falkTX
/
FFmpeg
mirror of https://github.com/falkTX/FFmpeg.git
1
0
Fork 0
Code Issues 0 Releases 338 Wiki Activity
You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
37311 Commits
32 Branches
802MB
Tree: 1fa24d08b2
Branches Tags
${ item.name }
Create branch ${ searchTerm }
from '1fa24d08b2'
${ noResults }
FFmpeg/libavcodec/dca.c

1989 lines
73KB
Raw Blame History 

  1. /*

  2.  * DCA compatible decoder

  3.  * Copyright (C) 2004 Gildas Bazin

  4.  * Copyright (C) 2004 Benjamin Zores

  5.  * Copyright (C) 2006 Benjamin Larsson

  6.  * Copyright (C) 2007 Konstantin Shishkov

  7.  *

  8.  * This file is part of FFmpeg.

  9.  *

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

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

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

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

  14.  *

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

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

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

  18.  * Lesser General Public License for more details.

  19.  *

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

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

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

  23.  */

  24. 

  25. #include <math.h>

  26. #include <stddef.h>

  27. #include <stdio.h>

  28. 

  29. #include "libavutil/common.h"

  30. #include "libavutil/intmath.h"

  31. #include "libavutil/intreadwrite.h"

  32. #include "libavutil/audioconvert.h"

  33. #include "avcodec.h"

  34. #include "dsputil.h"

  35. #include "fft.h"

  36. #include "get_bits.h"

  37. #include "put_bits.h"

  38. #include "dcadata.h"

  39. #include "dcahuff.h"

  40. #include "dca.h"

  41. #include "synth_filter.h"

  42. #include "dcadsp.h"

  43. #include "fmtconvert.h"

  44. 

  45. #if ARCH_ARM

  46. #   include "arm/dca.h"

  47. #endif

  48. 

  49. //#define TRACE

  50. 

  51. #define DCA_PRIM_CHANNELS_MAX  (7)

  52. #define DCA_SUBBANDS          (32)

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

  54. #define DCA_SUBSUBFRAMES_MAX   (4)

  55. #define DCA_SUBFRAMES_MAX     (16)

  56. #define DCA_BLOCKS_MAX        (16)

  57. #define DCA_LFE_MAX            (3)

  58. 

  59. enum DCAMode {

  60.     DCA_MONO = 0,

  61.     DCA_CHANNEL,

  62.     DCA_STEREO,

  63.     DCA_STEREO_SUMDIFF,

  64.     DCA_STEREO_TOTAL,

  65.     DCA_3F,

  66.     DCA_2F1R,

  67.     DCA_3F1R,

  68.     DCA_2F2R,

  69.     DCA_3F2R,

  70.     DCA_4F2R

  71. };

  72. 

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

  74. enum DCAExSSSpeakerMask {

  75.     DCA_EXSS_FRONT_CENTER          = 0x0001,

  76.     DCA_EXSS_FRONT_LEFT_RIGHT      = 0x0002,

  77.     DCA_EXSS_SIDE_REAR_LEFT_RIGHT  = 0x0004,

  78.     DCA_EXSS_LFE                   = 0x0008,

  79.     DCA_EXSS_REAR_CENTER           = 0x0010,

  80.     DCA_EXSS_FRONT_HIGH_LEFT_RIGHT = 0x0020,

  81.     DCA_EXSS_REAR_LEFT_RIGHT       = 0x0040,

  82.     DCA_EXSS_FRONT_HIGH_CENTER     = 0x0080,

  83.     DCA_EXSS_OVERHEAD              = 0x0100,

  84.     DCA_EXSS_CENTER_LEFT_RIGHT     = 0x0200,

  85.     DCA_EXSS_WIDE_LEFT_RIGHT       = 0x0400,

  86.     DCA_EXSS_SIDE_LEFT_RIGHT       = 0x0800,

  87.     DCA_EXSS_LFE2                  = 0x1000,

  88.     DCA_EXSS_SIDE_HIGH_LEFT_RIGHT  = 0x2000,

  89.     DCA_EXSS_REAR_HIGH_CENTER      = 0x4000,

  90.     DCA_EXSS_REAR_HIGH_LEFT_RIGHT  = 0x8000,

  91. };

  92. 

  93. enum DCAExtensionMask {

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

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

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

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

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

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

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

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

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

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

  104. };

  105. 

  106. /* -1 are reserved or unknown */

  107. static const int dca_ext_audio_descr_mask[] = {

  108.     DCA_EXT_XCH,

  109.     -1,

  110.     DCA_EXT_X96,

  111.     DCA_EXT_XCH | DCA_EXT_X96,

  112.     -1,

  113.     -1,

  114.     DCA_EXT_XXCH,

  115.     -1,

  116. };

  117. 

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

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

  120. 

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

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

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

  124.  *

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

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

  127.  * OV -> center back

  128.  * All 2 channel configurations -> AV_CH_LAYOUT_STEREO

  129.  */

  130. static const uint64_t dca_core_channel_layout[] = {

  131.     AV_CH_FRONT_CENTER,                                                     ///< 1, A

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

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

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

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

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

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

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

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

  140. 

  141.     AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_SIDE_LEFT |

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

  143. 

  144.     AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |

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

  146. 

  147.     AV_CH_LAYOUT_STEREO | AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT |

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

  149. 

  150.     AV_CH_FRONT_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |

  151.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_BACK_CENTER   |

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

  153. 

  154.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER   |

  155.     AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |

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

  157. 

  158.     AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |

  159.     AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |

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

  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_BACK_CENTER | AV_CH_SIDE_RIGHT,                 ///< 8, CL + C + CR + L + R + SL + S + SR

  165. };

  166. 

  167. static const int8_t dca_lfe_index[] = {

  168.     1, 2, 2, 2, 2, 3, 2, 3, 2, 3, 2, 3, 1, 3, 2, 3

  169. };

  170. 

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

  172.     { 0, -1, -1, -1, -1, -1, -1, -1, -1},

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

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

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

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

  177.     { 2,  0,  1, -1, -1, -1, -1, -1, -1},

  178.     { 0,  1,  3, -1, -1, -1, -1, -1, -1},

  179.     { 2,  0,  1,  4, -1, -1, -1, -1, -1},

  180.     { 0,  1,  3,  4, -1, -1, -1, -1, -1},

  181.     { 2,  0,  1,  4,  5, -1, -1, -1, -1},

  182.     { 3,  4,  0,  1,  5,  6, -1, -1, -1},

  183.     { 2,  0,  1,  4,  5,  6, -1, -1, -1},

  184.     { 0,  6,  4,  5,  2,  3, -1, -1, -1},

  185.     { 4,  2,  5,  0,  1,  6,  7, -1, -1},

  186.     { 5,  6,  0,  1,  7,  3,  8,  4, -1},

  187.     { 4,  2,  5,  0,  1,  6,  8,  7, -1},

  188. };

  189. 

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

  191.     { 0,  2, -1, -1, -1, -1, -1, -1, -1},

  192.     { 0,  1,  3, -1, -1, -1, -1, -1, -1},

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

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

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

  196.     { 2,  0,  1,  4, -1, -1, -1, -1, -1},

  197.     { 0,  1,  3,  4, -1, -1, -1, -1, -1},

  198.     { 2,  0,  1,  4,  5, -1, -1, -1, -1},

  199.     { 0,  1,  4,  5,  3, -1, -1, -1, -1},

  200.     { 2,  0,  1,  5,  6,  4, -1, -1, -1},

  201.     { 3,  4,  0,  1,  6,  7,  5, -1, -1},

  202.     { 2,  0,  1,  4,  5,  6,  7, -1, -1},

  203.     { 0,  6,  4,  5,  2,  3,  7, -1, -1},

  204.     { 4,  2,  5,  0,  1,  7,  8,  6, -1},

  205.     { 5,  6,  0,  1,  8,  3,  9,  4,  7},

  206.     { 4,  2,  5,  0,  1,  6,  9,  8,  7},

  207. };

  208. 

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

  210.     { 0, -1, -1, -1, -1, -1, -1, -1, -1},

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

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

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

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

  215.     { 2,  0,  1, -1, -1, -1, -1, -1, -1},

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

  217.     { 2,  0,  1,  3, -1, -1, -1, -1, -1},

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

  219.     { 2,  0,  1,  3,  4, -1, -1, -1, -1},

  220.     { 2,  3,  0,  1,  4,  5, -1, -1, -1},

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

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

  223.     { 3,  2,  4,  0,  1,  5,  6, -1, -1},

  224.     { 4,  5,  0,  1,  6,  2,  7,  3, -1},

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

  226. };

  227. 

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

  229.     { 0,  1, -1, -1, -1, -1, -1, -1, -1},

  230.     { 0,  1,  2, -1, -1, -1, -1, -1, -1},

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

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

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

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

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

  236.     { 2,  0,  1,  3,  4, -1, -1, -1, -1},

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

  238.     { 2,  0,  1,  4,  5,  3, -1, -1, -1},

  239.     { 2,  3,  0,  1,  5,  6,  4, -1, -1},

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

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

  242.     { 3,  2,  4,  0,  1,  6,  7,  5, -1},

  243.     { 4,  5,  0,  1,  7,  2,  8,  3,  6},

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

  245. };

  246. 

  247. #define DCA_DOLBY                  101           /* FIXME */

  248. 

  249. #define DCA_CHANNEL_BITS             6

  250. #define DCA_CHANNEL_MASK          0x3F

  251. 

  252. #define DCA_LFE                   0x80

  253. 

  254. #define HEADER_SIZE                 14

  255. 

  256. #define DCA_MAX_FRAME_SIZE       16384

  257. #define DCA_MAX_EXSS_HEADER_SIZE  4096

  258. 

  259. #define DCA_BUFFER_PADDING_SIZE   1024

  260. 

  261. /** Bit allocation */

  262. typedef struct {

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

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

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

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

  267. } BitAlloc;

  268. 

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

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

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

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

  273. 

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

  275.                                          int idx)

  276. {

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

  278.            ba->offset;

  279. }

  280. 

  281. typedef struct {

  282.     AVCodecContext *avctx;

  283.     AVFrame frame;

  284.     /* Frame header */

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

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

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

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

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

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

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

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

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

  294. 

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

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

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

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

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

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

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

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

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

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

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

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

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

  308.     int copy_history;           ///< copy history

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

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

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

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

  313. 

  314.     /* Primary audio coding header */

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

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

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

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

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

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

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

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

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

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

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

  326. 

  327.     /* Primary audio coding side information */

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

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

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

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

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

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

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

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

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

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

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

  339. 

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

  341. 

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

  343.     int lfe_scale_factor;

  344. 

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

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

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

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

  349.     int hist_index[DCA_PRIM_CHANNELS_MAX];

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

  351. 

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

  353.     float scale_bias;           ///< output scale

  354. 

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

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

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

  358. 

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

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

  361. 

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

  363.     GetBitContext gb;

  364.     /* Current position in DCA frame */

  365.     int current_subframe;

  366.     int current_subsubframe;

  367. 

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

  369. 

  370.     /* XCh extension information */

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

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

  373. 

  374.     /* ExSS header parser */

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

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

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

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

  379. 

  380.     int profile;

  381. 

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

  383.     DSPContext dsp;

  384.     FFTContext imdct;

  385.     SynthFilterContext synth;

  386.     DCADSPContext dcadsp;

  387.     FmtConvertContext fmt_conv;

  388. } DCAContext;

  389. 

  390. static const uint16_t dca_vlc_offs[] = {

  391.         0,   512,   640,   768,  1282,  1794,  2436,  3080,  3770,  4454,  5364,

  392.      5372,  5380,  5388,  5392,  5396,  5412,  5420,  5428,  5460,  5492,  5508,

  393.      5572,  5604,  5668,  5796,  5860,  5892,  6412,  6668,  6796,  7308,  7564,

  394.      7820,  8076,  8620,  9132,  9388,  9910, 10166, 10680, 11196, 11726, 12240,

  395.     12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264,

  396.     18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622,

  397. };

  398. 

  399. static av_cold void dca_init_vlcs(void)

  400. {

  401.     static int vlcs_initialized = 0;

  402.     int i, j, c = 14;

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

  404. 

  405.     if (vlcs_initialized)

  406.         return;

  407. 

  408.     dca_bitalloc_index.offset = 1;

  409.     dca_bitalloc_index.wrap = 2;

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

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

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

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

  414.                  bitalloc_12_bits[i], 1, 1,

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

  416.     }

  417.     dca_scalefactor.offset = -64;

  418.     dca_scalefactor.wrap = 2;

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

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

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

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

  423.                  scales_bits[i], 1, 1,

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

  425.     }

  426.     dca_tmode.offset = 0;

  427.     dca_tmode.wrap = 1;

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

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

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

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

  432.                  tmode_bits[i], 1, 1,

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

  434.     }

  435. 

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

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

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

  439.                 break;

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

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

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

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

  444. 

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

  446.                      bitalloc_sizes[i],

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

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

  449.             c++;

  450.         }

  451.     vlcs_initialized = 1;

  452. }

  453. 

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

  455. {

  456.     while (len--)

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

  458. }

  459. 

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

  461. {

  462.     int i, j;

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

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

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

  466. 

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

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

  469. 

  470.     if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)

  471.         s->prim_channels = DCA_PRIM_CHANNELS_MAX;

  472. 

  473. 

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

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

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

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

  478.     }

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

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

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

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

  483.     }

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

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

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

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

  488. 

  489.     /* Get codebooks quantization indexes */

  490.     if (!base_channel)

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

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

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

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

  495. 

  496.     /* Get scale factor adjustment */

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

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

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

  500. 

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

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

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

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

  505. 

  506.     if (s->crc_present) {

  507.         /* Audio header CRC check */

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

  509.     }

  510. 

  511.     s->current_subframe    = 0;

  512.     s->current_subsubframe = 0;

  513. 

  514. #ifdef TRACE

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

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

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

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

  519.                s->subband_activity[i]);

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

  521.                s->vq_start_subband[i]);

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

  523.                s->joint_intensity[i]);

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

  525.                s->transient_huffman[i]);

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

  527.                s->scalefactor_huffman[i]);

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

  529.                s->bitalloc_huffman[i]);

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

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

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

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

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

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

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

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

  538.     }

  539. #endif

  540. 

  541.     return 0;

  542. }

  543. 

  544. static int dca_parse_frame_header(DCAContext *s)

  545. {

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

  547. 

  548.     /* Sync code */

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

  550. 

  551.     /* Frame header */

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

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

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

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

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

  557.     if (s->frame_size < 95)

  558.         return AVERROR_INVALIDDATA;

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

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

  561.     if (!s->sample_rate)

  562.         return AVERROR_INVALIDDATA;

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

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

  565.     if (!s->bit_rate)

  566.         return AVERROR_INVALIDDATA;

  567. 

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

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

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

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

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

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

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

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

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

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

  578. 

  579.     /* TODO: check CRC */

  580.     if (s->crc_present)

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

  582. 

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

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

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

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

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

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

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

  590. 

  591.     /* FIXME: channels mixing levels */

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

  593.     if (s->lfe)

  594.         s->output |= DCA_LFE;

  595. 

  596. #ifdef TRACE

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

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

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

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

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

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

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

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

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

  606.            s->sample_rate);

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

  608.            s->bit_rate);

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

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

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

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

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

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

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

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

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

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

  619.            s->predictor_history);

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

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

  622.            s->multirate_inter);

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

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

  625.     av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

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

  632. #endif

  633. 

  634.     /* Primary audio coding header */

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

  636. 

  637.     return dca_parse_audio_coding_header(s, 0);

  638. }

  639. 

  640. 

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

  642. {

  643.     if (level < 5) {

  644.         /* huffman encoded */

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

  646.     } else if (level < 8)

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

  648.     return value;

  649. }

  650. 

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

  652. {

  653.     /* Primary audio coding side information */

  654.     int j, k;

  655. 

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

  657.         return AVERROR_INVALIDDATA;

  658. 

  659.     if (!base_channel) {

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

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

  662.     }

  663. 

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

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

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

  667.     }

  668. 

  669.     /* Get prediction codebook */

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

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

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

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

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

  675.             }

  676.         }

  677.     }

  678. 

  679.     /* Bit allocation index */

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

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

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

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

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

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

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

  687.                 av_log(s->avctx, AV_LOG_ERROR,

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

  689.                 return AVERROR_INVALIDDATA;

  690.             } else {

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

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

  693.             }

  694. 

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

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

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

  698.                 return AVERROR_INVALIDDATA;

  699.             }

  700.         }

  701.     }

  702. 

  703.     /* Transition mode */

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

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

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

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

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

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

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

  711.             }

  712.         }

  713.     }

  714. 

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

  716.         return AVERROR_INVALIDDATA;

  717. 

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

  719.         const uint32_t *scale_table;

  720.         int scale_sum;

  721. 

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

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

  724. 

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

  726.             scale_table = scale_factor_quant7;

  727.         else

  728.             scale_table = scale_factor_quant6;

  729. 

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

  731.         scale_sum = 0;

  732. 

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

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

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

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

  737.             }

  738. 

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

  740.                 /* Get second scale factor */

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

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

  743.             }

  744.         }

  745.     }

  746. 

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

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

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

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

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

  752.     }

  753. 

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

  755.         return AVERROR_INVALIDDATA;

  756. 

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

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

  759.         int source_channel;

  760. 

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

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

  763.             int scale = 0;

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

  765. 

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

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

  768. 

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

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

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

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

  773.             }

  774. 

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

  776.                 av_log(s->avctx, AV_LOG_DEBUG,

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

  778.                 s->debug_flag |= 0x02;

  779.             }

  780.         }

  781.     }

  782. 

  783.     /* Stereo downmix coefficients */

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

  785.         if (s->downmix) {

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

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

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

  789.             }

  790.         } else {

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

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

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

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

  795.             }

  796.         }

  797.     }

  798. 

  799.     /* Dynamic range coefficient */

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

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

  802. 

  803.     /* Side information CRC check word */

  804.     if (s->crc_present) {

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

  806.     }

  807. 

  808.     /*

  809.      * Primary audio data arrays

  810.      */

  811. 

  812.     /* VQ encoded high frequency subbands */

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

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

  815.             /* 1 vector -> 32 samples */

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

  817. 

  818.     /* Low frequency effect data */

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

  820.         /* LFE samples */

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

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

  823.         float lfe_scale;

  824. 

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

  826.             /* Signed 8 bits int */

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

  828.         }

  829. 

  830.         /* Scale factor index */

  831.         s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 8)];

  832. 

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

  834.         lfe_scale = 0.035 * s->lfe_scale_factor;

  835. 

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

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

  838.     }

  839. 

  840. #ifdef TRACE

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

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

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

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

  845. 

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

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

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

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

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

  851.     }

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

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

  854.             av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

  860.     }

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

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

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

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

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

  866.     }

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

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

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

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

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

  872.     }

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

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

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

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

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

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

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

  880.         }

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

  882.     }

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

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

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

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

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

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

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

  890.         }

  891.     }

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

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

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

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

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

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

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

  899.         }

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

  901.     }

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

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

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

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

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

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

  908. 

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

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

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

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

  913.     }

  914. #endif

  915. 

  916.     return 0;

  917. }

  918. 

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

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

  921.                             float scale)

  922. {

  923.     const float *prCoeff;

  924.     int i;

  925. 

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

  927.     int subindex;

  928. 

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

  930. 

  931.     /* Select filter */

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

  933.         prCoeff = fir_32bands_nonperfect;

  934.     else                        /* Perfect reconstruction */

  935.         prCoeff = fir_32bands_perfect;

  936. 

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

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

  939. 

  940.     /* Reconstructed channel sample index */

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

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

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

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

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

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

  947.         }

  948. 

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

  950.                                     s->subband_fir_hist[chans],

  951.                                     &s->hist_index[chans],

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

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

  954.         samples_out += 32;

  955.     }

  956. }

  957. 

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

  959.                                   int num_deci_sample, float *samples_in,

  960.                                   float *samples_out, float scale)

  961. {

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

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

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

  965.      *   from last subframe as history.

  966.      *

  967.      * samples_out: An array holding interpolated samples

  968.      */

  969. 

  970.     int decifactor;

  971.     const float *prCoeff;

  972.     int deciindex;

  973. 

  974.     /* Select decimation filter */

  975.     if (decimation_select == 1) {

  976.         decifactor = 64;

  977.         prCoeff = lfe_fir_128;

  978.     } else {

  979.         decifactor = 32;

  980.         prCoeff = lfe_fir_64;

  981.     }

  982.     /* Interpolation */

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

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

  985.         samples_in++;

  986.         samples_out += 2 * decifactor;

  987.     }

  988. }

  989. 

  990. /* downmixing routines */

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

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

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

  994. 

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

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

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

  998. 

  999. #define MIX_FRONT3(samples, coef)                                      \

  1000.     t = samples[i + c];                                                \

  1001.     u = samples[i + l];                                                \

  1002.     v = samples[i + r];                                                \

  1003.     samples[i]     = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \

  1004.     samples[i+256] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];

  1005. 

  1006. #define DOWNMIX_TO_STEREO(op1, op2)             \

  1007.     for (i = 0; i < 256; i++) {                 \

  1008.         op1                                     \

  1009.         op2                                     \

  1010.     }

  1011. 

  1012. static void dca_downmix(float *samples, int srcfmt,

  1013.                         int downmix_coef[DCA_PRIM_CHANNELS_MAX][2],

  1014.                         const int8_t *channel_mapping)

  1015. {

  1016.     int c, l, r, sl, sr, s;

  1017.     int i;

  1018.     float t, u, v;

  1019.     float coef[DCA_PRIM_CHANNELS_MAX][2];

  1020. 

  1021.     for (i = 0; i < DCA_PRIM_CHANNELS_MAX; i++) {

  1022.         coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]];

  1023.         coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]];

  1024.     }

  1025. 

  1026.     switch (srcfmt) {

  1027.     case DCA_MONO:

  1028.     case DCA_CHANNEL:

  1029.     case DCA_STEREO_TOTAL:

  1030.     case DCA_STEREO_SUMDIFF:

  1031.     case DCA_4F2R:

  1032.         av_log(NULL, 0, "Not implemented!\n");

  1033.         break;

  1034.     case DCA_STEREO:

  1035.         break;

  1036.     case DCA_3F:

  1037.         c = channel_mapping[0] * 256;

  1038.         l = channel_mapping[1] * 256;

  1039.         r = channel_mapping[2] * 256;

  1040.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );

  1041.         break;

  1042.     case DCA_2F1R:

  1043.         s = channel_mapping[2] * 256;

  1044.         DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + s, 2, coef), );

  1045.         break;

  1046.     case DCA_3F1R:

  1047.         c = channel_mapping[0] * 256;

  1048.         l = channel_mapping[1] * 256;

  1049.         r = channel_mapping[2] * 256;

  1050.         s = channel_mapping[3] * 256;

  1051.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1052.                           MIX_REAR1(samples, i + s, 3, coef));

  1053.         break;

  1054.     case DCA_2F2R:

  1055.         sl = channel_mapping[2] * 256;

  1056.         sr = channel_mapping[3] * 256;

  1057.         DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + sl, i + sr, 2, coef), );

  1058.         break;

  1059.     case DCA_3F2R:

  1060.         c  = channel_mapping[0] * 256;

  1061.         l  = channel_mapping[1] * 256;

  1062.         r  = channel_mapping[2] * 256;

  1063.         sl = channel_mapping[3] * 256;

  1064.         sr = channel_mapping[4] * 256;

  1065.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1066.                           MIX_REAR2(samples, i + sl, i + sr, 3, coef));

  1067.         break;

  1068.     }

  1069. }

  1070. 

  1071. 

  1072. #ifndef decode_blockcodes

  1073. /* Very compact version of the block code decoder that does not use table

  1074.  * look-up but is slightly slower */

  1075. static int decode_blockcode(int code, int levels, int *values)

  1076. {

  1077.     int i;

  1078.     int offset = (levels - 1) >> 1;

  1079. 

  1080.     for (i = 0; i < 4; i++) {

  1081.         int div = FASTDIV(code, levels);

  1082.         values[i] = code - offset - div * levels;

  1083.         code = div;

  1084.     }

  1085. 

  1086.     return code;

  1087. }

  1088. 

  1089. static int decode_blockcodes(int code1, int code2, int levels, int *values)

  1090. {

  1091.     return decode_blockcode(code1, levels, values) |

  1092.            decode_blockcode(code2, levels, values + 4);

  1093. }

  1094. #endif

  1095. 

  1096. static const uint8_t abits_sizes[7]  = { 7, 10, 12, 13, 15, 17, 19 };

  1097. static const uint8_t abits_levels[7] = { 3,  5,  7,  9, 13, 17, 25 };

  1098. 

  1099. #ifndef int8x8_fmul_int32

  1100. static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)

  1101. {

  1102.     float fscale = scale / 16.0;

  1103.     int i;

  1104.     for (i = 0; i < 8; i++)

  1105.         dst[i] = src[i] * fscale;

  1106. }

  1107. #endif

  1108. 

  1109. static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)

  1110. {

  1111.     int k, l;

  1112.     int subsubframe = s->current_subsubframe;

  1113. 

  1114.     const float *quant_step_table;

  1115. 

  1116.     /* FIXME */

  1117.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1118.     LOCAL_ALIGNED_16(int, block, [8]);

  1119. 

  1120.     /*

  1121.      * Audio data

  1122.      */

  1123. 

  1124.     /* Select quantization step size table */

  1125.     if (s->bit_rate_index == 0x1f)

  1126.         quant_step_table = lossless_quant_d;

  1127.     else

  1128.         quant_step_table = lossy_quant_d;

  1129. 

  1130.     for (k = base_channel; k < s->prim_channels; k++) {

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

  1132.             return AVERROR_INVALIDDATA;

  1133. 

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

  1135.             int m;

  1136. 

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

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

  1139. 

  1140.             float quant_step_size = quant_step_table[abits];

  1141. 

  1142.             /*

  1143.              * Determine quantization index code book and its type

  1144.              */

  1145. 

  1146.             /* Select quantization index code book */

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

  1148. 

  1149.             /*

  1150.              * Extract bits from the bit stream

  1151.              */

  1152.             if (!abits) {

  1153.                 memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));

  1154.             } else {

  1155.                 /* Deal with transients */

  1156.                 int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];

  1157.                 float rscale = quant_step_size * s->scale_factor[k][l][sfi] *

  1158.                                s->scalefactor_adj[k][sel];

  1159. 

  1160.                 if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {

  1161.                     if (abits <= 7) {

  1162.                         /* Block code */

  1163.                         int block_code1, block_code2, size, levels, err;

  1164. 

  1165.                         size   = abits_sizes[abits - 1];

  1166.                         levels = abits_levels[abits - 1];

  1167. 

  1168.                         block_code1 = get_bits(&s->gb, size);

  1169.                         block_code2 = get_bits(&s->gb, size);

  1170.                         err = decode_blockcodes(block_code1, block_code2,

  1171.                                                 levels, block);

  1172.                         if (err) {

  1173.                             av_log(s->avctx, AV_LOG_ERROR,

  1174.                                    "ERROR: block code look-up failed\n");

  1175.                             return AVERROR_INVALIDDATA;

  1176.                         }

  1177.                     } else {

  1178.                         /* no coding */

  1179.                         for (m = 0; m < 8; m++)

  1180.                             block[m] = get_sbits(&s->gb, abits - 3);

  1181.                     }

  1182.                 } else {

  1183.                     /* Huffman coded */

  1184.                     for (m = 0; m < 8; m++)

  1185.                         block[m] = get_bitalloc(&s->gb,

  1186.                                                 &dca_smpl_bitalloc[abits], sel);

  1187.                 }

  1188. 

  1189.                 s->fmt_conv.int32_to_float_fmul_scalar(subband_samples[k][l],

  1190.                                                        block, rscale, 8);

  1191.             }

  1192. 

  1193.             /*

  1194.              * Inverse ADPCM if in prediction mode

  1195.              */

  1196.             if (s->prediction_mode[k][l]) {

  1197.                 int n;

  1198.                 for (m = 0; m < 8; m++) {

  1199.                     for (n = 1; n <= 4; n++)

  1200.                         if (m >= n)

  1201.                             subband_samples[k][l][m] +=

  1202.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1203.                                  subband_samples[k][l][m - n] / 8192);

  1204.                         else if (s->predictor_history)

  1205.                             subband_samples[k][l][m] +=

  1206.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1207.                                  s->subband_samples_hist[k][l][m - n + 4] / 8192);

  1208.                 }

  1209.             }

  1210.         }

  1211. 

  1212.         /*

  1213.          * Decode VQ encoded high frequencies

  1214.          */

  1215.         for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {

  1216.             /* 1 vector -> 32 samples but we only need the 8 samples

  1217.              * for this subsubframe. */

  1218.             int hfvq = s->high_freq_vq[k][l];

  1219. 

  1220.             if (!s->debug_flag & 0x01) {

  1221.                 av_log(s->avctx, AV_LOG_DEBUG,

  1222.                        "Stream with high frequencies VQ coding\n");

  1223.                 s->debug_flag |= 0x01;

  1224.             }

  1225. 

  1226.             int8x8_fmul_int32(subband_samples[k][l],

  1227.                               &high_freq_vq[hfvq][subsubframe * 8],

  1228.                               s->scale_factor[k][l][0]);

  1229.         }

  1230.     }

  1231. 

  1232.     /* Check for DSYNC after subsubframe */

  1233.     if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {

  1234.         if (0xFFFF == get_bits(&s->gb, 16)) {   /* 0xFFFF */

  1235. #ifdef TRACE

  1236.             av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");

  1237. #endif

  1238.         } else {

  1239.             av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");

  1240.         }

  1241.     }

  1242. 

  1243.     /* Backup predictor history for adpcm */

  1244.     for (k = base_channel; k < s->prim_channels; k++)

  1245.         for (l = 0; l < s->vq_start_subband[k]; l++)

  1246.             memcpy(s->subband_samples_hist[k][l],

  1247.                    &subband_samples[k][l][4],

  1248.                    4 * sizeof(subband_samples[0][0][0]));

  1249. 

  1250.     return 0;

  1251. }

  1252. 

  1253. static int dca_filter_channels(DCAContext *s, int block_index)

  1254. {

  1255.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1256.     int k;

  1257. 

  1258.     /* 32 subbands QMF */

  1259.     for (k = 0; k < s->prim_channels; k++) {

  1260. /*        static float pcm_to_double[8] = { 32768.0, 32768.0, 524288.0, 524288.0,

  1261.                                             0, 8388608.0, 8388608.0 };*/

  1262.         qmf_32_subbands(s, k, subband_samples[k],

  1263.                         &s->samples[256 * s->channel_order_tab[k]],

  1264.                         M_SQRT1_2 * s->scale_bias /* pcm_to_double[s->source_pcm_res] */);

  1265.     }

  1266. 

  1267.     /* Down mixing */

  1268.     if (s->avctx->request_channels == 2 && s->prim_channels > 2) {

  1269.         dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab);

  1270.     }

  1271. 

  1272.     /* Generate LFE samples for this subsubframe FIXME!!! */

  1273.     if (s->output & DCA_LFE) {

  1274.         lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,

  1275.                               s->lfe_data + 2 * s->lfe * (block_index + 4),

  1276.                               &s->samples[256 * dca_lfe_index[s->amode]],

  1277.                               (1.0 / 256.0) * s->scale_bias);

  1278.         /* Outputs 20bits pcm samples */

  1279.     }

  1280. 

  1281.     return 0;

  1282. }

  1283. 

  1284. 

  1285. static int dca_subframe_footer(DCAContext *s, int base_channel)

  1286. {

  1287.     int aux_data_count = 0, i;

  1288. 

  1289.     /*

  1290.      * Unpack optional information

  1291.      */

  1292. 

  1293.     /* presumably optional information only appears in the core? */

  1294.     if (!base_channel) {

  1295.         if (s->timestamp)

  1296.             skip_bits_long(&s->gb, 32);

  1297. 

  1298.         if (s->aux_data)

  1299.             aux_data_count = get_bits(&s->gb, 6);

  1300. 

  1301.         for (i = 0; i < aux_data_count; i++)

  1302.             get_bits(&s->gb, 8);

  1303. 

  1304.         if (s->crc_present && (s->downmix || s->dynrange))

  1305.             get_bits(&s->gb, 16);

  1306.     }

  1307. 

  1308.     return 0;

  1309. }

  1310. 

  1311. /**

  1312.  * Decode a dca frame block

  1313.  *

  1314.  * @param s     pointer to the DCAContext

  1315.  */

  1316. 

  1317. static int dca_decode_block(DCAContext *s, int base_channel, int block_index)

  1318. {

  1319.     int ret;

  1320. 

  1321.     /* Sanity check */

  1322.     if (s->current_subframe >= s->subframes) {

  1323.         av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",

  1324.                s->current_subframe, s->subframes);

  1325.         return AVERROR_INVALIDDATA;

  1326.     }

  1327. 

  1328.     if (!s->current_subsubframe) {

  1329. #ifdef TRACE

  1330.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");

  1331. #endif

  1332.         /* Read subframe header */

  1333.         if ((ret = dca_subframe_header(s, base_channel, block_index)))

  1334.             return ret;

  1335.     }

  1336. 

  1337.     /* Read subsubframe */

  1338. #ifdef TRACE

  1339.     av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");

  1340. #endif

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

  1342.         return ret;

  1343. 

  1344.     /* Update state */

  1345.     s->current_subsubframe++;

  1346.     if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {

  1347.         s->current_subsubframe = 0;

  1348.         s->current_subframe++;

  1349.     }

  1350.     if (s->current_subframe >= s->subframes) {

  1351. #ifdef TRACE

  1352.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");

  1353. #endif

  1354.         /* Read subframe footer */

  1355.         if ((ret = dca_subframe_footer(s, base_channel)))

  1356.             return ret;

  1357.     }

  1358. 

  1359.     return 0;

  1360. }

  1361. 

  1362. /**

  1363.  * Convert bitstream to one representation based on sync marker

  1364.  */

  1365. static int dca_convert_bitstream(const uint8_t *src, int src_size, uint8_t *dst,

  1366.                                  int max_size)

  1367. {

  1368.     uint32_t mrk;

  1369.     int i, tmp;

  1370.     const uint16_t *ssrc = (const uint16_t *) src;

  1371.     uint16_t *sdst = (uint16_t *) dst;

  1372.     PutBitContext pb;

  1373. 

  1374.     if ((unsigned) src_size > (unsigned) max_size) {

  1375. //        av_log(NULL, AV_LOG_ERROR, "Input frame size larger than DCA_MAX_FRAME_SIZE!\n");

  1376. //        return -1;

  1377.         src_size = max_size;

  1378.     }

  1379. 

  1380.     mrk = AV_RB32(src);

  1381.     switch (mrk) {

  1382.     case DCA_MARKER_RAW_BE:

  1383.         memcpy(dst, src, src_size);

  1384.         return src_size;

  1385.     case DCA_MARKER_RAW_LE:

  1386.         for (i = 0; i < (src_size + 1) >> 1; i++)

  1387.             *sdst++ = av_bswap16(*ssrc++);

  1388.         return src_size;

  1389.     case DCA_MARKER_14B_BE:

  1390.     case DCA_MARKER_14B_LE:

  1391.         init_put_bits(&pb, dst, max_size);

  1392.         for (i = 0; i < (src_size + 1) >> 1; i++, src += 2) {

  1393.             tmp = ((mrk == DCA_MARKER_14B_BE) ? AV_RB16(src) : AV_RL16(src)) & 0x3FFF;

  1394.             put_bits(&pb, 14, tmp);

  1395.         }

  1396.         flush_put_bits(&pb);

  1397.         return (put_bits_count(&pb) + 7) >> 3;

  1398.     default:

  1399.         return AVERROR_INVALIDDATA;

  1400.     }

  1401. }

  1402. 

  1403. /**

  1404.  * Return the number of channels in an ExSS speaker mask (HD)

  1405.  */

  1406. static int dca_exss_mask2count(int mask)

  1407. {

  1408.     /* count bits that mean speaker pairs twice */

  1409.     return av_popcount(mask) +

  1410.            av_popcount(mask & (DCA_EXSS_CENTER_LEFT_RIGHT      |

  1411.                                DCA_EXSS_FRONT_LEFT_RIGHT       |

  1412.                                DCA_EXSS_FRONT_HIGH_LEFT_RIGHT  |

  1413.                                DCA_EXSS_WIDE_LEFT_RIGHT        |

  1414.                                DCA_EXSS_SIDE_LEFT_RIGHT        |

  1415.                                DCA_EXSS_SIDE_HIGH_LEFT_RIGHT   |

  1416.                                DCA_EXSS_SIDE_REAR_LEFT_RIGHT   |

  1417.                                DCA_EXSS_REAR_LEFT_RIGHT        |

  1418.                                DCA_EXSS_REAR_HIGH_LEFT_RIGHT));

  1419. }

  1420. 

  1421. /**

  1422.  * Skip mixing coefficients of a single mix out configuration (HD)

  1423.  */

  1424. static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch)

  1425. {

  1426.     int i;

  1427. 

  1428.     for (i = 0; i < channels; i++) {

  1429.         int mix_map_mask = get_bits(gb, out_ch);

  1430.         int num_coeffs = av_popcount(mix_map_mask);

  1431.         skip_bits_long(gb, num_coeffs * 6);

  1432.     }

  1433. }

  1434. 

  1435. /**

  1436.  * Parse extension substream asset header (HD)

  1437.  */

  1438. static int dca_exss_parse_asset_header(DCAContext *s)

  1439. {

  1440.     int header_pos = get_bits_count(&s->gb);

  1441.     int header_size;

  1442.     int channels = 0;

  1443.     int embedded_stereo = 0;

  1444.     int embedded_6ch    = 0;

  1445.     int drc_code_present;

  1446.     int av_uninit(extensions_mask);

  1447.     int i, j;

  1448. 

  1449.     if (get_bits_left(&s->gb) < 16)

  1450.         return -1;

  1451. 

  1452.     /* We will parse just enough to get to the extensions bitmask with which

  1453.      * we can set the profile value. */

  1454. 

  1455.     header_size = get_bits(&s->gb, 9) + 1;

  1456.     skip_bits(&s->gb, 3); // asset index

  1457. 

  1458.     if (s->static_fields) {

  1459.         if (get_bits1(&s->gb))

  1460.             skip_bits(&s->gb, 4); // asset type descriptor

  1461.         if (get_bits1(&s->gb))

  1462.             skip_bits_long(&s->gb, 24); // language descriptor

  1463. 

  1464.         if (get_bits1(&s->gb)) {

  1465.             /* How can one fit 1024 bytes of text here if the maximum value

  1466.              * for the asset header size field above was 512 bytes? */

  1467.             int text_length = get_bits(&s->gb, 10) + 1;

  1468.             if (get_bits_left(&s->gb) < text_length * 8)

  1469.                 return -1;

  1470.             skip_bits_long(&s->gb, text_length * 8); // info text

  1471.         }

  1472. 

  1473.         skip_bits(&s->gb, 5); // bit resolution - 1

  1474.         skip_bits(&s->gb, 4); // max sample rate code

  1475.         channels = get_bits(&s->gb, 8) + 1;

  1476. 

  1477.         if (get_bits1(&s->gb)) { // 1-to-1 channels to speakers

  1478.             int spkr_remap_sets;

  1479.             int spkr_mask_size = 16;

  1480.             int num_spkrs[7];

  1481. 

  1482.             if (channels > 2)

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

  1484.             if (channels > 6)

  1485.                 embedded_6ch = get_bits1(&s->gb);

  1486. 

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

  1488.                 spkr_mask_size = (get_bits(&s->gb, 2) + 1) << 2;

  1489.                 skip_bits(&s->gb, spkr_mask_size); // spkr activity mask

  1490.             }

  1491. 

  1492.             spkr_remap_sets = get_bits(&s->gb, 3);

  1493. 

  1494.             for (i = 0; i < spkr_remap_sets; i++) {

  1495.                 /* std layout mask for each remap set */

  1496.                 num_spkrs[i] = dca_exss_mask2count(get_bits(&s->gb, spkr_mask_size));

  1497.             }

  1498. 

  1499.             for (i = 0; i < spkr_remap_sets; i++) {

  1500.                 int num_dec_ch_remaps = get_bits(&s->gb, 5) + 1;

  1501.                 if (get_bits_left(&s->gb) < 0)

  1502.                     return -1;

  1503. 

  1504.                 for (j = 0; j < num_spkrs[i]; j++) {

  1505.                     int remap_dec_ch_mask = get_bits_long(&s->gb, num_dec_ch_remaps);

  1506.                     int num_dec_ch = av_popcount(remap_dec_ch_mask);

  1507.                     skip_bits_long(&s->gb, num_dec_ch * 5); // remap codes

  1508.                 }

  1509.             }

  1510. 

  1511.         } else {

  1512.             skip_bits(&s->gb, 3); // representation type

  1513.         }

  1514.     }

  1515. 

  1516.     drc_code_present = get_bits1(&s->gb);

  1517.     if (drc_code_present)

  1518.         get_bits(&s->gb, 8); // drc code

  1519. 

  1520.     if (get_bits1(&s->gb))

  1521.         skip_bits(&s->gb, 5); // dialog normalization code

  1522. 

  1523.     if (drc_code_present && embedded_stereo)

  1524.         get_bits(&s->gb, 8); // drc stereo code

  1525. 

  1526.     if (s->mix_metadata && get_bits1(&s->gb)) {

  1527.         skip_bits(&s->gb, 1); // external mix

  1528.         skip_bits(&s->gb, 6); // post mix gain code

  1529. 

  1530.         if (get_bits(&s->gb, 2) != 3) // mixer drc code

  1531.             skip_bits(&s->gb, 3); // drc limit

  1532.         else

  1533.             skip_bits(&s->gb, 8); // custom drc code

  1534. 

  1535.         if (get_bits1(&s->gb)) // channel specific scaling

  1536.             for (i = 0; i < s->num_mix_configs; i++)

  1537.                 skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6); // scale codes

  1538.         else

  1539.             skip_bits_long(&s->gb, s->num_mix_configs * 6); // scale codes

  1540. 

  1541.         for (i = 0; i < s->num_mix_configs; i++) {

  1542.             if (get_bits_left(&s->gb) < 0)

  1543.                 return -1;

  1544.             dca_exss_skip_mix_coeffs(&s->gb, channels, s->mix_config_num_ch[i]);

  1545.             if (embedded_6ch)

  1546.                 dca_exss_skip_mix_coeffs(&s->gb, 6, s->mix_config_num_ch[i]);

  1547.             if (embedded_stereo)

  1548.                 dca_exss_skip_mix_coeffs(&s->gb, 2, s->mix_config_num_ch[i]);

  1549.         }

  1550.     }

  1551. 

  1552.     switch (get_bits(&s->gb, 2)) {

  1553.     case 0: extensions_mask = get_bits(&s->gb, 12); break;

  1554.     case 1: extensions_mask = DCA_EXT_EXSS_XLL;     break;

  1555.     case 2: extensions_mask = DCA_EXT_EXSS_LBR;     break;

  1556.     case 3: extensions_mask = 0; /* aux coding */   break;

  1557.     }

  1558. 

  1559.     /* not parsed further, we were only interested in the extensions mask */

  1560. 

  1561.     if (get_bits_left(&s->gb) < 0)

  1562.         return -1;

  1563. 

  1564.     if (get_bits_count(&s->gb) - header_pos > header_size * 8) {

  1565.         av_log(s->avctx, AV_LOG_WARNING, "Asset header size mismatch.\n");

  1566.         return -1;

  1567.     }

  1568.     skip_bits_long(&s->gb, header_pos + header_size * 8 - get_bits_count(&s->gb));

  1569. 

  1570.     if (extensions_mask & DCA_EXT_EXSS_XLL)

  1571.         s->profile = FF_PROFILE_DTS_HD_MA;

  1572.     else if (extensions_mask & (DCA_EXT_EXSS_XBR | DCA_EXT_EXSS_X96 |

  1573.                                 DCA_EXT_EXSS_XXCH))

  1574.         s->profile = FF_PROFILE_DTS_HD_HRA;

  1575. 

  1576.     if (!(extensions_mask & DCA_EXT_CORE))

  1577.         av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n");

  1578.     if ((extensions_mask & DCA_CORE_EXTS) != s->core_ext_mask)

  1579.         av_log(s->avctx, AV_LOG_WARNING,

  1580.                "DTS extensions detection mismatch (%d, %d)\n",

  1581.                extensions_mask & DCA_CORE_EXTS, s->core_ext_mask);

  1582. 

  1583.     return 0;

  1584. }

  1585. 

  1586. /**

  1587.  * Parse extension substream header (HD)

  1588.  */

  1589. static void dca_exss_parse_header(DCAContext *s)

  1590. {

  1591.     int ss_index;

  1592.     int blownup;

  1593.     int num_audiop = 1;

  1594.     int num_assets = 1;

  1595.     int active_ss_mask[8];

  1596.     int i, j;

  1597. 

  1598.     if (get_bits_left(&s->gb) < 52)

  1599.         return;

  1600. 

  1601.     skip_bits(&s->gb, 8); // user data

  1602.     ss_index = get_bits(&s->gb, 2);

  1603. 

  1604.     blownup = get_bits1(&s->gb);

  1605.     skip_bits(&s->gb,  8 + 4 * blownup); // header_size

  1606.     skip_bits(&s->gb, 16 + 4 * blownup); // hd_size

  1607. 

  1608.     s->static_fields = get_bits1(&s->gb);

  1609.     if (s->static_fields) {

  1610.         skip_bits(&s->gb, 2); // reference clock code

  1611.         skip_bits(&s->gb, 3); // frame duration code

  1612. 

  1613.         if (get_bits1(&s->gb))

  1614.             skip_bits_long(&s->gb, 36); // timestamp

  1615. 

  1616.         /* a single stream can contain multiple audio assets that can be

  1617.          * combined to form multiple audio presentations */

  1618. 

  1619.         num_audiop = get_bits(&s->gb, 3) + 1;

  1620.         if (num_audiop > 1) {

  1621.             av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio presentations.");

  1622.             /* ignore such streams for now */

  1623.             return;

  1624.         }

  1625. 

  1626.         num_assets = get_bits(&s->gb, 3) + 1;

  1627.         if (num_assets > 1) {

  1628.             av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio assets.");

  1629.             /* ignore such streams for now */

  1630.             return;

  1631.         }

  1632. 

  1633.         for (i = 0; i < num_audiop; i++)

  1634.             active_ss_mask[i] = get_bits(&s->gb, ss_index + 1);

  1635. 

  1636.         for (i = 0; i < num_audiop; i++)

  1637.             for (j = 0; j <= ss_index; j++)

  1638.                 if (active_ss_mask[i] & (1 << j))

  1639.                     skip_bits(&s->gb, 8); // active asset mask

  1640. 

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

  1642.         if (s->mix_metadata) {

  1643.             int mix_out_mask_size;

  1644. 

  1645.             skip_bits(&s->gb, 2); // adjustment level

  1646.             mix_out_mask_size  = (get_bits(&s->gb, 2) + 1) << 2;

  1647.             s->num_mix_configs =  get_bits(&s->gb, 2) + 1;

  1648. 

  1649.             for (i = 0; i < s->num_mix_configs; i++) {

  1650.                 int mix_out_mask        = get_bits(&s->gb, mix_out_mask_size);

  1651.                 s->mix_config_num_ch[i] = dca_exss_mask2count(mix_out_mask);

  1652.             }

  1653.         }

  1654.     }

  1655. 

  1656.     for (i = 0; i < num_assets; i++)

  1657.         skip_bits_long(&s->gb, 16 + 4 * blownup);  // asset size

  1658. 

  1659.     for (i = 0; i < num_assets; i++) {

  1660.         if (dca_exss_parse_asset_header(s))

  1661.             return;

  1662.     }

  1663. 

  1664.     /* not parsed further, we were only interested in the extensions mask

  1665.      * from the asset header */

  1666. }

  1667. 

  1668. /**

  1669.  * Main frame decoding function

  1670.  * FIXME add arguments

  1671.  */

  1672. static int dca_decode_frame(AVCodecContext *avctx, void *data,

  1673.                             int *got_frame_ptr, AVPacket *avpkt)

  1674. {

  1675.     const uint8_t *buf = avpkt->data;

  1676.     int buf_size = avpkt->size;

  1677. 

  1678.     int lfe_samples;

  1679.     int num_core_channels = 0;

  1680.     int i, ret;

  1681.     float   *samples_flt;

  1682.     int16_t *samples_s16;

  1683.     DCAContext *s = avctx->priv_data;

  1684.     int channels;

  1685.     int core_ss_end;

  1686. 

  1687. 

  1688.     s->xch_present = 0;

  1689. 

  1690.     s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer,

  1691.                                                DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);

  1692.     if (s->dca_buffer_size == AVERROR_INVALIDDATA) {

  1693.         av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");

  1694.         return AVERROR_INVALIDDATA;

  1695.     }

  1696. 

  1697.     init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);

  1698.     if ((ret = dca_parse_frame_header(s)) < 0) {

  1699.         //seems like the frame is corrupt, try with the next one

  1700.         return ret;

  1701.     }

  1702.     //set AVCodec values with parsed data

  1703.     avctx->sample_rate = s->sample_rate;

  1704.     avctx->bit_rate    = s->bit_rate;

  1705.     avctx->frame_size  = s->sample_blocks * 32;

  1706. 

  1707.     s->profile = FF_PROFILE_DTS;

  1708. 

  1709.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1710.         if ((ret = dca_decode_block(s, 0, i))) {

  1711.             av_log(avctx, AV_LOG_ERROR, "error decoding block\n");

  1712.             return ret;

  1713.         }

  1714.     }

  1715. 

  1716.     /* record number of core channels incase less than max channels are requested */

  1717.     num_core_channels = s->prim_channels;

  1718. 

  1719.     if (s->ext_coding)

  1720.         s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];

  1721.     else

  1722.         s->core_ext_mask = 0;

  1723. 

  1724.     core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;

  1725. 

  1726.     /* only scan for extensions if ext_descr was unknown or indicated a

  1727.      * supported XCh extension */

  1728.     if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {

  1729. 

  1730.         /* if ext_descr was unknown, clear s->core_ext_mask so that the

  1731.          * extensions scan can fill it up */

  1732.         s->core_ext_mask = FFMAX(s->core_ext_mask, 0);

  1733. 

  1734.         /* extensions start at 32-bit boundaries into bitstream */

  1735.         skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1736. 

  1737.         while (core_ss_end - get_bits_count(&s->gb) >= 32) {

  1738.             uint32_t bits = get_bits_long(&s->gb, 32);

  1739. 

  1740.             switch (bits) {

  1741.             case 0x5a5a5a5a: {

  1742.                 int ext_amode, xch_fsize;

  1743. 

  1744.                 s->xch_base_channel = s->prim_channels;

  1745. 

  1746.                 /* validate sync word using XCHFSIZE field */

  1747.                 xch_fsize = show_bits(&s->gb, 10);

  1748.                 if ((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&

  1749.                     (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))

  1750.                     continue;

  1751. 

  1752.                 /* skip length-to-end-of-frame field for the moment */

  1753.                 skip_bits(&s->gb, 10);

  1754. 

  1755.                 s->core_ext_mask |= DCA_EXT_XCH;

  1756. 

  1757.                 /* extension amode(number of channels in extension) should be 1 */

  1758.                 /* AFAIK XCh is not used for more channels */

  1759.                 if ((ext_amode = get_bits(&s->gb, 4)) != 1) {

  1760.                     av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"

  1761.                            " supported!\n", ext_amode);

  1762.                     continue;

  1763.                 }

  1764. 

  1765.                 /* much like core primary audio coding header */

  1766.                 dca_parse_audio_coding_header(s, s->xch_base_channel);

  1767. 

  1768.                 for (i = 0; i < (s->sample_blocks / 8); i++)

  1769.                     if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {

  1770.                         av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");

  1771.                         continue;

  1772.                     }

  1773. 

  1774.                 s->xch_present = 1;

  1775.                 break;

  1776.             }

  1777.             case 0x47004a03:

  1778.                 /* XXCh: extended channels */

  1779.                 /* usually found either in core or HD part in DTS-HD HRA streams,

  1780.                  * but not in DTS-ES which contains XCh extensions instead */

  1781.                 s->core_ext_mask |= DCA_EXT_XXCH;

  1782.                 break;

  1783. 

  1784.             case 0x1d95f262: {

  1785.                 int fsize96 = show_bits(&s->gb, 12) + 1;

  1786.                 if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)

  1787.                     continue;

  1788. 

  1789.                 av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n",

  1790.                        get_bits_count(&s->gb));

  1791.                 skip_bits(&s->gb, 12);

  1792.                 av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);

  1793.                 av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));

  1794. 

  1795.                 s->core_ext_mask |= DCA_EXT_X96;

  1796.                 break;

  1797.             }

  1798.             }

  1799. 

  1800.             skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1801.         }

  1802.     } else {

  1803.         /* no supported extensions, skip the rest of the core substream */

  1804.         skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));

  1805.     }

  1806. 

  1807.     if (s->core_ext_mask & DCA_EXT_X96)

  1808.         s->profile = FF_PROFILE_DTS_96_24;

  1809.     else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))

  1810.         s->profile = FF_PROFILE_DTS_ES;

  1811. 

  1812.     /* check for ExSS (HD part) */

  1813.     if (s->dca_buffer_size - s->frame_size > 32 &&

  1814.         get_bits_long(&s->gb, 32) == DCA_HD_MARKER)

  1815.         dca_exss_parse_header(s);

  1816. 

  1817.     avctx->profile = s->profile;

  1818. 

  1819.     channels = s->prim_channels + !!s->lfe;

  1820. 

  1821.     if (s->amode < 16) {

  1822.         avctx->channel_layout = dca_core_channel_layout[s->amode];

  1823. 

  1824.         if (s->xch_present && (!avctx->request_channels ||

  1825.                                avctx->request_channels > num_core_channels + !!s->lfe)) {

  1826.             avctx->channel_layout |= AV_CH_BACK_CENTER;

  1827.             if (s->lfe) {

  1828.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1829.                 s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode];

  1830.             } else {

  1831.                 s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];

  1832.             }

  1833.         } else {

  1834.             channels = num_core_channels + !!s->lfe;

  1835.             s->xch_present = 0; /* disable further xch processing */

  1836.             if (s->lfe) {

  1837.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1838.                 s->channel_order_tab = dca_channel_reorder_lfe[s->amode];

  1839.             } else

  1840.                 s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];

  1841.         }

  1842. 

  1843.         if (channels > !!s->lfe &&

  1844.             s->channel_order_tab[channels - 1 - !!s->lfe] < 0)

  1845.             return AVERROR_INVALIDDATA;

  1846. 

  1847.         if (avctx->request_channels == 2 && s->prim_channels > 2) {

  1848.             channels = 2;

  1849.             s->output = DCA_STEREO;

  1850.             avctx->channel_layout = AV_CH_LAYOUT_STEREO;

  1851.         }

  1852.         else if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) {

  1853.             static const int8_t dca_channel_order_native[9] = { 0, 1, 2, 3, 4, 5, 6, 7, 8 };

  1854.             s->channel_order_tab = dca_channel_order_native;

  1855.         }

  1856.     } else {

  1857.         av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);

  1858.         return AVERROR_INVALIDDATA;

  1859.     }

  1860. 

  1861.     if (avctx->channels != channels) {

  1862.         if (avctx->channels)

  1863.             av_log(avctx, AV_LOG_INFO, "Number of channels changed in DCA decoder (%d -> %d)\n", avctx->channels, channels);

  1864.         avctx->channels = channels;

  1865.     }

  1866. 

  1867.     /* get output buffer */

  1868.     s->frame.nb_samples = 256 * (s->sample_blocks / 8);

  1869.     if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {

  1870.         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");

  1871.         return ret;

  1872.     }

  1873.     samples_flt = (float *)   s->frame.data[0];

  1874.     samples_s16 = (int16_t *) s->frame.data[0];

  1875. 

  1876.     /* filter to get final output */

  1877.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1878.         dca_filter_channels(s, i);

  1879. 

  1880.         /* If this was marked as a DTS-ES stream we need to subtract back- */

  1881.         /* channel from SL & SR to remove matrixed back-channel signal */

  1882.         if ((s->source_pcm_res & 1) && s->xch_present) {

  1883.             float *back_chan = s->samples + s->channel_order_tab[s->xch_base_channel]     * 256;

  1884.             float *lt_chan   = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256;

  1885.             float *rt_chan   = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;

  1886.             s->dsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);

  1887.             s->dsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);

  1888.         }

  1889. 

  1890.         if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {

  1891.             s->fmt_conv.float_interleave(samples_flt, s->samples_chanptr, 256,

  1892.                                          channels);

  1893.             samples_flt += 256 * channels;

  1894.         } else {

  1895.             s->fmt_conv.float_to_int16_interleave(samples_s16,

  1896.                                                   s->samples_chanptr, 256,

  1897.                                                   channels);

  1898.             samples_s16 += 256 * channels;

  1899.         }

  1900.     }

  1901. 

  1902.     /* update lfe history */

  1903.     lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);

  1904.     for (i = 0; i < 2 * s->lfe * 4; i++)

  1905.         s->lfe_data[i] = s->lfe_data[i + lfe_samples];

  1906. 

  1907.     *got_frame_ptr    = 1;

  1908.     *(AVFrame *) data = s->frame;

  1909. 

  1910.     return buf_size;

  1911. }

  1912. 

  1913. 

  1914. 

  1915. /**

  1916.  * DCA initialization

  1917.  *

  1918.  * @param avctx     pointer to the AVCodecContext

  1919.  */

  1920. 

  1921. static av_cold int dca_decode_init(AVCodecContext *avctx)

  1922. {

  1923.     DCAContext *s = avctx->priv_data;

  1924.     int i;

  1925. 

  1926.     s->avctx = avctx;

  1927.     dca_init_vlcs();

  1928. 

  1929.     dsputil_init(&s->dsp, avctx);

  1930.     ff_mdct_init(&s->imdct, 6, 1, 1.0);

  1931.     ff_synth_filter_init(&s->synth);

  1932.     ff_dcadsp_init(&s->dcadsp);

  1933.     ff_fmt_convert_init(&s->fmt_conv, avctx);

  1934. 

  1935.     for (i = 0; i < DCA_PRIM_CHANNELS_MAX + 1; i++)

  1936.         s->samples_chanptr[i] = s->samples + i * 256;

  1937. 

  1938.     if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {

  1939.         avctx->sample_fmt = AV_SAMPLE_FMT_FLT;

  1940.         s->scale_bias     = 1.0 / 32768.0;

  1941.     } else {

  1942.         avctx->sample_fmt = AV_SAMPLE_FMT_S16;

  1943.         s->scale_bias     = 1.0;

  1944.     }

  1945. 

  1946.     /* allow downmixing to stereo */

  1947.     if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&

  1948.         avctx->request_channels == 2) {

  1949.         avctx->channels = avctx->request_channels;

  1950.     }

  1951. 

  1952.     avcodec_get_frame_defaults(&s->frame);

  1953.     avctx->coded_frame = &s->frame;

  1954. 

  1955.     return 0;

  1956. }

  1957. 

  1958. static av_cold int dca_decode_end(AVCodecContext *avctx)

  1959. {

  1960.     DCAContext *s = avctx->priv_data;

  1961.     ff_mdct_end(&s->imdct);

  1962.     return 0;

  1963. }

  1964. 

  1965. static const AVProfile profiles[] = {

  1966.     { FF_PROFILE_DTS,        "DTS"        },

  1967.     { FF_PROFILE_DTS_ES,     "DTS-ES"     },

  1968.     { FF_PROFILE_DTS_96_24,  "DTS 96/24"  },

  1969.     { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },

  1970.     { FF_PROFILE_DTS_HD_MA,  "DTS-HD MA"  },

  1971.     { FF_PROFILE_UNKNOWN },

  1972. };

  1973. 

  1974. AVCodec ff_dca_decoder = {

  1975.     .name            = "dca",

  1976.     .type            = AVMEDIA_TYPE_AUDIO,

  1977.     .id              = CODEC_ID_DTS,

  1978.     .priv_data_size  = sizeof(DCAContext),

  1979.     .init            = dca_decode_init,

  1980.     .decode          = dca_decode_frame,

  1981.     .close           = dca_decode_end,

  1982.     .long_name       = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),

  1983.     .capabilities    = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,

  1984.     .sample_fmts     = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,

  1985.                                                        AV_SAMPLE_FMT_S16,

  1986.                                                        AV_SAMPLE_FMT_NONE },

  1987.     .profiles        = NULL_IF_CONFIG_SMALL(profiles),

  1988. };