 16;

  1462.             int num_spkrs[7];

  1463. 

  1464.             if (channels > 2)

  1465.                 embedded_stereo = get_bits1(&s->gb);

  1466.             if (channels > 6)

  1467.                 embedded_6ch = get_bits1(&s->gb);

  1468. 

  1469.             if (get_bits1(&s->gb)) {

  1470.                 spkr_mask_size = (get_bits(&s->gb, 2) + 1) << 2;

  1471.                 skip_bits(&s->gb, spkr_mask_size); // spkr activity mask

  1472.             }

  1473. 

  1474.             spkr_remap_sets = get_bits(&s->gb, 3);

  1475. 

  1476.             for (i = 0; i < spkr_remap_sets; i++) {

  1477.                 /* std layout mask for each remap set */

  1478.                 num_spkrs[i] = dca_exss_mask2count(get_bits(&s->gb, spkr_mask_size));

  1479.             }

  1480. 

  1481.             for (i = 0; i < spkr_remap_sets; i++) {

  1482.                 int num_dec_ch_remaps = get_bits(&s->gb, 5) + 1;

  1483.                 if (get_bits_left(&s->gb) < 0)

  1484.                     return -1;

  1485. 

  1486.                 for (j = 0; j < num_spkrs[i]; j++) {

  1487.                     int remap_dec_ch_mask = get_bits_long(&s->gb, num_dec_ch_remaps);

  1488.                     int num_dec_ch = av_popcount(remap_dec_ch_mask);

  1489.                     skip_bits_long(&s->gb, num_dec_ch * 5); // remap codes

  1490.                 }

  1491.             }

  1492. 

  1493.         } else {

  1494.             skip_bits(&s->gb, 3); // representation type

  1495.         }

  1496.     }

  1497. 

  1498.     drc_code_present = get_bits1(&s->gb);

  1499.     if (drc_code_present)

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

  1501. 

  1502.     if (get_bits1(&s->gb))

  1503.         skip_bits(&s->gb, 5); // dialog normalization code

  1504. 

  1505.     if (drc_code_present && embedded_stereo)

  1506.         get_bits(&s->gb, 8); // drc stereo code

  1507. 

  1508.     if (s->mix_metadata && get_bits1(&s->gb)) {

  1509.         skip_bits(&s->gb, 1); // external mix

  1510.         skip_bits(&s->gb, 6); // post mix gain code

  1511. 

  1512.         if (get_bits(&s->gb, 2) != 3) // mixer drc code

  1513.             skip_bits(&s->gb, 3); // drc limit

  1514.         else

  1515.             skip_bits(&s->gb, 8); // custom drc code

  1516. 

  1517.         if (get_bits1(&s->gb)) // channel specific scaling

  1518.             for (i = 0; i < s->num_mix_configs; i++)

  1519.                 skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6); // scale codes

  1520.         else

  1521.             skip_bits_long(&s->gb, s->num_mix_configs * 6); // scale codes

  1522. 

  1523.         for (i = 0; i < s->num_mix_configs; i++) {

  1524.             if (get_bits_left(&s->gb) < 0)

  1525.                 return -1;

  1526.             dca_exss_skip_mix_coeffs(&s->gb, channels, s->mix_config_num_ch[i]);

  1527.             if (embedded_6ch)

  1528.                 dca_exss_skip_mix_coeffs(&s->gb, 6, s->mix_config_num_ch[i]);

  1529.             if (embedded_stereo)

  1530.                 dca_exss_skip_mix_coeffs(&s->gb, 2, s->mix_config_num_ch[i]);

  1531.         }

  1532.     }

  1533. 

  1534.     switch (get_bits(&s->gb, 2)) {

  1535.     case 0: extensions_mask = get_bits(&s->gb, 12); break;

  1536.     case 1: extensions_mask = DCA_EXT_EXSS_XLL;     break;

  1537.     case 2: extensions_mask = DCA_EXT_EXSS_LBR;     break;

  1538.     case 3: extensions_mask = 0; /* aux coding */   break;

  1539.     }

  1540. 

  1541.     /* not parsed further, we were only interested in the extensions mask */

  1542. 

  1543.     if (get_bits_left(&s->gb) < 0)

  1544.         return -1;

  1545. 

  1546.     if (get_bits_count(&s->gb) - header_pos > header_size * 8) {

  1547.         av_log(s->avctx, AV_LOG_WARNING, "Asset header size mismatch.\n");

  1548.         return -1;

  1549.     }

  1550.     skip_bits_long(&s->gb, header_pos + header_size * 8 - get_bits_count(&s->gb));

  1551. 

  1552.     if (extensions_mask & DCA_EXT_EXSS_XLL)

  1553.         s->profile = FF_PROFILE_DTS_HD_MA;

  1554.     else if (extensions_mask & (DCA_EXT_EXSS_XBR | DCA_EXT_EXSS_X96 |

  1555.                                 DCA_EXT_EXSS_XXCH))

  1556.         s->profile = FF_PROFILE_DTS_HD_HRA;

  1557. 

  1558.     if (!(extensions_mask & DCA_EXT_CORE))

  1559.         av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n");

  1560.     if ((extensions_mask & DCA_CORE_EXTS) != s->core_ext_mask)

  1561.         av_log(s->avctx, AV_LOG_WARNING,

  1562.                "DTS extensions detection mismatch (%d, %d)\n",

  1563.                extensions_mask & DCA_CORE_EXTS, s->core_ext_mask);

  1564. 

  1565.     return 0;

  1566. }

  1567. 

  1568. /**

  1569.  * Parse extension substream header (HD)

  1570.  */

  1571. static void dca_exss_parse_header(DCAContext *s)

  1572. {

  1573.     int ss_index;

  1574.     int blownup;

  1575.     int num_audiop = 1;

  1576.     int num_assets = 1;

  1577.     int active_ss_mask[8];

  1578.     int i, j;

  1579. 

  1580.     if (get_bits_left(&s->gb) < 52)

  1581.         return;

  1582. 

  1583.     skip_bits(&s->gb, 8); // user data

  1584.     ss_index = get_bits(&s->gb, 2);

  1585. 

  1586.     blownup = get_bits1(&s->gb);

  1587.     skip_bits(&s->gb,  8 + 4 * blownup); // header_size

  1588.     skip_bits(&s->gb, 16 + 4 * blownup); // hd_size

  1589. 

  1590.     s->static_fields = get_bits1(&s->gb);

  1591.     if (s->static_fields) {

  1592.         skip_bits(&s->gb, 2); // reference clock code

  1593.         skip_bits(&s->gb, 3); // frame duration code

  1594. 

  1595.         if (get_bits1(&s->gb))

  1596.             skip_bits_long(&s->gb, 36); // timestamp

  1597. 

  1598.         /* a single stream can contain multiple audio assets that can be

  1599.          * combined to form multiple audio presentations */

  1600. 

  1601.         num_audiop = get_bits(&s->gb, 3) + 1;

  1602.         if (num_audiop > 1) {

  1603.             av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio presentations.");

  1604.             /* ignore such streams for now */

  1605.             return;

  1606.         }

  1607. 

  1608.         num_assets = get_bits(&s->gb, 3) + 1;

  1609.         if (num_assets > 1) {

  1610.             av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio assets.");

  1611.             /* ignore such streams for now */

  1612.             return;

  1613.         }

  1614. 

  1615.         for (i = 0; i < num_audiop; i++)

  1616.             active_ss_mask[i] = get_bits(&s->gb, ss_index + 1);

  1617. 

  1618.         for (i = 0; i < num_audiop; i++)

  1619.             for (j = 0; j <= ss_index; j++)

  1620.                 if (active_ss_mask[i] & (1 << j))

  1621.                     skip_bits(&s->gb, 8); // active asset mask

  1622. 

  1623.         s->mix_metadata = get_bits1(&s->gb);

  1624.         if (s->mix_metadata) {

  1625.             int mix_out_mask_size;

  1626. 

  1627.             skip_bits(&s->gb, 2); // adjustment level

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

  1629.             s->num_mix_configs =  get_bits(&s->gb, 2) + 1;

  1630. 

  1631.             for (i = 0; i < s->num_mix_configs; i++) {

  1632.                 int mix_out_mask        = get_bits(&s->gb, mix_out_mask_size);

  1633.                 s->mix_config_num_ch[i] = dca_exss_mask2count(mix_out_mask);

  1634.             }

  1635.         }

  1636.     }

  1637. 

  1638.     for (i = 0; i < num_assets; i++)

  1639.         skip_bits_long(&s->gb, 16 + 4 * blownup);  // asset size

  1640. 

  1641.     for (i = 0; i < num_assets; i++) {

  1642.         if (dca_exss_parse_asset_header(s))

  1643.             return;

  1644.     }

  1645. 

  1646.     /* not parsed further, we were only interested in the extensions mask

  1647.      * from the asset header */

  1648. }

  1649. 

  1650. /**

  1651.  * Main frame decoding function

  1652.  * FIXME add arguments

  1653.  */

  1654. static int dca_decode_frame(AVCodecContext *avctx, void *data,

  1655.                             int *got_frame_ptr, AVPacket *avpkt)

  1656. {

  1657.     const uint8_t *buf = avpkt->data;

  1658.     int buf_size = avpkt->size;

  1659. 

  1660.     int lfe_samples;

  1661.     int num_core_channels = 0;

  1662.     int i, ret;

  1663.     float   *samples_flt;

  1664.     int16_t *samples_s16;

  1665.     DCAContext *s = avctx->priv_data;

  1666.     int channels;

  1667.     int core_ss_end;

  1668. 

  1669. 

  1670.     s->xch_present = 0;

  1671. 

  1672.     s->dca_buffer_size = ff_dca_convert_bitstream(buf, buf_size, s->dca_buffer,

  1673.                                                   DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);

  1674.     if (s->dca_buffer_size == AVERROR_INVALIDDATA) {

  1675.         av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");

  1676.         return AVERROR_INVALIDDATA;

  1677.     }

  1678. 

  1679.     init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);

  1680.     if ((ret = dca_parse_frame_header(s)) < 0) {

  1681.         //seems like the frame is corrupt, try with the next one

  1682.         return ret;

  1683.     }

  1684.     //set AVCodec values with parsed data

  1685.     avctx->sample_rate = s->sample_rate;

  1686.     avctx->bit_rate    = s->bit_rate;

  1687. 

  1688.     s->profile = FF_PROFILE_DTS;

  1689. 

  1690.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1691.         if ((ret = dca_decode_block(s, 0, i))) {

  1692.             av_log(avctx, AV_LOG_ERROR, "error decoding block\n");

  1693.             return ret;

  1694.         }

  1695.     }

  1696. 

  1697.     /* record number of core channels incase less than max channels are requested */

  1698.     num_core_channels = s->prim_channels;

  1699. 

  1700.     if (s->ext_coding)

  1701.         s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];

  1702.     else

  1703.         s->core_ext_mask = 0;

  1704. 

  1705.     core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;

  1706. 

  1707.     /* only scan for extensions if ext_descr was unknown or indicated a

  1708.      * supported XCh extension */

  1709.     if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {

  1710. 

  1711.         /* if ext_descr was unknown, clear s->core_ext_mask so that the

  1712.          * extensions scan can fill it up */

  1713.         s->core_ext_mask = FFMAX(s->core_ext_mask, 0);

  1714. 

  1715.         /* extensions start at 32-bit boundaries into bitstream */

  1716.         skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1717. 

  1718.         while (core_ss_end - get_bits_count(&s->gb) >= 32) {

  1719.             uint32_t bits = get_bits_long(&s->gb, 32);

  1720. 

  1721.             switch (bits) {

  1722.             case 0x5a5a5a5a: {

  1723.                 int ext_amode, xch_fsize;

  1724. 

  1725.                 s->xch_base_channel = s->prim_channels;

  1726. 

  1727.                 /* validate sync word using XCHFSIZE field */

  1728.                 xch_fsize = show_bits(&s->gb, 10);

  1729.                 if ((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&

  1730.                     (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))

  1731.                     continue;

  1732. 

  1733.                 /* skip length-to-end-of-frame field for the moment */

  1734.                 skip_bits(&s->gb, 10);

  1735. 

  1736.                 s->core_ext_mask |= DCA_EXT_XCH;

  1737. 

  1738.                 /* extension amode(number of channels in extension) should be 1 */

  1739.                 /* AFAIK XCh is not used for more channels */

  1740.                 if ((ext_amode = get_bits(&s->gb, 4)) != 1) {

  1741.                     av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"

  1742.                            " supported!\n", ext_amode);

  1743.                     continue;

  1744.                 }

  1745. 

  1746.                 /* much like core primary audio coding header */

  1747.                 dca_parse_audio_coding_header(s, s->xch_base_channel);

  1748. 

  1749.                 for (i = 0; i < (s->sample_blocks / 8); i++)

  1750.                     if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {

  1751.                         av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");

  1752.                         continue;

  1753.                     }

  1754. 

  1755.                 s->xch_present = 1;

  1756.                 break;

  1757.             }

  1758.             case 0x47004a03:

  1759.                 /* XXCh: extended channels */

  1760.                 /* usually found either in core or HD part in DTS-HD HRA streams,

  1761.                  * but not in DTS-ES which contains XCh extensions instead */

  1762.                 s->core_ext_mask |= DCA_EXT_XXCH;

  1763.                 break;

  1764. 

  1765.             case 0x1d95f262: {

  1766.                 int fsize96 = show_bits(&s->gb, 12) + 1;

  1767.                 if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)

  1768.                     continue;

  1769. 

  1770.                 av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n",

  1771.                        get_bits_count(&s->gb));

  1772.                 skip_bits(&s->gb, 12);

  1773.                 av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);

  1774.                 av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));

  1775. 

  1776.                 s->core_ext_mask |= DCA_EXT_X96;

  1777.                 break;

  1778.             }

  1779.             }

  1780. 

  1781.             skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1782.         }

  1783.     } else {

  1784.         /* no supported extensions, skip the rest of the core substream */

  1785.         skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));

  1786.     }

  1787. 

  1788.     if (s->core_ext_mask & DCA_EXT_X96)

  1789.         s->profile = FF_PROFILE_DTS_96_24;

  1790.     else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))

  1791.         s->profile = FF_PROFILE_DTS_ES;

  1792. 

  1793.     /* check for ExSS (HD part) */

  1794.     if (s->dca_buffer_size - s->frame_size > 32 &&

  1795.         get_bits_long(&s->gb, 32) == DCA_HD_MARKER)

  1796.         dca_exss_parse_header(s);

  1797. 

  1798.     avctx->profile = s->profile;

  1799. 

  1800.     channels = s->prim_channels + !!s->lfe;

  1801. 

  1802.     if (s->amode < 16) {

  1803.         avctx->channel_layout = dca_core_channel_layout[s->amode];

  1804. 

  1805.         if (s->xch_present && (!avctx->request_channels ||

  1806.                                avctx->request_channels > num_core_channels + !!s->lfe)) {

  1807.             avctx->channel_layout |= AV_CH_BACK_CENTER;

  1808.             if (s->lfe) {

  1809.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

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

  1811.             } else {

  1812.                 s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];

  1813.             }

  1814.         } else {

  1815.             channels = num_core_channels + !!s->lfe;

  1816.             s->xch_present = 0; /* disable further xch processing */

  1817.             if (s->lfe) {

  1818.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

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

  1820.             } else

  1821.                 s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];

  1822.         }

  1823. 

  1824.         if (channels > !!s->lfe &&

  1825.             s->channel_order_tab[channels - 1 - !!s->lfe] < 0)

  1826.             return AVERROR_INVALIDDATA;

  1827. 

  1828.         if (avctx->request_channels == 2 && s->prim_channels > 2) {

  1829.             channels = 2;

  1830.             s->output = DCA_STEREO;

  1831.             avctx->channel_layout = AV_CH_LAYOUT_STEREO;

  1832.         }

  1833.         else if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) {

  1834.             static const int8_t dca_channel_order_native[9] = { 0, 1, 2, 3, 4, 5, 6, 7, 8 };

  1835.             s->channel_order_tab = dca_channel_order_native;

  1836.         }

  1837.     } else {

  1838.         av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);

  1839.         return AVERROR_INVALIDDATA;

  1840.     }

  1841. 

  1842.     if (avctx->channels != channels) {

  1843.         if (avctx->channels)

  1844.             av_log(avctx, AV_LOG_INFO, "Number of channels changed in DCA decoder (%d -> %d)\n", avctx->channels, channels);

  1845.         avctx->channels = channels;

  1846.     }

  1847. 

  1848.     /* get output buffer */

  1849.     s->frame.nb_samples = 256 * (s->sample_blocks / 8);

  1850.     if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {

  1851.         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");

  1852.         return ret;

  1853.     }

  1854.     samples_flt = (float *)   s->frame.data[0];

  1855.     samples_s16 = (int16_t *) s->frame.data[0];

  1856. 

  1857.     /* filter to get final output */

  1858.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1859.         dca_filter_channels(s, i);

  1860. 

  1861.         /* If this was marked as a DTS-ES stream we need to subtract back- */

  1862.         /* channel from SL & SR to remove matrixed back-channel signal */

  1863.         if ((s->source_pcm_res & 1) && s->xch_present) {

  1864.             float *back_chan = s->samples + s->channel_order_tab[s->xch_base_channel]     * 256;

  1865.             float *lt_chan   = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256;

  1866.             float *rt_chan   = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;

  1867.             s->dsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);

  1868.             s->dsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);

  1869.         }

  1870. 

  1871.         if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {

  1872.             s->fmt_conv.float_interleave(samples_flt, s->samples_chanptr, 256,

  1873.                                          channels);

  1874.             samples_flt += 256 * channels;

  1875.         } else {

  1876.             s->fmt_conv.float_to_int16_interleave(samples_s16,

  1877.                                                   s->samples_chanptr, 256,

  1878.                                                   channels);

  1879.             samples_s16 += 256 * channels;

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

  1890. 

  1891.     return buf_size;

  1892. }

  1893. 

  1894. 

  1895. 

  1896. /**

  1897.  * DCA initialization

  1898.  *

  1899.  * @param avctx     pointer to the AVCodecContext

  1900.  */

  1901. 

  1902. static av_cold int dca_decode_init(AVCodecContext *avctx)

  1903. {

  1904.     DCAContext *s = avctx->priv_data;

  1905.     int i;

  1906. 

  1907.     s->avctx = avctx;

  1908.     dca_init_vlcs();

  1909. 

  1910.     ff_dsputil_init(&s->dsp, avctx);

  1911.     ff_mdct_init(&s->imdct, 6, 1, 1.0);

  1912.     ff_synth_filter_init(&s->synth);

  1913.     ff_dcadsp_init(&s->dcadsp);

  1914.     ff_fmt_convert_init(&s->fmt_conv, avctx);

  1915. 

  1916.     for (i = 0; i < DCA_PRIM_CHANNELS_MAX + 1; i++)

  1917.         s->samples_chanptr[i] = s->samples + i * 256;

  1918. 

  1919.     if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {

  1920.         avctx->sample_fmt = AV_SAMPLE_FMT_FLT;

  1921.         s->scale_bias     = 1.0 / 32768.0;

  1922.     } else {

  1923.         avctx->sample_fmt = AV_SAMPLE_FMT_S16;

  1924.         s->scale_bias     = 1.0;

  1925.     }

  1926. 

  1927.     /* allow downmixing to stereo */

  1928.     if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&

  1929.         avctx->request_channels == 2) {

  1930.         avctx->channels = avctx->request_channels;

  1931.     }

  1932. 

  1933.     avcodec_get_frame_defaults(&s->frame);

  1934.     avctx->coded_frame = &s->frame;

  1935. 

  1936.     return 0;

  1937. }

  1938. 

  1939. static av_cold int dca_decode_end(AVCodecContext *avctx)

  1940. {

  1941.     DCAContext *s = avctx->priv_data;

  1942.     ff_mdct_end(&s->imdct);

  1943.     return 0;

  1944. }

  1945. 

  1946. static const AVProfile profiles[] = {

  1947.     { FF_PROFILE_DTS,        "DTS"        },

  1948.     { FF_PROFILE_DTS_ES,     "DTS-ES"     },

  1949.     { FF_PROFILE_DTS_96_24,  "DTS 96/24"  },

  1950.     { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },

  1951.     { FF_PROFILE_DTS_HD_MA,  "DTS-HD MA"  },

  1952.     { FF_PROFILE_UNKNOWN },

  1953. };

  1954. 

  1955. AVCodec ff_dca_decoder = {

  1956.     .name            = "dca",

  1957.     .type            = AVMEDIA_TYPE_AUDIO,

  1958.     .id              = CODEC_ID_DTS,

  1959.     .priv_data_size  = sizeof(DCAContext),

  1960.     .init            = dca_decode_init,

  1961.     .decode          = dca_decode_frame,

  1962.     .close           = dca_decode_end,

  1963.     .long_name       = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),

  1964.     .capabilities    = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,

  1965.     .sample_fmts     = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,

  1966.                                                        AV_SAMPLE_FMT_S16,

  1967.                                                        AV_SAMPLE_FMT_NONE },

  1968.     .profiles        = NULL_IF_CONFIG_SMALL(profiles),

  1969. };