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FFmpeg/libavcodec/dcadec.c

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  1. /*

  2.  * DCA compatible decoder

  3.  * Copyright (C) 2004 Gildas Bazin

  4.  * Copyright (C) 2004 Benjamin Zores

  5.  * Copyright (C) 2006 Benjamin Larsson

  6.  * Copyright (C) 2007 Konstantin Shishkov

  7.  *

  8.  * This file is part of Libav.

  9.  *

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

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

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

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

  14.  *

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

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

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

  18.  * Lesser General Public License for more details.

  19.  *

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

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

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

  23.  */

  24. 

  25. #include <math.h>

  26. #include <stddef.h>

  27. #include <stdio.h>

  28. 

  29. #include "libavutil/channel_layout.h"

  30. #include "libavutil/common.h"

  31. #include "libavutil/float_dsp.h"

  32. #include "libavutil/internal.h"

  33. #include "libavutil/intreadwrite.h"

  34. #include "libavutil/mathematics.h"

  35. #include "libavutil/opt.h"

  36. #include "libavutil/samplefmt.h"

  37. #include "avcodec.h"

  38. #include "fft.h"

  39. #include "get_bits.h"

  40. #include "put_bits.h"

  41. #include "dcadata.h"

  42. #include "dcahuff.h"

  43. #include "dca.h"

  44. #include "mathops.h"

  45. #include "synth_filter.h"

  46. #include "dcadsp.h"

  47. #include "fmtconvert.h"

  48. #include "internal.h"

  49. 

  50. #if ARCH_ARM

  51. #   include "arm/dca.h"

  52. #endif

  53. #if ARCH_X86

  54. #   include "x86/dca.h"

  55. #endif

  56. 

  57. //#define TRACE

  58. 

  59. #define DCA_PRIM_CHANNELS_MAX  (7)

  60. #define DCA_SUBBANDS          (32)

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

  62. #define DCA_SUBSUBFRAMES_MAX   (4)

  63. #define DCA_SUBFRAMES_MAX     (16)

  64. #define DCA_BLOCKS_MAX        (16)

  65. #define DCA_LFE_MAX            (3)

  66. 

  67. enum DCAMode {

  68.     DCA_MONO = 0,

  69.     DCA_CHANNEL,

  70.     DCA_STEREO,

  71.     DCA_STEREO_SUMDIFF,

  72.     DCA_STEREO_TOTAL,

  73.     DCA_3F,

  74.     DCA_2F1R,

  75.     DCA_3F1R,

  76.     DCA_2F2R,

  77.     DCA_3F2R,

  78.     DCA_4F2R

  79. };

  80. 

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

  82. enum DCAExSSSpeakerMask {

  83.     DCA_EXSS_FRONT_CENTER          = 0x0001,

  84.     DCA_EXSS_FRONT_LEFT_RIGHT      = 0x0002,

  85.     DCA_EXSS_SIDE_REAR_LEFT_RIGHT  = 0x0004,

  86.     DCA_EXSS_LFE                   = 0x0008,

  87.     DCA_EXSS_REAR_CENTER           = 0x0010,

  88.     DCA_EXSS_FRONT_HIGH_LEFT_RIGHT = 0x0020,

  89.     DCA_EXSS_REAR_LEFT_RIGHT       = 0x0040,

  90.     DCA_EXSS_FRONT_HIGH_CENTER     = 0x0080,

  91.     DCA_EXSS_OVERHEAD              = 0x0100,

  92.     DCA_EXSS_CENTER_LEFT_RIGHT     = 0x0200,

  93.     DCA_EXSS_WIDE_LEFT_RIGHT       = 0x0400,

  94.     DCA_EXSS_SIDE_LEFT_RIGHT       = 0x0800,

  95.     DCA_EXSS_LFE2                  = 0x1000,

  96.     DCA_EXSS_SIDE_HIGH_LEFT_RIGHT  = 0x2000,

  97.     DCA_EXSS_REAR_HIGH_CENTER      = 0x4000,

  98.     DCA_EXSS_REAR_HIGH_LEFT_RIGHT  = 0x8000,

  99. };

  100. 

  101. enum DCAExtensionMask {

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

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

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

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

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

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

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

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

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

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

  112. };

  113. 

  114. /* -1 are reserved or unknown */

  115. static const int dca_ext_audio_descr_mask[] = {

  116.     DCA_EXT_XCH,

  117.     -1,

  118.     DCA_EXT_X96,

  119.     DCA_EXT_XCH | DCA_EXT_X96,

  120.     -1,

  121.     -1,

  122.     DCA_EXT_XXCH,

  123.     -1,

  124. };

  125. 

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

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

  128. 

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

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

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

  132.  *

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

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

  135.  * OV -> center back

  136.  * All 2 channel configurations -> AV_CH_LAYOUT_STEREO

  137.  */

  138. static const uint64_t dca_core_channel_layout[] = {

  139.     AV_CH_FRONT_CENTER,                                                     ///< 1, A

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

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

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

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

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

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

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

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

  148. 

  149.     AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_SIDE_LEFT |

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

  151. 

  152.     AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |

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

  154. 

  155.     AV_CH_LAYOUT_STEREO | AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT |

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

  157. 

  158.     AV_CH_FRONT_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |

  159.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_BACK_CENTER   |

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

  161. 

  162.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER   |

  163.     AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |

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

  165. 

  166.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |

  167.     AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |

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

  169. 

  170.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER   |

  171.     AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |

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

  173. };

  174. 

  175. static const int8_t dca_lfe_index[] = {

  176.     1, 2, 2, 2, 2, 3, 2, 3, 2, 3, 2, 3, 1, 3, 2, 3

  177. };

  178. 

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

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

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

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

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

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

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

  186.     { 0,  1,  3, -1, -1, -1, -1, -1, -1},

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

  188.     { 0,  1,  3,  4, -1, -1, -1, -1, -1},

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

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

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

  192.     { 0,  6,  4,  5,  2,  3, -1, -1, -1},

  193.     { 4,  2,  5,  0,  1,  6,  7, -1, -1},

  194.     { 5,  6,  0,  1,  7,  3,  8,  4, -1},

  195.     { 4,  2,  5,  0,  1,  6,  8,  7, -1},

  196. };

  197. 

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

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

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

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

  202.     { 0,  1,  3, -1, -1, -1, -1, -1, -1},

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

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

  205.     { 0,  1,  3,  4, -1, -1, -1, -1, -1},

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

  207.     { 0,  1,  4,  5,  3, -1, -1, -1, -1},

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

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

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

  211.     { 0,  6,  4,  5,  2,  3,  7, -1, -1},

  212.     { 4,  2,  5,  0,  1,  7,  8,  6, -1},

  213.     { 5,  6,  0,  1,  8,  3,  9,  4,  7},

  214.     { 4,  2,  5,  0,  1,  6,  9,  8,  7},

  215. };

  216. 

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

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

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

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

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

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

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

  224.     { 0,  1,  2, -1, -1, -1, -1, -1, -1},

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

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

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

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

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

  230.     { 0,  5,  3,  4,  1,  2, -1, -1, -1},

  231.     { 3,  2,  4,  0,  1,  5,  6, -1, -1},

  232.     { 4,  5,  0,  1,  6,  2,  7,  3, -1},

  233.     { 3,  2,  4,  0,  1,  5,  7,  6, -1},

  234. };

  235. 

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

  237.     { 0,  1, -1, -1, -1, -1, -1, -1, -1},

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

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

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

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

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

  243.     { 0,  1,  2,  3, -1, -1, -1, -1, -1},

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

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

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

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

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

  249.     { 0,  5,  3,  4,  1,  2,  6, -1, -1},

  250.     { 3,  2,  4,  0,  1,  6,  7,  5, -1},

  251.     { 4,  5,  0,  1,  7,  2,  8,  3,  6},

  252.     { 3,  2,  4,  0,  1,  5,  8,  7,  6},

  253. };

  254. 

  255. #define DCA_DOLBY                  101           /* FIXME */

  256. 

  257. #define DCA_CHANNEL_BITS             6

  258. #define DCA_CHANNEL_MASK          0x3F

  259. 

  260. #define DCA_LFE                   0x80

  261. 

  262. #define HEADER_SIZE                 14

  263. 

  264. #define DCA_MAX_FRAME_SIZE       16384

  265. #define DCA_MAX_EXSS_HEADER_SIZE  4096

  266. 

  267. #define DCA_BUFFER_PADDING_SIZE   1024

  268. 

  269. #define DCA_NSYNCAUX        0x9A1105A0

  270. 

  271. /** Bit allocation */

  272. typedef struct {

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

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

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

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

  277. } BitAlloc;

  278. 

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

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

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

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

  283. 

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

  285.                                          int idx)

  286. {

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

  288.            ba->offset;

  289. }

  290. 

  291. typedef struct {

  292.     AVClass *class;             ///< class for AVOptions

  293.     AVCodecContext *avctx;

  294.     /* Frame header */

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

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

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

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

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

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

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

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

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

  304. 

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

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

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

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

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

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

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

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

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

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

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

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

  317.     int copy_history;           ///< copy history

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

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

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

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

  322. 

  323.     /* Primary audio coding header */

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

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

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

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

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

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

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

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

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

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

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

  335. 

  336.     /* Primary audio coding side information */

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

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

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

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

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

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

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

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

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

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

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

  348. 

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

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

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

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

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

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

  355. 

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

  357. 

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

  359.     int lfe_scale_factor;

  360. 

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

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

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

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

  365.     int hist_index[DCA_PRIM_CHANNELS_MAX];

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

  367. 

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

  369. 

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

  371.     float *samples_chanptr[DCA_PRIM_CHANNELS_MAX + 1];

  372.     float *extra_channels[DCA_PRIM_CHANNELS_MAX + 1];

  373.     uint8_t *extra_channels_buffer;

  374.     unsigned int extra_channels_buffer_size;

  375. 

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

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

  378. 

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

  380.     GetBitContext gb;

  381.     /* Current position in DCA frame */

  382.     int current_subframe;

  383.     int current_subsubframe;

  384. 

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

  386. 

  387.     /* XCh extension information */

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

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

  390.     int xch_disable;            ///< whether the XCh extension should be decoded or not

  391. 

  392.     /* ExSS header parser */

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

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

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

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

  397. 

  398.     int profile;

  399. 

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

  401.     AVFloatDSPContext fdsp;

  402.     FFTContext imdct;

  403.     SynthFilterContext synth;

  404.     DCADSPContext dcadsp;

  405.     FmtConvertContext fmt_conv;

  406. } DCAContext;

  407. 

  408. static const uint16_t dca_vlc_offs[] = {

  409.         0,   512,   640,   768,  1282,  1794,  2436,  3080,  3770,  4454,  5364,

  410.      5372,  5380,  5388,  5392,  5396,  5412,  5420,  5428,  5460,  5492,  5508,

  411.      5572,  5604,  5668,  5796,  5860,  5892,  6412,  6668,  6796,  7308,  7564,

  412.      7820,  8076,  8620,  9132,  9388,  9910, 10166, 10680, 11196, 11726, 12240,

  413.     12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264,

  414.     18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622,

  415. };

  416. 

  417. static av_cold void dca_init_vlcs(void)

  418. {

  419.     static int vlcs_initialized = 0;

  420.     int i, j, c = 14;

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

  422. 

  423.     if (vlcs_initialized)

  424.         return;

  425. 

  426.     dca_bitalloc_index.offset = 1;

  427.     dca_bitalloc_index.wrap = 2;

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

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

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

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

  432.                  bitalloc_12_bits[i], 1, 1,

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

  434.     }

  435.     dca_scalefactor.offset = -64;

  436.     dca_scalefactor.wrap = 2;

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

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

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

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

  441.                  scales_bits[i], 1, 1,

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

  443.     }

  444.     dca_tmode.offset = 0;

  445.     dca_tmode.wrap = 1;

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

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

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

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

  450.                  tmode_bits[i], 1, 1,

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

  452.     }

  453. 

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

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

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

  457.                 break;

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

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

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

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

  462. 

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

  464.                      bitalloc_sizes[i],

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

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

  467.             c++;

  468.         }

  469.     vlcs_initialized = 1;

  470. }

  471. 

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

  473. {

  474.     while (len--)

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

  476. }

  477. 

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

  479. {

  480.     int i, j;

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

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

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

  484. 

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

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

  487. 

  488.     if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)

  489.         s->prim_channels = DCA_PRIM_CHANNELS_MAX;

  490. 

  491. 

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

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

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

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

  496.     }

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

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

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

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

  501.     }

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

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

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

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

  506. 

  507.     /* Get codebooks quantization indexes */

  508.     if (!base_channel)

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

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

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

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

  513. 

  514.     /* Get scale factor adjustment */

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

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

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

  518. 

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

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

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

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

  523. 

  524.     if (s->crc_present) {

  525.         /* Audio header CRC check */

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

  527.     }

  528. 

  529.     s->current_subframe    = 0;

  530.     s->current_subsubframe = 0;

  531. 

  532. #ifdef TRACE

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

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

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

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

  537.                s->subband_activity[i]);

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

  539.                s->vq_start_subband[i]);

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

  541.                s->joint_intensity[i]);

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

  543.                s->transient_huffman[i]);

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

  545.                s->scalefactor_huffman[i]);

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

  547.                s->bitalloc_huffman[i]);

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

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

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

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

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

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

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

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

  556.     }

  557. #endif

  558. 

  559.     return 0;

  560. }

  561. 

  562. static int dca_parse_frame_header(DCAContext *s)

  563. {

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

  565. 

  566.     /* Sync code */

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

  568. 

  569.     /* Frame header */

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

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

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

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

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

  575.     if (s->frame_size < 95)

  576.         return AVERROR_INVALIDDATA;

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

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

  579.     if (!s->sample_rate)

  580.         return AVERROR_INVALIDDATA;

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

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

  583.     if (!s->bit_rate)

  584.         return AVERROR_INVALIDDATA;

  585. 

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

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

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

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

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

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

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

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

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

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

  596. 

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

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

  599.         return AVERROR_INVALIDDATA;

  600.     }

  601. 

  602.     /* TODO: check CRC */

  603.     if (s->crc_present)

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

  605. 

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

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

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

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

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

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

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

  613. 

  614.     /* FIXME: channels mixing levels */

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

  616.     if (s->lfe)

  617.         s->output |= DCA_LFE;

  618. 

  619. #ifdef TRACE

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

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

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

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

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

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

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

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

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

  629.            s->sample_rate);

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

  631.            s->bit_rate);

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

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

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

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

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

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

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

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

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

  641.            s->predictor_history);

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

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

  644.            s->multirate_inter);

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

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

  647.     av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

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

  654. #endif

  655. 

  656.     /* Primary audio coding header */

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

  658. 

  659.     return dca_parse_audio_coding_header(s, 0);

  660. }

  661. 

  662. 

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

  664. {

  665.     if (level < 5) {

  666.         /* huffman encoded */

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

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

  669.     } else if (level < 8) {

  670.         if (level + 1 > log2range) {

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

  672.             value = get_bits(gb, log2range);

  673.         } else {

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

  675.         }

  676.     }

  677.     return value;

  678. }

  679. 

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

  681. {

  682.     /* Primary audio coding side information */

  683.     int j, k;

  684. 

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

  686.         return AVERROR_INVALIDDATA;

  687. 

  688.     if (!base_channel) {

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

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

  691.     }

  692. 

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

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

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

  696.     }

  697. 

  698.     /* Get prediction codebook */

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

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

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

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

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

  704.             }

  705.         }

  706.     }

  707. 

  708.     /* Bit allocation index */

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

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

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

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

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

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

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

  716.                 av_log(s->avctx, AV_LOG_ERROR,

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

  718.                 return AVERROR_INVALIDDATA;

  719.             } else {

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

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

  722.             }

  723. 

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

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

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

  727.                 return AVERROR_INVALIDDATA;

  728.             }

  729.         }

  730.     }

  731. 

  732.     /* Transition mode */

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

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

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

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

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

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

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

  740.             }

  741.         }

  742.     }

  743. 

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

  745.         return AVERROR_INVALIDDATA;

  746. 

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

  748.         const uint32_t *scale_table;

  749.         int scale_sum, log_size;

  750. 

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

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

  753. 

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

  755.             scale_table = scale_factor_quant7;

  756.             log_size = 7;

  757.         } else {

  758.             scale_table = scale_factor_quant6;

  759.             log_size = 6;

  760.         }

  761. 

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

  763.         scale_sum = 0;

  764. 

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

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

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

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

  769.             }

  770. 

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

  772.                 /* Get second scale factor */

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

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

  775.             }

  776.         }

  777.     }

  778. 

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

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

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

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

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

  784.     }

  785. 

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

  787.         return AVERROR_INVALIDDATA;

  788. 

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

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

  791.         int source_channel;

  792. 

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

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

  795.             int scale = 0;

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

  797. 

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

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

  800. 

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

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

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

  804.             }

  805. 

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

  807.                 av_log(s->avctx, AV_LOG_DEBUG,

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

  809.                 s->debug_flag |= 0x02;

  810.             }

  811.         }

  812.     }

  813. 

  814.     /* Dynamic range coefficient */

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

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

  817. 

  818.     /* Side information CRC check word */

  819.     if (s->crc_present) {

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

  821.     }

  822. 

  823.     /*

  824.      * Primary audio data arrays

  825.      */

  826. 

  827.     /* VQ encoded high frequency subbands */

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

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

  830.             /* 1 vector -> 32 samples */

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

  832. 

  833.     /* Low frequency effect data */

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

  835.         /* LFE samples */

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

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

  838.         float lfe_scale;

  839. 

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

  841.             /* Signed 8 bits int */

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

  843.         }

  844. 

  845.         /* Scale factor index */

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

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

  848. 

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

  850.         lfe_scale = 0.035 * s->lfe_scale_factor;

  851. 

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

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

  854.     }

  855. 

  856. #ifdef TRACE

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

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

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

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

  861. 

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

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

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

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

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

  867.     }

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

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

  870.             av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

  876.     }

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

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

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

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

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

  882.     }

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

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

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

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

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

  888.     }

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

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

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

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

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

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

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

  896.         }

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

  898.     }

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

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

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

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

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

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

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

  906.         }

  907.     }

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

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

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

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

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

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

  914. 

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

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

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

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

  919.     }

  920. #endif

  921. 

  922.     return 0;

  923. }

  924. 

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

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

  927.                             float scale)

  928. {

  929.     const float *prCoeff;

  930. 

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

  932. 

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

  934. 

  935.     /* Select filter */

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

  937.         prCoeff = fir_32bands_nonperfect;

  938.     else                        /* Perfect reconstruction */

  939.         prCoeff = fir_32bands_perfect;

  940. 

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

  942.                               s->subband_fir_hist[chans],

  943.                               &s->hist_index[chans],

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

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

  946. }

  947. 

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

  949.                                   int num_deci_sample, float *samples_in,

  950.                                   float *samples_out, float scale)

  951. {

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

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

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

  955.      *   from last subframe as history.

  956.      *

  957.      * samples_out: An array holding interpolated samples

  958.      */

  959. 

  960.     int idx;

  961.     const float *prCoeff;

  962.     int deciindex;

  963. 

  964.     /* Select decimation filter */

  965.     if (decimation_select == 1) {

  966.         idx = 1;

  967.         prCoeff = lfe_fir_128;

  968.     } else {

  969.         idx = 0;

  970.         prCoeff = lfe_fir_64;

  971.     }

  972.     /* Interpolation */

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

  974.         s->dcadsp.lfe_fir[idx](samples_out, samples_in, prCoeff, scale);

  975.         samples_in++;

  976.         samples_out += 2 * 32 * (1 + idx);

  977.     }

  978. }

  979. 

  980. /* downmixing routines */

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

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

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

  984. 

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

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

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

  988. 

  989. #define MIX_FRONT3(samples, coef)                                      \

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

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

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

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

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

  995. 

  996. #define DOWNMIX_TO_STEREO(op1, op2)             \

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

  998.         op1                                     \

  999.         op2                                     \

  1000.     }

  1001. 

  1002. static void dca_downmix(float **samples, int srcfmt, int lfe_present,

  1003.                         float coef[DCA_PRIM_CHANNELS_MAX + 1][2],

  1004.                         const int8_t *channel_mapping)

  1005. {

  1006.     int c, l, r, sl, sr, s;

  1007.     int i;

  1008.     float t, u, v;

  1009. 

  1010.     switch (srcfmt) {

  1011.     case DCA_MONO:

  1012.     case DCA_4F2R:

  1013.         av_log(NULL, 0, "Not implemented!\n");

  1014.         break;

  1015.     case DCA_CHANNEL:

  1016.     case DCA_STEREO:

  1017.     case DCA_STEREO_TOTAL:

  1018.     case DCA_STEREO_SUMDIFF:

  1019.         break;

  1020.     case DCA_3F:

  1021.         c = channel_mapping[0];

  1022.         l = channel_mapping[1];

  1023.         r = channel_mapping[2];

  1024.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );

  1025.         break;

  1026.     case DCA_2F1R:

  1027.         s = channel_mapping[2];

  1028.         DOWNMIX_TO_STEREO(MIX_REAR1(samples, s, 2, coef), );

  1029.         break;

  1030.     case DCA_3F1R:

  1031.         c = channel_mapping[0];

  1032.         l = channel_mapping[1];

  1033.         r = channel_mapping[2];

  1034.         s = channel_mapping[3];

  1035.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1036.                           MIX_REAR1(samples, s, 3, coef));

  1037.         break;

  1038.     case DCA_2F2R:

  1039.         sl = channel_mapping[2];

  1040.         sr = channel_mapping[3];

  1041.         DOWNMIX_TO_STEREO(MIX_REAR2(samples, sl, sr, 2, coef), );

  1042.         break;

  1043.     case DCA_3F2R:

  1044.         c  = channel_mapping[0];

  1045.         l  = channel_mapping[1];

  1046.         r  = channel_mapping[2];

  1047.         sl = channel_mapping[3];

  1048.         sr = channel_mapping[4];

  1049.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1050.                           MIX_REAR2(samples, sl, sr, 3, coef));

  1051.         break;

  1052.     }

  1053.     if (lfe_present) {

  1054.         int lf_buf = dca_lfe_index[srcfmt];

  1055.         int lf_idx = dca_channels [srcfmt];

  1056.         for (i = 0; i < 256; i++) {

  1057.             samples[0][i] += samples[lf_buf][i] * coef[lf_idx][0];

  1058.             samples[1][i] += samples[lf_buf][i] * coef[lf_idx][1];

  1059.         }

  1060.     }

  1061. }

  1062. 

  1063. 

  1064. #ifndef decode_blockcodes

  1065. /* Very compact version of the block code decoder that does not use table

  1066.  * look-up but is slightly slower */

  1067. static int decode_blockcode(int code, int levels, int32_t *values)

  1068. {

  1069.     int i;

  1070.     int offset = (levels - 1) >> 1;

  1071. 

  1072.     for (i = 0; i < 4; i++) {

  1073.         int div = FASTDIV(code, levels);

  1074.         values[i] = code - offset - div * levels;

  1075.         code = div;

  1076.     }

  1077. 

  1078.     return code;

  1079. }

  1080. 

  1081. static int decode_blockcodes(int code1, int code2, int levels, int32_t *values)

  1082. {

  1083.     return decode_blockcode(code1, levels, values) |

  1084.            decode_blockcode(code2, levels, values + 4);

  1085. }

  1086. #endif

  1087. 

  1088. static const uint8_t abits_sizes[7]  = { 7, 10, 12, 13, 15, 17, 19 };

  1089. static const uint8_t abits_levels[7] = { 3,  5,  7,  9, 13, 17, 25 };

  1090. 

  1091. #ifndef int8x8_fmul_int32

  1092. static inline void int8x8_fmul_int32(DCADSPContext *dsp, float *dst,

  1093.                                      const int8_t *src, int scale)

  1094. {

  1095.     dsp->int8x8_fmul_int32(dst, src, scale);

  1096. }

  1097. #endif

  1098. 

  1099. static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)

  1100. {

  1101.     int k, l;

  1102.     int subsubframe = s->current_subsubframe;

  1103. 

  1104.     const float *quant_step_table;

  1105. 

  1106.     /* FIXME */

  1107.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1108.     LOCAL_ALIGNED_16(int32_t, block, [8 * DCA_SUBBANDS]);

  1109. 

  1110.     /*

  1111.      * Audio data

  1112.      */

  1113. 

  1114.     /* Select quantization step size table */

  1115.     if (s->bit_rate_index == 0x1f)

  1116.         quant_step_table = lossless_quant_d;

  1117.     else

  1118.         quant_step_table = lossy_quant_d;

  1119. 

  1120.     for (k = base_channel; k < s->prim_channels; k++) {

  1121.         float rscale[DCA_SUBBANDS];

  1122. 

  1123.         if (get_bits_left(&s->gb) < 0)

  1124.             return AVERROR_INVALIDDATA;

  1125. 

  1126.         for (l = 0; l < s->vq_start_subband[k]; l++) {

  1127.             int m;

  1128. 

  1129.             /* Select the mid-tread linear quantizer */

  1130.             int abits = s->bitalloc[k][l];

  1131. 

  1132.             float quant_step_size = quant_step_table[abits];

  1133. 

  1134.             /*

  1135.              * Determine quantization index code book and its type

  1136.              */

  1137. 

  1138.             /* Select quantization index code book */

  1139.             int sel = s->quant_index_huffman[k][abits];

  1140. 

  1141.             /*

  1142.              * Extract bits from the bit stream

  1143.              */

  1144.             if (!abits) {

  1145.                 rscale[l] = 0;

  1146.                 memset(block + 8 * l, 0, 8 * sizeof(block[0]));

  1147.             } else {

  1148.                 /* Deal with transients */

  1149.                 int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];

  1150.                 rscale[l] = quant_step_size * s->scale_factor[k][l][sfi] *

  1151.                                s->scalefactor_adj[k][sel];

  1152. 

  1153.                 if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {

  1154.                     if (abits <= 7) {

  1155.                         /* Block code */

  1156.                         int block_code1, block_code2, size, levels, err;

  1157. 

  1158.                         size   = abits_sizes[abits - 1];

  1159.                         levels = abits_levels[abits - 1];

  1160. 

  1161.                         block_code1 = get_bits(&s->gb, size);

  1162.                         block_code2 = get_bits(&s->gb, size);

  1163.                         err = decode_blockcodes(block_code1, block_code2,

  1164.                                                 levels, block + 8 * l);

  1165.                         if (err) {

  1166.                             av_log(s->avctx, AV_LOG_ERROR,

  1167.                                    "ERROR: block code look-up failed\n");

  1168.                             return AVERROR_INVALIDDATA;

  1169.                         }

  1170.                     } else {

  1171.                         /* no coding */

  1172.                         for (m = 0; m < 8; m++)

  1173.                             block[8 * l + m] = get_sbits(&s->gb, abits - 3);

  1174.                     }

  1175.                 } else {

  1176.                     /* Huffman coded */

  1177.                     for (m = 0; m < 8; m++)

  1178.                         block[8 * l + m] = get_bitalloc(&s->gb,

  1179.                                                 &dca_smpl_bitalloc[abits], sel);

  1180.                 }

  1181. 

  1182.             }

  1183.         }

  1184. 

  1185.         s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[k][0],

  1186.                                                block, rscale, 8 * s->vq_start_subband[k]);

  1187. 

  1188.         for (l = 0; l < s->vq_start_subband[k]; l++) {

  1189.             int m;

  1190.             /*

  1191.              * Inverse ADPCM if in prediction mode

  1192.              */

  1193.             if (s->prediction_mode[k][l]) {

  1194.                 int n;

  1195.                 for (m = 0; m < 8; m++) {

  1196.                     for (n = 1; n <= 4; n++)

  1197.                         if (m >= n)

  1198.                             subband_samples[k][l][m] +=

  1199.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1200.                                  subband_samples[k][l][m - n] / 8192);

  1201.                         else if (s->predictor_history)

  1202.                             subband_samples[k][l][m] +=

  1203.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1204.                                  s->subband_samples_hist[k][l][m - n + 4] / 8192);

  1205.                 }

  1206.             }

  1207.         }

  1208. 

  1209.         /*

  1210.          * Decode VQ encoded high frequencies

  1211.          */

  1212.         for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {

  1213.             /* 1 vector -> 32 samples but we only need the 8 samples

  1214.              * for this subsubframe. */

  1215.             int hfvq = s->high_freq_vq[k][l];

  1216. 

  1217.             if (!s->debug_flag & 0x01) {

  1218.                 av_log(s->avctx, AV_LOG_DEBUG,

  1219.                        "Stream with high frequencies VQ coding\n");

  1220.                 s->debug_flag |= 0x01;

  1221.             }

  1222. 

  1223.             int8x8_fmul_int32(&s->dcadsp, subband_samples[k][l],

  1224.                               &high_freq_vq[hfvq][subsubframe * 8],

  1225.                               s->scale_factor[k][l][0]);

  1226.         }

  1227.     }

  1228. 

  1229.     /* Check for DSYNC after subsubframe */

  1230.     if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {

  1231.         if (0xFFFF == get_bits(&s->gb, 16)) {   /* 0xFFFF */

  1232. #ifdef TRACE

  1233.             av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");

  1234. #endif

  1235.         } else {

  1236.             av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");

  1237.             return AVERROR_INVALIDDATA;

  1238.         }

  1239.     }

  1240. 

  1241.     /* Backup predictor history for adpcm */

  1242.     for (k = base_channel; k < s->prim_channels; k++)

  1243.         for (l = 0; l < s->vq_start_subband[k]; l++)

  1244.             memcpy(s->subband_samples_hist[k][l],

  1245.                    &subband_samples[k][l][4],

  1246.                    4 * sizeof(subband_samples[0][0][0]));

  1247. 

  1248.     return 0;

  1249. }

  1250. 

  1251. static int dca_filter_channels(DCAContext *s, int block_index)

  1252. {

  1253.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1254.     int k;

  1255. 

  1256.     /* 32 subbands QMF */

  1257.     for (k = 0; k < s->prim_channels; k++) {

  1258. /*        static float pcm_to_double[8] = { 32768.0, 32768.0, 524288.0, 524288.0,

  1259.                                             0, 8388608.0, 8388608.0 };*/

  1260.         if (s->channel_order_tab[k] >= 0)

  1261.             qmf_32_subbands(s, k, subband_samples[k],

  1262.                             s->samples_chanptr[s->channel_order_tab[k]],

  1263.                             M_SQRT1_2 / 32768.0 /* pcm_to_double[s->source_pcm_res] */);

  1264.     }

  1265. 

  1266.     /* Generate LFE samples for this subsubframe FIXME!!! */

  1267.     if (s->lfe) {

  1268.         lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,

  1269.                               s->lfe_data + 2 * s->lfe * (block_index + 4),

  1270.                               s->samples_chanptr[dca_lfe_index[s->amode]],

  1271.                               1.0 / (256.0 * 32768.0));

  1272.         /* Outputs 20bits pcm samples */

  1273.     }

  1274. 

  1275.     /* Downmixing to Stereo */

  1276.     if (s->prim_channels + !!s->lfe > 2 &&

  1277.         s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {

  1278.         dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef,

  1279.                     s->channel_order_tab);

  1280.     }

  1281. 

  1282.     return 0;

  1283. }

  1284. 

  1285. 

  1286. static int dca_subframe_footer(DCAContext *s, int base_channel)

  1287. {

  1288.     int in, out, aux_data_count, aux_data_end, reserved;

  1289.     uint32_t nsyncaux;

  1290. 

  1291.     /*

  1292.      * Unpack optional information

  1293.      */

  1294. 

  1295.     /* presumably optional information only appears in the core? */

  1296.     if (!base_channel) {

  1297.         if (s->timestamp)

  1298.             skip_bits_long(&s->gb, 32);

  1299. 

  1300.         if (s->aux_data) {

  1301.             aux_data_count = get_bits(&s->gb, 6);

  1302. 

  1303.             // align (32-bit)

  1304.             skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1305. 

  1306.             aux_data_end = 8 * aux_data_count + get_bits_count(&s->gb);

  1307. 

  1308.             if ((nsyncaux = get_bits_long(&s->gb, 32)) != DCA_NSYNCAUX) {

  1309.                 av_log(s->avctx, AV_LOG_ERROR, "nSYNCAUX mismatch %#"PRIx32"\n",

  1310.                        nsyncaux);

  1311.                 return AVERROR_INVALIDDATA;

  1312.             }

  1313. 

  1314.             if (get_bits1(&s->gb)) { // bAUXTimeStampFlag

  1315.                 avpriv_request_sample(s->avctx,

  1316.                                       "Auxiliary Decode Time Stamp Flag");

  1317.                 // align (4-bit)

  1318.                 skip_bits(&s->gb, (-get_bits_count(&s->gb)) & 4);

  1319.                 // 44 bits: nMSByte (8), nMarker (4), nLSByte (28), nMarker (4)

  1320.                 skip_bits_long(&s->gb, 44);

  1321.             }

  1322. 

  1323.             if ((s->core_downmix = get_bits1(&s->gb))) {

  1324.                 int am = get_bits(&s->gb, 3);

  1325.                 switch (am) {

  1326.                 case 0:

  1327.                     s->core_downmix_amode = DCA_MONO;

  1328.                     break;

  1329.                 case 1:

  1330.                     s->core_downmix_amode = DCA_STEREO;

  1331.                     break;

  1332.                 case 2:

  1333.                     s->core_downmix_amode = DCA_STEREO_TOTAL;

  1334.                     break;

  1335.                 case 3:

  1336.                     s->core_downmix_amode = DCA_3F;

  1337.                     break;

  1338.                 case 4:

  1339.                     s->core_downmix_amode = DCA_2F1R;

  1340.                     break;

  1341.                 case 5:

  1342.                     s->core_downmix_amode = DCA_2F2R;

  1343.                     break;

  1344.                 case 6:

  1345.                     s->core_downmix_amode = DCA_3F1R;

  1346.                     break;

  1347.                 default:

  1348.                     av_log(s->avctx, AV_LOG_ERROR,

  1349.                            "Invalid mode %d for embedded downmix coefficients\n",

  1350.                            am);

  1351.                     return AVERROR_INVALIDDATA;

  1352.                 }

  1353.                 for (out = 0; out < dca_channels[s->core_downmix_amode]; out++) {

  1354.                     for (in = 0; in < s->prim_channels + !!s->lfe; in++) {

  1355.                         uint16_t tmp = get_bits(&s->gb, 9);

  1356.                         if ((tmp & 0xFF) > 241) {

  1357.                             av_log(s->avctx, AV_LOG_ERROR,

  1358.                                    "Invalid downmix coefficient code %"PRIu16"\n",

  1359.                                    tmp);

  1360.                             return AVERROR_INVALIDDATA;

  1361.                         }

  1362.                         s->core_downmix_codes[in][out] = tmp;

  1363.                     }

  1364.                 }

  1365.             }

  1366. 

  1367.             align_get_bits(&s->gb); // byte align

  1368.             skip_bits(&s->gb, 16);  // nAUXCRC16

  1369. 

  1370.             // additional data (reserved, cf. ETSI TS 102 114 V1.4.1)

  1371.             if ((reserved = (aux_data_end - get_bits_count(&s->gb))) < 0) {

  1372.                  av_log(s->avctx, AV_LOG_ERROR,

  1373.                         "Overread auxiliary data by %d bits\n", -reserved);

  1374.                 return AVERROR_INVALIDDATA;

  1375.             } else if (reserved) {

  1376.                 avpriv_request_sample(s->avctx,

  1377.                                       "Core auxiliary data reserved content");

  1378.                 skip_bits_long(&s->gb, reserved);

  1379.             }

  1380.         }

  1381. 

  1382.         if (s->crc_present && s->dynrange)

  1383.             get_bits(&s->gb, 16);

  1384.     }

  1385. 

  1386.     return 0;

  1387. }

  1388. 

  1389. /**

  1390.  * Decode a dca frame block

  1391.  *

  1392.  * @param s     pointer to the DCAContext

  1393.  */

  1394. 

  1395. static int dca_decode_block(DCAContext *s, int base_channel, int block_index)

  1396. {

  1397.     int ret;

  1398. 

  1399.     /* Sanity check */

  1400.     if (s->current_subframe >= s->subframes) {

  1401.         av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",

  1402.                s->current_subframe, s->subframes);

  1403.         return AVERROR_INVALIDDATA;

  1404.     }

  1405. 

  1406.     if (!s->current_subsubframe) {

  1407. #ifdef TRACE

  1408.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");

  1409. #endif

  1410.         /* Read subframe header */

  1411.         if ((ret = dca_subframe_header(s, base_channel, block_index)))

  1412.             return ret;

  1413.     }

  1414. 

  1415.     /* Read subsubframe */

  1416. #ifdef TRACE

  1417.     av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");

  1418. #endif

  1419.     if ((ret = dca_subsubframe(s, base_channel, block_index)))

  1420.         return ret;

  1421. 

  1422.     /* Update state */

  1423.     s->current_subsubframe++;

  1424.     if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {

  1425.         s->current_subsubframe = 0;

  1426.         s->current_subframe++;

  1427.     }

  1428.     if (s->current_subframe >= s->subframes) {

  1429. #ifdef TRACE

  1430.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");

  1431. #endif

  1432.         /* Read subframe footer */

  1433.         if ((ret = dca_subframe_footer(s, base_channel)))

  1434.             return ret;

  1435.     }

  1436. 

  1437.     return 0;

  1438. }

  1439. 

  1440. /**

  1441.  * Return the number of channels in an ExSS speaker mask (HD)

  1442.  */

  1443. static int dca_exss_mask2count(int mask)

  1444. {

  1445.     /* count bits that mean speaker pairs twice */

  1446.     return av_popcount(mask) +

  1447.            av_popcount(mask & (DCA_EXSS_CENTER_LEFT_RIGHT      |

  1448.                                DCA_EXSS_FRONT_LEFT_RIGHT       |

  1449.                                DCA_EXSS_FRONT_HIGH_LEFT_RIGHT  |

  1450.                                DCA_EXSS_WIDE_LEFT_RIGHT        |

  1451.                                DCA_EXSS_SIDE_LEFT_RIGHT        |

  1452.                                DCA_EXSS_SIDE_HIGH_LEFT_RIGHT   |

  1453.                                DCA_EXSS_SIDE_REAR_LEFT_RIGHT   |

  1454.                                DCA_EXSS_REAR_LEFT_RIGHT        |

  1455.                                DCA_EXSS_REAR_HIGH_LEFT_RIGHT));

  1456. }

  1457. 

  1458. /**

  1459.  * Skip mixing coefficients of a single mix out configuration (HD)

  1460.  */

  1461. static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch)

  1462. {

  1463.     int i;

  1464. 

  1465.     for (i = 0; i < channels; i++) {

  1466.         int mix_map_mask = get_bits(gb, out_ch);

  1467.         int num_coeffs = av_popcount(mix_map_mask);

  1468.         skip_bits_long(gb, num_coeffs * 6);

  1469.     }

  1470. }

  1471. 

  1472. /**

  1473.  * Parse extension substream asset header (HD)

  1474.  */

  1475. static int dca_exss_parse_asset_header(DCAContext *s)

  1476. {

  1477.     int header_pos = get_bits_count(&s->gb);

  1478.     int header_size;

  1479.     int channels;

  1480.     int embedded_stereo = 0;

  1481.     int embedded_6ch    = 0;

  1482.     int drc_code_present;

  1483.     int extensions_mask;

  1484.     int i, j;

  1485. 

  1486.     if (get_bits_left(&s->gb) < 16)

  1487.         return -1;

  1488. 

  1489.     /* We will parse just enough to get to the extensions bitmask with which

  1490.      * we can set the profile value. */

  1491. 

  1492.     header_size = get_bits(&s->gb, 9) + 1;

  1493.     skip_bits(&s->gb, 3); // asset index

  1494. 

  1495.     if (s->static_fields) {

  1496.         if (get_bits1(&s->gb))

  1497.             skip_bits(&s->gb, 4); // asset type descriptor

  1498.         if (get_bits1(&s->gb))

  1499.             skip_bits_long(&s->gb, 24); // language descriptor

  1500. 

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

  1502.             /* How can one fit 1024 bytes of text here if the maximum value

  1503.              * for the asset header size field above was 512 bytes? */

  1504.             int text_length = get_bits(&s->gb, 10) + 1;

  1505.             if (get_bits_left(&s->gb) < text_length * 8)

  1506.                 return -1;

  1507.             skip_bits_long(&s->gb, text_length * 8); // info text

  1508.         }

  1509. 

  1510.         skip_bits(&s->gb, 5); // bit resolution - 1

  1511.         skip_bits(&s->gb, 4); // max sample rate code

  1512.         channels = get_bits(&s->gb, 8) + 1;

  1513. 

  1514.         if (get_bits1(&s->gb)) { // 1-to-1 channels to speakers

  1515.             int spkr_remap_sets;

  1516.             int spkr_mask_size = 16;

  1517.             int num_spkrs[7];

  1518. 

  1519.             if (channels > 2)

  1520.                 embedded_stereo = get_bits1(&s->gb);

  1521.             if (channels > 6)

  1522.                 embedded_6ch = get_bits1(&s->gb);

  1523. 

  1524.             if (get_bits1(&s->gb)) {

  1525.                 spkr_mask_size = (get_bits(&s->gb, 2) + 1) << 2;

  1526.                 skip_bits(&s->gb, spkr_mask_size); // spkr activity mask

  1527.             }

  1528. 

  1529.             spkr_remap_sets = get_bits(&s->gb, 3);

  1530. 

  1531.             for (i = 0; i < spkr_remap_sets; i++) {

  1532.                 /* std layout mask for each remap set */

  1533.                 num_spkrs[i] = dca_exss_mask2count(get_bits(&s->gb, spkr_mask_size));

  1534.             }

  1535. 

  1536.             for (i = 0; i < spkr_remap_sets; i++) {

  1537.                 int num_dec_ch_remaps = get_bits(&s->gb, 5) + 1;

  1538.                 if (get_bits_left(&s->gb) < 0)

  1539.                     return -1;

  1540. 

  1541.                 for (j = 0; j < num_spkrs[i]; j++) {

  1542.                     int remap_dec_ch_mask = get_bits_long(&s->gb, num_dec_ch_remaps);

  1543.                     int num_dec_ch = av_popcount(remap_dec_ch_mask);

  1544.                     skip_bits_long(&s->gb, num_dec_ch * 5); // remap codes

  1545.                 }

  1546.             }

  1547. 

  1548.         } else {

  1549.             skip_bits(&s->gb, 3); // representation type

  1550.         }

  1551.     }

  1552. 

  1553.     drc_code_present = get_bits1(&s->gb);

  1554.     if (drc_code_present)

  1555.         get_bits(&s->gb, 8); // drc code

  1556. 

  1557.     if (get_bits1(&s->gb))

  1558.         skip_bits(&s->gb, 5); // dialog normalization code

  1559. 

  1560.     if (drc_code_present && embedded_stereo)

  1561.         get_bits(&s->gb, 8); // drc stereo code

  1562. 

  1563.     if (s->mix_metadata && get_bits1(&s->gb)) {

  1564.         skip_bits(&s->gb, 1); // external mix

  1565.         skip_bits(&s->gb, 6); // post mix gain code

  1566. 

  1567.         if (get_bits(&s->gb, 2) != 3) // mixer drc code

  1568.             skip_bits(&s->gb, 3); // drc limit

  1569.         else

  1570.             skip_bits(&s->gb, 8); // custom drc code

  1571. 

  1572.         if (get_bits1(&s->gb)) // channel specific scaling

  1573.             for (i = 0; i < s->num_mix_configs; i++)

  1574.                 skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6); // scale codes

  1575.         else

  1576.             skip_bits_long(&s->gb, s->num_mix_configs * 6); // scale codes

  1577. 

  1578.         for (i = 0; i < s->num_mix_configs; i++) {

  1579.             if (get_bits_left(&s->gb) < 0)

  1580.                 return -1;

  1581.             dca_exss_skip_mix_coeffs(&s->gb, channels, s->mix_config_num_ch[i]);

  1582.             if (embedded_6ch)

  1583.                 dca_exss_skip_mix_coeffs(&s->gb, 6, s->mix_config_num_ch[i]);

  1584.             if (embedded_stereo)

  1585.                 dca_exss_skip_mix_coeffs(&s->gb, 2, s->mix_config_num_ch[i]);

  1586.         }

  1587.     }

  1588. 

  1589.     switch (get_bits(&s->gb, 2)) {

  1590.     case 0: extensions_mask = get_bits(&s->gb, 12); break;

  1591.     case 1: extensions_mask = DCA_EXT_EXSS_XLL;     break;

  1592.     case 2: extensions_mask = DCA_EXT_EXSS_LBR;     break;

  1593.     case 3: extensions_mask = 0; /* aux coding */   break;

  1594.     }

  1595. 

  1596.     /* not parsed further, we were only interested in the extensions mask */

  1597. 

  1598.     if (get_bits_left(&s->gb) < 0)

  1599.         return -1;

  1600. 

  1601.     if (get_bits_count(&s->gb) - header_pos > header_size * 8) {

  1602.         av_log(s->avctx, AV_LOG_WARNING, "Asset header size mismatch.\n");

  1603.         return -1;

  1604.     }

  1605.     skip_bits_long(&s->gb, header_pos + header_size * 8 - get_bits_count(&s->gb));

  1606. 

  1607.     if (extensions_mask & DCA_EXT_EXSS_XLL)

  1608.         s->profile = FF_PROFILE_DTS_HD_MA;

  1609.     else if (extensions_mask & (DCA_EXT_EXSS_XBR | DCA_EXT_EXSS_X96 |

  1610.                                 DCA_EXT_EXSS_XXCH))

  1611.         s->profile = FF_PROFILE_DTS_HD_HRA;

  1612. 

  1613.     if (!(extensions_mask & DCA_EXT_CORE))

  1614.         av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n");

  1615.     if ((extensions_mask & DCA_CORE_EXTS) != s->core_ext_mask)

  1616.         av_log(s->avctx, AV_LOG_WARNING,

  1617.                "DTS extensions detection mismatch (%d, %d)\n",

  1618.                extensions_mask & DCA_CORE_EXTS, s->core_ext_mask);

  1619. 

  1620.     return 0;

  1621. }

  1622. 

  1623. /**

  1624.  * Parse extension substream header (HD)

  1625.  */

  1626. static void dca_exss_parse_header(DCAContext *s)

  1627. {

  1628.     int ss_index;

  1629.     int blownup;

  1630.     int num_audiop = 1;

  1631.     int num_assets = 1;

  1632.     int active_ss_mask[8];

  1633.     int i, j;

  1634. 

  1635.     if (get_bits_left(&s->gb) < 52)

  1636.         return;

  1637. 

  1638.     skip_bits(&s->gb, 8); // user data

  1639.     ss_index = get_bits(&s->gb, 2);

  1640. 

  1641.     blownup = get_bits1(&s->gb);

  1642.     skip_bits(&s->gb,  8 + 4 * blownup); // header_size

  1643.     skip_bits(&s->gb, 16 + 4 * blownup); // hd_size

  1644. 

  1645.     s->static_fields = get_bits1(&s->gb);

  1646.     if (s->static_fields) {

  1647.         skip_bits(&s->gb, 2); // reference clock code

  1648.         skip_bits(&s->gb, 3); // frame duration code

  1649. 

  1650.         if (get_bits1(&s->gb))

  1651.             skip_bits_long(&s->gb, 36); // timestamp

  1652. 

  1653.         /* a single stream can contain multiple audio assets that can be

  1654.          * combined to form multiple audio presentations */

  1655. 

  1656.         num_audiop = get_bits(&s->gb, 3) + 1;

  1657.         if (num_audiop > 1) {

  1658.             avpriv_request_sample(s->avctx,

  1659.                                   "Multiple DTS-HD audio presentations");

  1660.             /* ignore such streams for now */

  1661.             return;

  1662.         }

  1663. 

  1664.         num_assets = get_bits(&s->gb, 3) + 1;

  1665.         if (num_assets > 1) {

  1666.             avpriv_request_sample(s->avctx, "Multiple DTS-HD audio assets");

  1667.             /* ignore such streams for now */

  1668.             return;

  1669.         }

  1670. 

  1671.         for (i = 0; i < num_audiop; i++)

  1672.             active_ss_mask[i] = get_bits(&s->gb, ss_index + 1);

  1673. 

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

  1675.             for (j = 0; j <= ss_index; j++)

  1676.                 if (active_ss_mask[i] & (1 << j))

  1677.                     skip_bits(&s->gb, 8); // active asset mask

  1678. 

  1679.         s->mix_metadata = get_bits1(&s->gb);

  1680.         if (s->mix_metadata) {

  1681.             int mix_out_mask_size;

  1682. 

  1683.             skip_bits(&s->gb, 2); // adjustment level

  1684.             mix_out_mask_size  = (get_bits(&s->gb, 2) + 1) << 2;

  1685.             s->num_mix_configs =  get_bits(&s->gb, 2) + 1;

  1686. 

  1687.             for (i = 0; i < s->num_mix_configs; i++) {

  1688.                 int mix_out_mask        = get_bits(&s->gb, mix_out_mask_size);

  1689.                 s->mix_config_num_ch[i] = dca_exss_mask2count(mix_out_mask);

  1690.             }

  1691.         }

  1692.     }

  1693. 

  1694.     for (i = 0; i < num_assets; i++)

  1695.         skip_bits_long(&s->gb, 16 + 4 * blownup);  // asset size

  1696. 

  1697.     for (i = 0; i < num_assets; i++) {

  1698.         if (dca_exss_parse_asset_header(s))

  1699.             return;

  1700.     }

  1701. 

  1702.     /* not parsed further, we were only interested in the extensions mask

  1703.      * from the asset header */

  1704. }

  1705. 

  1706. /**

  1707.  * Main frame decoding function

  1708.  * FIXME add arguments

  1709.  */

  1710. static int dca_decode_frame(AVCodecContext *avctx, void *data,

  1711.                             int *got_frame_ptr, AVPacket *avpkt)

  1712. {

  1713.     AVFrame *frame     = data;

  1714.     const uint8_t *buf = avpkt->data;

  1715.     int buf_size = avpkt->size;

  1716. 

  1717.     int lfe_samples;

  1718.     int num_core_channels = 0;

  1719.     int i, ret;

  1720.     float  **samples_flt;

  1721.     DCAContext *s = avctx->priv_data;

  1722.     int channels, full_channels;

  1723.     int core_ss_end;

  1724. 

  1725. 

  1726.     s->xch_present = 0;

  1727. 

  1728.     s->dca_buffer_size = ff_dca_convert_bitstream(buf, buf_size, s->dca_buffer,

  1729.                                                   DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);

  1730.     if (s->dca_buffer_size == AVERROR_INVALIDDATA) {

  1731.         av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");

  1732.         return AVERROR_INVALIDDATA;

  1733.     }

  1734. 

  1735.     init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);

  1736.     if ((ret = dca_parse_frame_header(s)) < 0) {

  1737.         //seems like the frame is corrupt, try with the next one

  1738.         return ret;

  1739.     }

  1740.     //set AVCodec values with parsed data

  1741.     avctx->sample_rate = s->sample_rate;

  1742.     avctx->bit_rate    = s->bit_rate;

  1743. 

  1744.     s->profile = FF_PROFILE_DTS;

  1745. 

  1746.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1747.         if ((ret = dca_decode_block(s, 0, i))) {

  1748.             av_log(avctx, AV_LOG_ERROR, "error decoding block\n");

  1749.             return ret;

  1750.         }

  1751.     }

  1752. 

  1753.     /* record number of core channels incase less than max channels are requested */

  1754.     num_core_channels = s->prim_channels;

  1755. 

  1756.     if (s->ext_coding)

  1757.         s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];

  1758.     else

  1759.         s->core_ext_mask = 0;

  1760. 

  1761.     core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;

  1762. 

  1763.     /* only scan for extensions if ext_descr was unknown or indicated a

  1764.      * supported XCh extension */

  1765.     if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {

  1766. 

  1767.         /* if ext_descr was unknown, clear s->core_ext_mask so that the

  1768.          * extensions scan can fill it up */

  1769.         s->core_ext_mask = FFMAX(s->core_ext_mask, 0);

  1770. 

  1771.         /* extensions start at 32-bit boundaries into bitstream */

  1772.         skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1773. 

  1774.         while (core_ss_end - get_bits_count(&s->gb) >= 32) {

  1775.             uint32_t bits = get_bits_long(&s->gb, 32);

  1776. 

  1777.             switch (bits) {

  1778.             case 0x5a5a5a5a: {

  1779.                 int ext_amode, xch_fsize;

  1780. 

  1781.                 s->xch_base_channel = s->prim_channels;

  1782. 

  1783.                 /* validate sync word using XCHFSIZE field */

  1784.                 xch_fsize = show_bits(&s->gb, 10);

  1785.                 if ((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&

  1786.                     (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))

  1787.                     continue;

  1788. 

  1789.                 /* skip length-to-end-of-frame field for the moment */

  1790.                 skip_bits(&s->gb, 10);

  1791. 

  1792.                 s->core_ext_mask |= DCA_EXT_XCH;

  1793. 

  1794.                 /* extension amode(number of channels in extension) should be 1 */

  1795.                 /* AFAIK XCh is not used for more channels */

  1796.                 if ((ext_amode = get_bits(&s->gb, 4)) != 1) {

  1797.                     av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"

  1798.                            " supported!\n", ext_amode);

  1799.                     continue;

  1800.                 }

  1801. 

  1802.                 /* much like core primary audio coding header */

  1803.                 dca_parse_audio_coding_header(s, s->xch_base_channel);

  1804. 

  1805.                 for (i = 0; i < (s->sample_blocks / 8); i++)

  1806.                     if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {

  1807.                         av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");

  1808.                         continue;

  1809.                     }

  1810. 

  1811.                 s->xch_present = 1;

  1812.                 break;

  1813.             }

  1814.             case 0x47004a03:

  1815.                 /* XXCh: extended channels */

  1816.                 /* usually found either in core or HD part in DTS-HD HRA streams,

  1817.                  * but not in DTS-ES which contains XCh extensions instead */

  1818.                 s->core_ext_mask |= DCA_EXT_XXCH;

  1819.                 break;

  1820. 

  1821.             case 0x1d95f262: {

  1822.                 int fsize96 = show_bits(&s->gb, 12) + 1;

  1823.                 if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)

  1824.                     continue;

  1825. 

  1826.                 av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n",

  1827.                        get_bits_count(&s->gb));

  1828.                 skip_bits(&s->gb, 12);

  1829.                 av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);

  1830.                 av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));

  1831. 

  1832.                 s->core_ext_mask |= DCA_EXT_X96;

  1833.                 break;

  1834.             }

  1835.             }

  1836. 

  1837.             skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1838.         }

  1839.     } else {

  1840.         /* no supported extensions, skip the rest of the core substream */

  1841.         skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));

  1842.     }

  1843. 

  1844.     if (s->core_ext_mask & DCA_EXT_X96)

  1845.         s->profile = FF_PROFILE_DTS_96_24;

  1846.     else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))

  1847.         s->profile = FF_PROFILE_DTS_ES;

  1848. 

  1849.     /* check for ExSS (HD part) */

  1850.     if (s->dca_buffer_size - s->frame_size > 32 &&

  1851.         get_bits_long(&s->gb, 32) == DCA_HD_MARKER)

  1852.         dca_exss_parse_header(s);

  1853. 

  1854.     avctx->profile = s->profile;

  1855. 

  1856.     full_channels = channels = s->prim_channels + !!s->lfe;

  1857. 

  1858.     if (s->amode < 16) {

  1859.         avctx->channel_layout = dca_core_channel_layout[s->amode];

  1860. 

  1861. #if FF_API_REQUEST_CHANNELS

  1862. FF_DISABLE_DEPRECATION_WARNINGS

  1863.         if (s->xch_present && !s->xch_disable &&

  1864.             (!avctx->request_channels ||

  1865.              avctx->request_channels > num_core_channels + !!s->lfe)) {

  1866. FF_ENABLE_DEPRECATION_WARNINGS

  1867. #else

  1868.         if (s->xch_present && !s->xch_disable) {

  1869. #endif

  1870.             avctx->channel_layout |= AV_CH_BACK_CENTER;

  1871.             if (s->lfe) {

  1872.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1873.                 s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode];

  1874.             } else {

  1875.                 s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];

  1876.             }

  1877.         } else {

  1878.             channels = num_core_channels + !!s->lfe;

  1879.             s->xch_present = 0; /* disable further xch processing */

  1880.             if (s->lfe) {

  1881.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1882.                 s->channel_order_tab = dca_channel_reorder_lfe[s->amode];

  1883.             } else

  1884.                 s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];

  1885.         }

  1886. 

  1887.         if (channels > !!s->lfe &&

  1888.             s->channel_order_tab[channels - 1 - !!s->lfe] < 0)

  1889.             return AVERROR_INVALIDDATA;

  1890. 

  1891.         if (s->prim_channels + !!s->lfe > 2 &&

  1892.             avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {

  1893.             channels = 2;

  1894.             s->output = s->prim_channels == 2 ? s->amode : DCA_STEREO;

  1895.             avctx->channel_layout = AV_CH_LAYOUT_STEREO;

  1896. 

  1897.             /* Stereo downmix coefficients

  1898.              *

  1899.              * The decoder can only downmix to 2-channel, so we need to ensure

  1900.              * embedded downmix coefficients are actually targeting 2-channel.

  1901.              */

  1902.             if (s->core_downmix && (s->core_downmix_amode == DCA_STEREO ||

  1903.                                     s->core_downmix_amode == DCA_STEREO_TOTAL)) {

  1904.                 int sign, code;

  1905.                 for (i = 0; i < s->prim_channels + !!s->lfe; i++) {

  1906.                     sign = s->core_downmix_codes[i][0] & 0x100 ? 1 : -1;

  1907.                     code = s->core_downmix_codes[i][0] & 0x0FF;

  1908.                     s->downmix_coef[i][0] = (!code ? 0.0f :

  1909.                                              sign * dca_dmixtable[code - 1]);

  1910.                     sign = s->core_downmix_codes[i][1] & 0x100 ? 1 : -1;

  1911.                     code = s->core_downmix_codes[i][1] & 0x0FF;

  1912.                     s->downmix_coef[i][1] = (!code ? 0.0f :

  1913.                                              sign * dca_dmixtable[code - 1]);

  1914.                 }

  1915.                 s->output = s->core_downmix_amode;

  1916.             } else {

  1917.                 int am = s->amode & DCA_CHANNEL_MASK;

  1918.                 if (am >= FF_ARRAY_ELEMS(dca_default_coeffs)) {

  1919.                     av_log(s->avctx, AV_LOG_ERROR,

  1920.                            "Invalid channel mode %d\n", am);

  1921.                     return AVERROR_INVALIDDATA;

  1922.                 }

  1923.                 if (s->prim_channels + !!s->lfe >

  1924.                     FF_ARRAY_ELEMS(dca_default_coeffs[0])) {

  1925.                     avpriv_request_sample(s->avctx, "Downmixing %d channels",

  1926.                                           s->prim_channels + !!s->lfe);

  1927.                     return AVERROR_PATCHWELCOME;

  1928.                 }

  1929.                 for (i = 0; i < s->prim_channels + !!s->lfe; i++) {

  1930.                     s->downmix_coef[i][0] = dca_default_coeffs[am][i][0];

  1931.                     s->downmix_coef[i][1] = dca_default_coeffs[am][i][1];

  1932.                 }

  1933.             }

  1934.             av_dlog(s->avctx, "Stereo downmix coeffs:\n");

  1935.             for (i = 0; i < s->prim_channels + !!s->lfe; i++) {

  1936.                 av_dlog(s->avctx, "L, input channel %d = %f\n", i,

  1937.                         s->downmix_coef[i][0]);

  1938.                 av_dlog(s->avctx, "R, input channel %d = %f\n", i,

  1939.                         s->downmix_coef[i][1]);

  1940.             }

  1941.             av_dlog(s->avctx, "\n");

  1942.         }

  1943.     } else {

  1944.         av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);

  1945.         return AVERROR_INVALIDDATA;

  1946.     }

  1947.     avctx->channels = channels;

  1948. 

  1949.     /* get output buffer */

  1950.     frame->nb_samples = 256 * (s->sample_blocks / 8);

  1951.     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {

  1952.         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");

  1953.         return ret;

  1954.     }

  1955.     samples_flt = (float **)frame->extended_data;

  1956. 

  1957.     /* allocate buffer for extra channels if downmixing */

  1958.     if (avctx->channels < full_channels) {

  1959.         ret = av_samples_get_buffer_size(NULL, full_channels - channels,

  1960.                                          frame->nb_samples,

  1961.                                          avctx->sample_fmt, 0);

  1962.         if (ret < 0)

  1963.             return ret;

  1964. 

  1965.         av_fast_malloc(&s->extra_channels_buffer,

  1966.                        &s->extra_channels_buffer_size, ret);

  1967.         if (!s->extra_channels_buffer)

  1968.             return AVERROR(ENOMEM);

  1969. 

  1970.         ret = av_samples_fill_arrays((uint8_t **)s->extra_channels, NULL,

  1971.                                      s->extra_channels_buffer,

  1972.                                      full_channels - channels,

  1973.                                      frame->nb_samples, avctx->sample_fmt, 0);

  1974.         if (ret < 0)

  1975.             return ret;

  1976.     }

  1977. 

  1978.     /* filter to get final output */

  1979.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1980.         int ch;

  1981. 

  1982.         for (ch = 0; ch < channels; ch++)

  1983.             s->samples_chanptr[ch] = samples_flt[ch] + i * 256;

  1984.         for (; ch < full_channels; ch++)

  1985.             s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * 256;

  1986. 

  1987.         dca_filter_channels(s, i);

  1988. 

  1989.         /* If this was marked as a DTS-ES stream we need to subtract back- */

  1990.         /* channel from SL & SR to remove matrixed back-channel signal */

  1991.         if ((s->source_pcm_res & 1) && s->xch_present) {

  1992.             float *back_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel]];

  1993.             float *lt_chan   = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 2]];

  1994.             float *rt_chan   = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 1]];

  1995.             s->fdsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);

  1996.             s->fdsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);

  1997.         }

  1998.     }

  1999. 

  2000.     /* update lfe history */

  2001.     lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);

  2002.     for (i = 0; i < 2 * s->lfe * 4; i++)

  2003.         s->lfe_data[i] = s->lfe_data[i + lfe_samples];

  2004. 

  2005.     /* AVMatrixEncoding

  2006.      *

  2007.      * DCA_STEREO_TOTAL (Lt/Rt) is equivalent to Dolby Surround */

  2008.     ret = ff_side_data_update_matrix_encoding(frame,

  2009.                                               (s->output & ~DCA_LFE) == DCA_STEREO_TOTAL ?

  2010.                                               AV_MATRIX_ENCODING_DOLBY : AV_MATRIX_ENCODING_NONE);

  2011.     if (ret < 0)

  2012.         return ret;

  2013. 

  2014.     *got_frame_ptr = 1;

  2015. 

  2016.     return buf_size;

  2017. }

  2018. 

  2019. 

  2020. 

  2021. /**

  2022.  * DCA initialization

  2023.  *

  2024.  * @param avctx     pointer to the AVCodecContext

  2025.  */

  2026. 

  2027. static av_cold int dca_decode_init(AVCodecContext *avctx)

  2028. {

  2029.     DCAContext *s = avctx->priv_data;

  2030. 

  2031.     s->avctx = avctx;

  2032.     dca_init_vlcs();

  2033. 

  2034.     avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);

  2035.     ff_mdct_init(&s->imdct, 6, 1, 1.0);

  2036.     ff_synth_filter_init(&s->synth);

  2037.     ff_dcadsp_init(&s->dcadsp);

  2038.     ff_fmt_convert_init(&s->fmt_conv, avctx);

  2039. 

  2040.     avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

  2041. 

  2042.     /* allow downmixing to stereo */

  2043. #if FF_API_REQUEST_CHANNELS

  2044. FF_DISABLE_DEPRECATION_WARNINGS

  2045.     if (avctx->request_channels == 2)

  2046.         avctx->request_channel_layout = AV_CH_LAYOUT_STEREO;

  2047. FF_ENABLE_DEPRECATION_WARNINGS

  2048. #endif

  2049.     if (avctx->channels > 2 &&

  2050.         avctx->request_channel_layout == AV_CH_LAYOUT_STEREO)

  2051.         avctx->channels = 2;

  2052. 

  2053.     return 0;

  2054. }

  2055. 

  2056. static av_cold int dca_decode_end(AVCodecContext *avctx)

  2057. {

  2058.     DCAContext *s = avctx->priv_data;

  2059.     ff_mdct_end(&s->imdct);

  2060.     av_freep(&s->extra_channels_buffer);

  2061.     return 0;

  2062. }

  2063. 

  2064. static const AVProfile profiles[] = {

  2065.     { FF_PROFILE_DTS,        "DTS"        },

  2066.     { FF_PROFILE_DTS_ES,     "DTS-ES"     },

  2067.     { FF_PROFILE_DTS_96_24,  "DTS 96/24"  },

  2068.     { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },

  2069.     { FF_PROFILE_DTS_HD_MA,  "DTS-HD MA"  },

  2070.     { FF_PROFILE_UNKNOWN },

  2071. };

  2072. 

  2073. static const AVOption options[] = {

  2074.     { "disable_xch", "disable decoding of the XCh extension", offsetof(DCAContext, xch_disable), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM|AV_OPT_FLAG_AUDIO_PARAM },

  2075.     { NULL },

  2076. };

  2077. 

  2078. static const AVClass dca_decoder_class = {

  2079.     .class_name = "DCA decoder",

  2080.     .item_name  = av_default_item_name,

  2081.     .option     = options,

  2082.     .version    = LIBAVUTIL_VERSION_INT,

  2083. };

  2084. 

  2085. AVCodec ff_dca_decoder = {

  2086.     .name            = "dca",

  2087.     .long_name       = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),

  2088.     .type            = AVMEDIA_TYPE_AUDIO,

  2089.     .id              = AV_CODEC_ID_DTS,

  2090.     .priv_data_size  = sizeof(DCAContext),

  2091.     .init            = dca_decode_init,

  2092.     .decode          = dca_decode_frame,

  2093.     .close           = dca_decode_end,

  2094.     .capabilities    = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,

  2095.     .sample_fmts     = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,

  2096.                                                        AV_SAMPLE_FMT_NONE },

  2097.     .profiles        = NULL_IF_CONFIG_SMALL(profiles),

  2098.     .priv_class      = &dca_decoder_class,

  2099. };