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FFmpeg/libavcodec/dca.c

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  1. /*

  2.  * DCA compatible decoder

  3.  * Copyright (C) 2004 Gildas Bazin

  4.  * Copyright (C) 2004 Benjamin Zores

  5.  * Copyright (C) 2006 Benjamin Larsson

  6.  * Copyright (C) 2007 Konstantin Shishkov

  7.  *

  8.  * This file is part of FFmpeg.

  9.  *

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

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

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

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

  14.  *

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

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

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

  18.  * Lesser General Public License for more details.

  19.  *

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

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

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

  23.  */

  24. 

  25. #include <math.h>

  26. #include <stddef.h>

  27. #include <stdio.h>

  28. 

  29. #include "libavutil/common.h"

  30. #include "libavutil/intmath.h"

  31. #include "libavutil/intreadwrite.h"

  32. #include "libavcore/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. //#define TRACE

  46. 

  47. #define DCA_PRIM_CHANNELS_MAX (7)

  48. #define DCA_SUBBANDS (32)

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

  50. #define DCA_SUBSUBFRAMES_MAX (4)

  51. #define DCA_SUBFRAMES_MAX (16)

  52. #define DCA_BLOCKS_MAX (16)

  53. #define DCA_LFE_MAX (3)

  54. 

  55. enum DCAMode {

  56.     DCA_MONO = 0,

  57.     DCA_CHANNEL,

  58.     DCA_STEREO,

  59.     DCA_STEREO_SUMDIFF,

  60.     DCA_STEREO_TOTAL,

  61.     DCA_3F,

  62.     DCA_2F1R,

  63.     DCA_3F1R,

  64.     DCA_2F2R,

  65.     DCA_3F2R,

  66.     DCA_4F2R

  67. };

  68. 

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

  70. enum DCAExSSSpeakerMask {

  71.     DCA_EXSS_FRONT_CENTER          = 0x0001,

  72.     DCA_EXSS_FRONT_LEFT_RIGHT      = 0x0002,

  73.     DCA_EXSS_SIDE_REAR_LEFT_RIGHT  = 0x0004,

  74.     DCA_EXSS_LFE                   = 0x0008,

  75.     DCA_EXSS_REAR_CENTER           = 0x0010,

  76.     DCA_EXSS_FRONT_HIGH_LEFT_RIGHT = 0x0020,

  77.     DCA_EXSS_REAR_LEFT_RIGHT       = 0x0040,

  78.     DCA_EXSS_FRONT_HIGH_CENTER     = 0x0080,

  79.     DCA_EXSS_OVERHEAD              = 0x0100,

  80.     DCA_EXSS_CENTER_LEFT_RIGHT     = 0x0200,

  81.     DCA_EXSS_WIDE_LEFT_RIGHT       = 0x0400,

  82.     DCA_EXSS_SIDE_LEFT_RIGHT       = 0x0800,

  83.     DCA_EXSS_LFE2                  = 0x1000,

  84.     DCA_EXSS_SIDE_HIGH_LEFT_RIGHT  = 0x2000,

  85.     DCA_EXSS_REAR_HIGH_CENTER      = 0x4000,

  86.     DCA_EXSS_REAR_HIGH_LEFT_RIGHT  = 0x8000,

  87. };

  88. 

  89. enum DCAExtensionMask {

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

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

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

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

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

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

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

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

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

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

  100. };

  101. 

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

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

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

  105.  *

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

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

  108.  * OV -> center back

  109.  * All 2 channel configurations -> CH_LAYOUT_STEREO

  110.  */

  111. 

  112. static const int64_t dca_core_channel_layout[] = {

  113.     AV_CH_FRONT_CENTER,                                                      ///< 1, A

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

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

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

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

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

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

  120.     AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER|CH_BACK_CENTER,                   ///< 4, C + L + R+ S

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

  122.     AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT, ///< 5, C + L + R+ SL+SR

  123.     AV_CH_LAYOUT_STEREO|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT|AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER,                    ///< 6, CL + CR + L + R + SL + SR

  124.     AV_CH_LAYOUT_STEREO|AV_CH_BACK_LEFT|AV_CH_BACK_RIGHT|AV_CH_FRONT_CENTER|AV_CH_BACK_CENTER,                                      ///< 6, C + L + R+ LR + RR + OV

  125.     AV_CH_FRONT_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER|AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_BACK_CENTER|AV_CH_BACK_LEFT|AV_CH_BACK_RIGHT,   ///< 6, CF+ CR+LF+ RF+LR + RR

  126.     AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER|AV_CH_LAYOUT_STEREO|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT, ///< 7, CL + C + CR + L + R + SL + SR

  127.     AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER|AV_CH_LAYOUT_STEREO|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT|AV_CH_BACK_LEFT|AV_CH_BACK_RIGHT, ///< 8, CL + CR + L + R + SL1 + SL2+ SR1 + SR2

  128.     AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER|AV_CH_LAYOUT_STEREO|AV_CH_SIDE_LEFT|AV_CH_BACK_CENTER|AV_CH_SIDE_RIGHT, ///< 8, CL + C+ CR + L + R + SL + S+ SR

  129. };

  130. 

  131. static const int8_t dca_lfe_index[] = {

  132.     1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3

  133. };

  134. 

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

  136.     { 0, -1, -1, -1, -1, -1, -1, -1, -1},

  137.     { 0,  1, -1, -1, -1, -1, -1, -1, -1},

  138.     { 0,  1, -1, -1, -1, -1, -1, -1, -1},

  139.     { 0,  1, -1, -1, -1, -1, -1, -1, -1},

  140.     { 0,  1, -1, -1, -1, -1, -1, -1, -1},

  141.     { 2,  0,  1, -1, -1, -1, -1, -1, -1},

  142.     { 0,  1,  3, -1, -1, -1, -1, -1, -1},

  143.     { 2,  0,  1,  4, -1, -1, -1, -1, -1},

  144.     { 0,  1,  3,  4, -1, -1, -1, -1, -1},

  145.     { 2,  0,  1,  4,  5, -1, -1, -1, -1},

  146.     { 3,  4,  0,  1,  5,  6, -1, -1, -1},

  147.     { 2,  0,  1,  4,  5,  6, -1, -1, -1},

  148.     { 0,  6,  4,  5,  2,  3, -1, -1, -1},

  149.     { 4,  2,  5,  0,  1,  6,  7, -1, -1},

  150.     { 5,  6,  0,  1,  7,  3,  8,  4, -1},

  151.     { 4,  2,  5,  0,  1,  6,  8,  7, -1},

  152. };

  153. 

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

  155.     { 0,  2, -1, -1, -1, -1, -1, -1, -1},

  156.     { 0,  1,  3, -1, -1, -1, -1, -1, -1},

  157.     { 0,  1,  3, -1, -1, -1, -1, -1, -1},

  158.     { 0,  1,  3, -1, -1, -1, -1, -1, -1},

  159.     { 0,  1,  3, -1, -1, -1, -1, -1, -1},

  160.     { 2,  0,  1,  4, -1, -1, -1, -1, -1},

  161.     { 0,  1,  3,  4, -1, -1, -1, -1, -1},

  162.     { 2,  0,  1,  4,  5, -1, -1, -1, -1},

  163.     { 0,  1,  4,  5,  3, -1, -1, -1, -1},

  164.     { 2,  0,  1,  5,  6,  4, -1, -1, -1},

  165.     { 3,  4,  0,  1,  6,  7,  5, -1, -1},

  166.     { 2,  0,  1,  4,  5,  6,  7, -1, -1},

  167.     { 0,  6,  4,  5,  2,  3,  7, -1, -1},

  168.     { 4,  2,  5,  0,  1,  7,  8,  6, -1},

  169.     { 5,  6,  0,  1,  8,  3,  9,  4,  7},

  170.     { 4,  2,  5,  0,  1,  6,  9,  8,  7},

  171. };

  172. 

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

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

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

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

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

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

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

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

  181.     { 2,  0,  1,  3, -1, -1, -1, -1, -1},

  182.     { 0,  1,  2,  3, -1, -1, -1, -1, -1},

  183.     { 2,  0,  1,  3,  4, -1, -1, -1, -1},

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

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

  186.     { 0,  5,  3,  4,  1,  2, -1, -1, -1},

  187.     { 3,  2,  4,  0,  1,  5,  6, -1, -1},

  188.     { 4,  5,  0,  1,  6,  2,  7,  3, -1},

  189.     { 3,  2,  4,  0,  1,  5,  7,  6, -1},

  190. };

  191. 

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

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

  194.     { 0,  1,  2, -1, -1, -1, -1, -1, -1},

  195.     { 0,  1,  2, -1, -1, -1, -1, -1, -1},

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

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

  198.     { 2,  0,  1,  3, -1, -1, -1, -1, -1},

  199.     { 0,  1,  2,  3, -1, -1, -1, -1, -1},

  200.     { 2,  0,  1,  3,  4, -1, -1, -1, -1},

  201.     { 0,  1,  3,  4,  2, -1, -1, -1, -1},

  202.     { 2,  0,  1,  4,  5,  3, -1, -1, -1},

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

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

  205.     { 0,  5,  3,  4,  1,  2,  6, -1, -1},

  206.     { 3,  2,  4,  0,  1,  6,  7,  5, -1},

  207.     { 4,  5,  0,  1,  7,  2,  8,  3,  6},

  208.     { 3,  2,  4,  0,  1,  5,  8,  7,  6},

  209. };

  210. 

  211. #define DCA_DOLBY 101           /* FIXME */

  212. 

  213. #define DCA_CHANNEL_BITS 6

  214. #define DCA_CHANNEL_MASK 0x3F

  215. 

  216. #define DCA_LFE 0x80

  217. 

  218. #define HEADER_SIZE 14

  219. 

  220. #define DCA_MAX_FRAME_SIZE 16384

  221. #define DCA_MAX_EXSS_HEADER_SIZE 4096

  222. 

  223. #define DCA_BUFFER_PADDING_SIZE 1024

  224. 

  225. /** Bit allocation */

  226. typedef struct {

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

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

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

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

  231. } BitAlloc;

  232. 

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

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

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

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

  237. 

  238. static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba, int idx)

  239. {

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

  241. }

  242. 

  243. typedef struct {

  244.     AVCodecContext *avctx;

  245.     /* Frame header */

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

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

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

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

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

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

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

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

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

  255. 

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

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

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

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

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

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

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

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

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

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

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

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

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

  269.     int copy_history;           ///< copy history

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

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

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

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

  274. 

  275.     /* Primary audio coding header */

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

  277.     int is_channels_set;        ///< check for if the channel number is already set

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

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

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

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

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

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

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

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

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

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

  288. 

  289.     /* Primary audio coding side information */

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

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

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

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

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

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

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

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

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

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

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

  301. 

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

  303. 

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

  305.     int lfe_scale_factor;

  306. 

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

  308.     float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];

  309.     DECLARE_ALIGNED(16, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512];

  310.     DECLARE_ALIGNED(16, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32];

  311.     int hist_index[DCA_PRIM_CHANNELS_MAX];

  312.     DECLARE_ALIGNED(16, float, raXin)[32];

  313. 

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

  315.     float scale_bias;           ///< output scale

  316. 

  317.     DECLARE_ALIGNED(16, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];

  318.     DECLARE_ALIGNED(16, float, samples)[(DCA_PRIM_CHANNELS_MAX+1)*256];

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

  320. 

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

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

  323. 

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

  325.     GetBitContext gb;

  326.     /* Current position in DCA frame */

  327.     int current_subframe;

  328.     int current_subsubframe;

  329. 

  330.     /* XCh extension information */

  331.     int xch_present;

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

  333. 

  334.     /* Other detected extensions in the core substream */

  335.     int xxch_present;

  336.     int x96_present;

  337. 

  338.     /* ExSS header parser */

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

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

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

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

  343. 

  344.     int profile;

  345. 

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

  347.     DSPContext dsp;

  348.     FFTContext imdct;

  349.     SynthFilterContext synth;

  350.     DCADSPContext dcadsp;

  351.     FmtConvertContext fmt_conv;

  352. } DCAContext;

  353. 

  354. static const uint16_t dca_vlc_offs[] = {

  355.         0,   512,   640,   768,  1282,  1794,  2436,  3080,  3770,  4454,  5364,

  356.      5372,  5380,  5388,  5392,  5396,  5412,  5420,  5428,  5460,  5492,  5508,

  357.      5572,  5604,  5668,  5796,  5860,  5892,  6412,  6668,  6796,  7308,  7564,

  358.      7820,  8076,  8620,  9132,  9388,  9910, 10166, 10680, 11196, 11726, 12240,

  359.     12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264,

  360.     18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622,

  361. };

  362. 

  363. static av_cold void dca_init_vlcs(void)

  364. {

  365.     static int vlcs_initialized = 0;

  366.     int i, j, c = 14;

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

  368. 

  369.     if (vlcs_initialized)

  370.         return;

  371. 

  372.     dca_bitalloc_index.offset = 1;

  373.     dca_bitalloc_index.wrap = 2;

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

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

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

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

  378.                  bitalloc_12_bits[i], 1, 1,

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

  380.     }

  381.     dca_scalefactor.offset = -64;

  382.     dca_scalefactor.wrap = 2;

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

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

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

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

  387.                  scales_bits[i], 1, 1,

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

  389.     }

  390.     dca_tmode.offset = 0;

  391.     dca_tmode.wrap = 1;

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

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

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

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

  396.                  tmode_bits[i], 1, 1,

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

  398.     }

  399. 

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

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

  402.             if (!bitalloc_codes[i][j]) break;

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

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

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

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

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

  408.                      bitalloc_sizes[i],

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

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

  411.             c++;

  412.         }

  413.     vlcs_initialized = 1;

  414. }

  415. 

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

  417. {

  418.     while(len--)

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

  420. }

  421. 

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

  423. {

  424.     int i, j;

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

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

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

  428. 

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

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

  431. 

  432.     if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)

  433.         s->prim_channels = DCA_PRIM_CHANNELS_MAX;

  434. 

  435. 

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

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

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

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

  440.     }

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

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

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

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

  445.     }

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

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

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

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

  450. 

  451.     /* Get codebooks quantization indexes */

  452.     if (!base_channel)

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

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

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

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

  457. 

  458.     /* Get scale factor adjustment */

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

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

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

  462. 

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

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

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

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

  467. 

  468.     if (s->crc_present) {

  469.         /* Audio header CRC check */

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

  471.     }

  472. 

  473.     s->current_subframe = 0;

  474.     s->current_subsubframe = 0;

  475. 

  476. #ifdef TRACE

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

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

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

  480.         av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", s->subband_activity[i]);

  481.         av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", s->vq_start_subband[i]);

  482.         av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", s->joint_intensity[i]);

  483.         av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", s->transient_huffman[i]);

  484.         av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", s->scalefactor_huffman[i]);

  485.         av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", s->bitalloc_huffman[i]);

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

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

  488.             av_log(s->avctx, AV_LOG_DEBUG, " %i",

  489.                    s->quant_index_huffman[i][j]);

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

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

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

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

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

  495.     }

  496. #endif

  497. 

  498.   return 0;

  499. }

  500. 

  501. static int dca_parse_frame_header(DCAContext * s)

  502. {

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

  504. 

  505.     /* Sync code */

  506.     get_bits(&s->gb, 32);

  507. 

  508.     /* Frame header */

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

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

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

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

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

  514.     if (s->frame_size < 95)

  515.         return -1;

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

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

  518.     if (!s->sample_rate)

  519.         return -1;

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

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

  522.     if (!s->bit_rate)

  523.         return -1;

  524. 

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

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

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

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

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

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

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

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

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

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

  535. 

  536.     /* TODO: check CRC */

  537.     if (s->crc_present)

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

  539. 

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

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

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

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

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

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

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

  547. 

  548.     /* FIXME: channels mixing levels */

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

  550.     if (s->lfe) s->output |= DCA_LFE;

  551. 

  552. #ifdef TRACE

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

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

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

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

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

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

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

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

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

  562.            s->sample_rate);

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

  564.            s->bit_rate);

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

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

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

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

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

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

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

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

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

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

  575.            s->predictor_history);

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

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

  578.            s->multirate_inter);

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

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

  581.     av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

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

  588. #endif

  589. 

  590.     /* Primary audio coding header */

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

  592. 

  593.     return dca_parse_audio_coding_header(s, 0);

  594. }

  595. 

  596. 

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

  598. {

  599.    if (level < 5) {

  600.        /* huffman encoded */

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

  602.    } else if (level < 8)

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

  604.    return value;

  605. }

  606. 

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

  608. {

  609.     /* Primary audio coding side information */

  610.     int j, k;

  611. 

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

  613.         return -1;

  614. 

  615.     if (!base_channel) {

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

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

  618.     }

  619. 

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

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

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

  623.     }

  624. 

  625.     /* Get prediction codebook */

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

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

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

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

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

  631.             }

  632.         }

  633.     }

  634. 

  635.     /* Bit allocation index */

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

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

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

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

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

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

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

  643.                 av_log(s->avctx, AV_LOG_ERROR,

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

  645.                 return -1;

  646.             } else {

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

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

  649.             }

  650. 

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

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

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

  654.                 return -1;

  655.             }

  656.         }

  657.     }

  658. 

  659.     /* Transition mode */

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

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

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

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

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

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

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

  667.             }

  668.         }

  669.     }

  670. 

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

  672.         return -1;

  673. 

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

  675.         const uint32_t *scale_table;

  676.         int scale_sum;

  677. 

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

  679. 

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

  681.             scale_table = scale_factor_quant7;

  682.         else

  683.             scale_table = scale_factor_quant6;

  684. 

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

  686.         scale_sum = 0;

  687. 

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

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

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

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

  692.             }

  693. 

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

  695.                 /* Get second scale factor */

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

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

  698.             }

  699.         }

  700.     }

  701. 

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

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

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

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

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

  707.     }

  708. 

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

  710.         return -1;

  711. 

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

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

  714.         int source_channel;

  715. 

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

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

  718.             int scale = 0;

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

  720. 

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

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

  723. 

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

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

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

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

  728.             }

  729. 

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

  731.                 av_log(s->avctx, AV_LOG_DEBUG,

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

  733.                 s->debug_flag |= 0x02;

  734.             }

  735.         }

  736.     }

  737. 

  738.     /* Stereo downmix coefficients */

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

  740.         if (s->downmix) {

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

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

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

  744.             }

  745.         } else {

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

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

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

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

  750.             }

  751.         }

  752.     }

  753. 

  754.     /* Dynamic range coefficient */

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

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

  757. 

  758.     /* Side information CRC check word */

  759.     if (s->crc_present) {

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

  761.     }

  762. 

  763.     /*

  764.      * Primary audio data arrays

  765.      */

  766. 

  767.     /* VQ encoded high frequency subbands */

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

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

  770.             /* 1 vector -> 32 samples */

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

  772. 

  773.     /* Low frequency effect data */

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

  775.         /* LFE samples */

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

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

  778.         float lfe_scale;

  779. 

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

  781.             /* Signed 8 bits int */

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

  783.         }

  784. 

  785.         /* Scale factor index */

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

  787. 

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

  789.         lfe_scale = 0.035 * s->lfe_scale_factor;

  790. 

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

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

  793.     }

  794. 

  795. #ifdef TRACE

  796.     av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n", s->subsubframes[s->current_subframe]);

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

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

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

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

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

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

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

  804.     }

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

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

  807.                 av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

  813.     }

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

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

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

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

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

  819.     }

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

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

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

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

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

  825.     }

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

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

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

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

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

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

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

  833.         }

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

  835.     }

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

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

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

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

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

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

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

  843.         }

  844.     }

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

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

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

  848.             av_log(s->avctx, AV_LOG_DEBUG, "Channel 0,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][0]]);

  849.             av_log(s->avctx, AV_LOG_DEBUG, "Channel 1,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][1]]);

  850.         }

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

  852.     }

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

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

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

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

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

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

  859. 

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

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

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

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

  864.     }

  865. #endif

  866. 

  867.     return 0;

  868. }

  869. 

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

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

  872.                             float scale)

  873. {

  874.     const float *prCoeff;

  875.     int i;

  876. 

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

  878.     int subindex;

  879. 

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

  881. 

  882.     /* Select filter */

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

  884.         prCoeff = fir_32bands_nonperfect;

  885.     else                        /* Perfect reconstruction */

  886.         prCoeff = fir_32bands_perfect;

  887. 

  888.     /* Reconstructed channel sample index */

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

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

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

  892.             uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ ((i-1)&2)<<30;

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

  894.         }

  895.         for (; i < 32; i++)

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

  897. 

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

  899.                               s->subband_fir_hist[chans], &s->hist_index[chans],

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

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

  902.         samples_out+= 32;

  903. 

  904.     }

  905. }

  906. 

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

  908.                                   int num_deci_sample, float *samples_in,

  909.                                   float *samples_out, float scale)

  910. {

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

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

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

  914.      *   from last subframe as history.

  915.      *

  916.      * samples_out: An array holding interpolated samples

  917.      */

  918. 

  919.     int decifactor;

  920.     const float *prCoeff;

  921.     int deciindex;

  922. 

  923.     /* Select decimation filter */

  924.     if (decimation_select == 1) {

  925.         decifactor = 64;

  926.         prCoeff = lfe_fir_128;

  927.     } else {

  928.         decifactor = 32;

  929.         prCoeff = lfe_fir_64;

  930.     }

  931.     /* Interpolation */

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

  933.         s->dcadsp.lfe_fir(samples_out, samples_in, prCoeff, decifactor,

  934.                           scale);

  935.         samples_in++;

  936.         samples_out += 2 * decifactor;

  937.     }

  938. }

  939. 

  940. /* downmixing routines */

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

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

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

  944. 

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

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

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

  948. 

  949. #define MIX_FRONT3(samples, coef) \

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

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

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

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

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

  955. 

  956. #define DOWNMIX_TO_STEREO(op1, op2) \

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

  958.         op1 \

  959.         op2 \

  960.     }

  961. 

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

  963.                         int downmix_coef[DCA_PRIM_CHANNELS_MAX][2],

  964.                         const int8_t *channel_mapping)

  965. {

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

  967.     int i;

  968.     float t, u, v;

  969.     float coef[DCA_PRIM_CHANNELS_MAX][2];

  970. 

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

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

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

  974.     }

  975. 

  976.     switch (srcfmt) {

  977.     case DCA_MONO:

  978.     case DCA_CHANNEL:

  979.     case DCA_STEREO_TOTAL:

  980.     case DCA_STEREO_SUMDIFF:

  981.     case DCA_4F2R:

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

  983.         break;

  984.     case DCA_STEREO:

  985.         break;

  986.     case DCA_3F:

  987.         c = channel_mapping[0] * 256;

  988.         l = channel_mapping[1] * 256;

  989.         r = channel_mapping[2] * 256;

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

  991.         break;

  992.     case DCA_2F1R:

  993.         s = channel_mapping[2] * 256;

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

  995.         break;

  996.     case DCA_3F1R:

  997.         c = channel_mapping[0] * 256;

  998.         l = channel_mapping[1] * 256;

  999.         r = channel_mapping[2] * 256;

  1000.         s = channel_mapping[3] * 256;

  1001.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1002.                           MIX_REAR1(samples, i + s, 3, coef));

  1003.         break;

  1004.     case DCA_2F2R:

  1005.         sl = channel_mapping[2] * 256;

  1006.         sr = channel_mapping[3] * 256;

  1007.         DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + sl, i + sr, 2, coef),);

  1008.         break;

  1009.     case DCA_3F2R:

  1010.         c =  channel_mapping[0] * 256;

  1011.         l =  channel_mapping[1] * 256;

  1012.         r =  channel_mapping[2] * 256;

  1013.         sl = channel_mapping[3] * 256;

  1014.         sr = channel_mapping[4] * 256;

  1015.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1016.                           MIX_REAR2(samples, i + sl, i + sr, 3, coef));

  1017.         break;

  1018.     }

  1019. }

  1020. 

  1021. 

  1022. /* Very compact version of the block code decoder that does not use table

  1023.  * look-up but is slightly slower */

  1024. static int decode_blockcode(int code, int levels, int *values)

  1025. {

  1026.     int i;

  1027.     int offset = (levels - 1) >> 1;

  1028. 

  1029.     for (i = 0; i < 4; i++) {

  1030.         int div = FASTDIV(code, levels);

  1031.         values[i] = code - offset - div*levels;

  1032.         code = div;

  1033.     }

  1034. 

  1035.     if (code == 0)

  1036.         return 0;

  1037.     else {

  1038.         av_log(NULL, AV_LOG_ERROR, "ERROR: block code look-up failed\n");

  1039.         return -1;

  1040.     }

  1041. }

  1042. 

  1043. static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };

  1044. static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };

  1045. 

  1046. static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)

  1047. {

  1048.     int k, l;

  1049.     int subsubframe = s->current_subsubframe;

  1050. 

  1051.     const float *quant_step_table;

  1052. 

  1053.     /* FIXME */

  1054.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1055.     LOCAL_ALIGNED_16(int, block, [8]);

  1056. 

  1057.     /*

  1058.      * Audio data

  1059.      */

  1060. 

  1061.     /* Select quantization step size table */

  1062.     if (s->bit_rate_index == 0x1f)

  1063.         quant_step_table = lossless_quant_d;

  1064.     else

  1065.         quant_step_table = lossy_quant_d;

  1066. 

  1067.     for (k = base_channel; k < s->prim_channels; k++) {

  1068.         if (get_bits_left(&s->gb) < 0)

  1069.             return -1;

  1070. 

  1071.         for (l = 0; l < s->vq_start_subband[k]; l++) {

  1072.             int m;

  1073. 

  1074.             /* Select the mid-tread linear quantizer */

  1075.             int abits = s->bitalloc[k][l];

  1076. 

  1077.             float quant_step_size = quant_step_table[abits];

  1078. 

  1079.             /*

  1080.              * Determine quantization index code book and its type

  1081.              */

  1082. 

  1083.             /* Select quantization index code book */

  1084.             int sel = s->quant_index_huffman[k][abits];

  1085. 

  1086.             /*

  1087.              * Extract bits from the bit stream

  1088.              */

  1089.             if (!abits){

  1090.                 memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));

  1091.             } else {

  1092.                 /* Deal with transients */

  1093.                 int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];

  1094.                 float rscale = quant_step_size * s->scale_factor[k][l][sfi] * s->scalefactor_adj[k][sel];

  1095. 

  1096.                 if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){

  1097.                     if (abits <= 7){

  1098.                         /* Block code */

  1099.                         int block_code1, block_code2, size, levels;

  1100. 

  1101.                         size = abits_sizes[abits-1];

  1102.                         levels = abits_levels[abits-1];

  1103. 

  1104.                         block_code1 = get_bits(&s->gb, size);

  1105.                         /* FIXME Should test return value */

  1106.                         decode_blockcode(block_code1, levels, block);

  1107.                         block_code2 = get_bits(&s->gb, size);

  1108.                         decode_blockcode(block_code2, levels, &block[4]);

  1109.                     }else{

  1110.                         /* no coding */

  1111.                         for (m = 0; m < 8; m++)

  1112.                             block[m] = get_sbits(&s->gb, abits - 3);

  1113.                     }

  1114.                 }else{

  1115.                     /* Huffman coded */

  1116.                     for (m = 0; m < 8; m++)

  1117.                         block[m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);

  1118.                 }

  1119. 

  1120.                 s->fmt_conv.int32_to_float_fmul_scalar(subband_samples[k][l],

  1121.                                                   block, rscale, 8);

  1122.             }

  1123. 

  1124.             /*

  1125.              * Inverse ADPCM if in prediction mode

  1126.              */

  1127.             if (s->prediction_mode[k][l]) {

  1128.                 int n;

  1129.                 for (m = 0; m < 8; m++) {

  1130.                     for (n = 1; n <= 4; n++)

  1131.                         if (m >= n)

  1132.                             subband_samples[k][l][m] +=

  1133.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1134.                                  subband_samples[k][l][m - n] / 8192);

  1135.                         else if (s->predictor_history)

  1136.                             subband_samples[k][l][m] +=

  1137.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1138.                                  s->subband_samples_hist[k][l][m - n +

  1139.                                                                4] / 8192);

  1140.                 }

  1141.             }

  1142.         }

  1143. 

  1144.         /*

  1145.          * Decode VQ encoded high frequencies

  1146.          */

  1147.         for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {

  1148.             /* 1 vector -> 32 samples but we only need the 8 samples

  1149.              * for this subsubframe. */

  1150.             int m;

  1151. 

  1152.             if (!s->debug_flag & 0x01) {

  1153.                 av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n");

  1154.                 s->debug_flag |= 0x01;

  1155.             }

  1156. 

  1157.             for (m = 0; m < 8; m++) {

  1158.                 subband_samples[k][l][m] =

  1159.                     high_freq_vq[s->high_freq_vq[k][l]][subsubframe * 8 +

  1160.                                                         m]

  1161.                     * (float) s->scale_factor[k][l][0] / 16.0;

  1162.             }

  1163.         }

  1164.     }

  1165. 

  1166.     /* Check for DSYNC after subsubframe */

  1167.     if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {

  1168.         if (0xFFFF == get_bits(&s->gb, 16)) {   /* 0xFFFF */

  1169. #ifdef TRACE

  1170.             av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");

  1171. #endif

  1172.         } else {

  1173.             av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");

  1174.         }

  1175.     }

  1176. 

  1177.     /* Backup predictor history for adpcm */

  1178.     for (k = base_channel; k < s->prim_channels; k++)

  1179.         for (l = 0; l < s->vq_start_subband[k]; l++)

  1180.             memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4],

  1181.                         4 * sizeof(subband_samples[0][0][0]));

  1182. 

  1183.     return 0;

  1184. }

  1185. 

  1186. static int dca_filter_channels(DCAContext * s, int block_index)

  1187. {

  1188.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1189.     int k;

  1190. 

  1191.     /* 32 subbands QMF */

  1192.     for (k = 0; k < s->prim_channels; k++) {

  1193. /*        static float pcm_to_double[8] =

  1194.             {32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/

  1195.          qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * s->channel_order_tab[k]],

  1196.                          M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ );

  1197.     }

  1198. 

  1199.     /* Down mixing */

  1200.     if (s->avctx->request_channels == 2 && s->prim_channels > 2) {

  1201.         dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab);

  1202.     }

  1203. 

  1204.     /* Generate LFE samples for this subsubframe FIXME!!! */

  1205.     if (s->output & DCA_LFE) {

  1206.         lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,

  1207.                               s->lfe_data + 2 * s->lfe * (block_index + 4),

  1208.                               &s->samples[256 * dca_lfe_index[s->amode]],

  1209.                               (1.0/256.0)*s->scale_bias);

  1210.         /* Outputs 20bits pcm samples */

  1211.     }

  1212. 

  1213.     return 0;

  1214. }

  1215. 

  1216. 

  1217. static int dca_subframe_footer(DCAContext * s, int base_channel)

  1218. {

  1219.     int aux_data_count = 0, i;

  1220. 

  1221.     /*

  1222.      * Unpack optional information

  1223.      */

  1224. 

  1225.     /* presumably optional information only appears in the core? */

  1226.     if (!base_channel) {

  1227.         if (s->timestamp)

  1228.             get_bits(&s->gb, 32);

  1229. 

  1230.         if (s->aux_data)

  1231.             aux_data_count = get_bits(&s->gb, 6);

  1232. 

  1233.         for (i = 0; i < aux_data_count; i++)

  1234.             get_bits(&s->gb, 8);

  1235. 

  1236.         if (s->crc_present && (s->downmix || s->dynrange))

  1237.             get_bits(&s->gb, 16);

  1238.     }

  1239. 

  1240.     return 0;

  1241. }

  1242. 

  1243. /**

  1244.  * Decode a dca frame block

  1245.  *

  1246.  * @param s     pointer to the DCAContext

  1247.  */

  1248. 

  1249. static int dca_decode_block(DCAContext * s, int base_channel, int block_index)

  1250. {

  1251. 

  1252.     /* Sanity check */

  1253.     if (s->current_subframe >= s->subframes) {

  1254.         av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",

  1255.                s->current_subframe, s->subframes);

  1256.         return -1;

  1257.     }

  1258. 

  1259.     if (!s->current_subsubframe) {

  1260. #ifdef TRACE

  1261.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");

  1262. #endif

  1263.         /* Read subframe header */

  1264.         if (dca_subframe_header(s, base_channel, block_index))

  1265.             return -1;

  1266.     }

  1267. 

  1268.     /* Read subsubframe */

  1269. #ifdef TRACE

  1270.     av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");

  1271. #endif

  1272.     if (dca_subsubframe(s, base_channel, block_index))

  1273.         return -1;

  1274. 

  1275.     /* Update state */

  1276.     s->current_subsubframe++;

  1277.     if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {

  1278.         s->current_subsubframe = 0;

  1279.         s->current_subframe++;

  1280.     }

  1281.     if (s->current_subframe >= s->subframes) {

  1282. #ifdef TRACE

  1283.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");

  1284. #endif

  1285.         /* Read subframe footer */

  1286.         if (dca_subframe_footer(s, base_channel))

  1287.             return -1;

  1288.     }

  1289. 

  1290.     return 0;

  1291. }

  1292. 

  1293. /**

  1294.  * Convert bitstream to one representation based on sync marker

  1295.  */

  1296. static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * dst,

  1297.                           int max_size)

  1298. {

  1299.     uint32_t mrk;

  1300.     int i, tmp;

  1301.     const uint16_t *ssrc = (const uint16_t *) src;

  1302.     uint16_t *sdst = (uint16_t *) dst;

  1303.     PutBitContext pb;

  1304. 

  1305.     if ((unsigned)src_size > (unsigned)max_size) {

  1306. //        av_log(NULL, AV_LOG_ERROR, "Input frame size larger then DCA_MAX_FRAME_SIZE!\n");

  1307. //        return -1;

  1308.         src_size = max_size;

  1309.     }

  1310. 

  1311.     mrk = AV_RB32(src);

  1312.     switch (mrk) {

  1313.     case DCA_MARKER_RAW_BE:

  1314.         memcpy(dst, src, src_size);

  1315.         return src_size;

  1316.     case DCA_MARKER_RAW_LE:

  1317.         for (i = 0; i < (src_size + 1) >> 1; i++)

  1318.             *sdst++ = av_bswap16(*ssrc++);

  1319.         return src_size;

  1320.     case DCA_MARKER_14B_BE:

  1321.     case DCA_MARKER_14B_LE:

  1322.         init_put_bits(&pb, dst, max_size);

  1323.         for (i = 0; i < (src_size + 1) >> 1; i++, src += 2) {

  1324.             tmp = ((mrk == DCA_MARKER_14B_BE) ? AV_RB16(src) : AV_RL16(src)) & 0x3FFF;

  1325.             put_bits(&pb, 14, tmp);

  1326.         }

  1327.         flush_put_bits(&pb);

  1328.         return (put_bits_count(&pb) + 7) >> 3;

  1329.     default:

  1330.         return -1;

  1331.     }

  1332. }

  1333. 

  1334. /**

  1335.  * Return the number of channels in an ExSS speaker mask (HD)

  1336.  */

  1337. static int dca_exss_mask2count(int mask)

  1338. {

  1339.     /* count bits that mean speaker pairs twice */

  1340.     return av_popcount(mask)

  1341.         + av_popcount(mask & (

  1342.             DCA_EXSS_CENTER_LEFT_RIGHT

  1343.           | DCA_EXSS_FRONT_LEFT_RIGHT

  1344.           | DCA_EXSS_FRONT_HIGH_LEFT_RIGHT

  1345.           | DCA_EXSS_WIDE_LEFT_RIGHT

  1346.           | DCA_EXSS_SIDE_LEFT_RIGHT

  1347.           | DCA_EXSS_SIDE_HIGH_LEFT_RIGHT

  1348.           | DCA_EXSS_SIDE_REAR_LEFT_RIGHT

  1349.           | DCA_EXSS_REAR_LEFT_RIGHT

  1350.           | DCA_EXSS_REAR_HIGH_LEFT_RIGHT

  1351.           ));

  1352. }

  1353. 

  1354. /**

  1355.  * Skip mixing coefficients of a single mix out configuration (HD)

  1356.  */

  1357. static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch)

  1358. {

  1359.     int i;

  1360. 

  1361.     for (i = 0; i < channels; i++) {

  1362.         int mix_map_mask = get_bits(gb, out_ch);

  1363.         int num_coeffs = av_popcount(mix_map_mask);

  1364.         skip_bits_long(gb, num_coeffs * 6);

  1365.     }

  1366. }

  1367. 

  1368. /**

  1369.  * Parse extension substream asset header (HD)

  1370.  */

  1371. static int dca_exss_parse_asset_header(DCAContext *s)

  1372. {

  1373.     int header_pos = get_bits_count(&s->gb);

  1374.     int header_size;

  1375.     int channels;

  1376.     int embedded_stereo = 0;

  1377.     int embedded_6ch = 0;

  1378.     int drc_code_present;

  1379.     int extensions_mask;

  1380.     int i, j;

  1381. 

  1382.     if (get_bits_left(&s->gb) < 16)

  1383.         return -1;

  1384. 

  1385.     /* We will parse just enough to get to the extensions bitmask with which

  1386.      * we can set the profile value. */

  1387. 

  1388.     header_size = get_bits(&s->gb, 9) + 1;

  1389.     skip_bits(&s->gb, 3); // asset index

  1390. 

  1391.     if (s->static_fields) {

  1392.         if (get_bits1(&s->gb))

  1393.             skip_bits(&s->gb, 4); // asset type descriptor

  1394.         if (get_bits1(&s->gb))

  1395.             skip_bits_long(&s->gb, 24); // language descriptor

  1396. 

  1397.         if (get_bits1(&s->gb)) {

  1398.             /* How can one fit 1024 bytes of text here if the maximum value

  1399.              * for the asset header size field above was 512 bytes? */

  1400.             int text_length = get_bits(&s->gb, 10) + 1;

  1401.             if (get_bits_left(&s->gb) < text_length * 8)

  1402.                 return -1;

  1403.             skip_bits_long(&s->gb, text_length * 8); // info text

  1404.         }

  1405. 

  1406.         skip_bits(&s->gb, 5); // bit resolution - 1

  1407.         skip_bits(&s->gb, 4); // max sample rate code

  1408.         channels = get_bits(&s->gb, 8) + 1;

  1409. 

  1410.         if (get_bits1(&s->gb)) { // 1-to-1 channels to speakers

  1411.             int spkr_remap_sets;

  1412.             int spkr_mask_size = 16;

  1413.             int num_spkrs[7];

  1414. 

  1415.             if (channels > 2)

  1416.                 embedded_stereo = get_bits1(&s->gb);

  1417.             if (channels > 6)

  1418.                 embedded_6ch = get_bits1(&s->gb);

  1419. 

  1420.             if (get_bits1(&s->gb)) {

  1421.                 spkr_mask_size = (get_bits(&s->gb, 2) + 1) << 2;

  1422.                 skip_bits(&s->gb, spkr_mask_size); // spkr activity mask

  1423.             }

  1424. 

  1425.             spkr_remap_sets = get_bits(&s->gb, 3);

  1426. 

  1427.             for (i = 0; i < spkr_remap_sets; i++) {

  1428.                 /* std layout mask for each remap set */

  1429.                 num_spkrs[i] = dca_exss_mask2count(get_bits(&s->gb, spkr_mask_size));

  1430.             }

  1431. 

  1432.             for (i = 0; i < spkr_remap_sets; i++) {

  1433.                 int num_dec_ch_remaps = get_bits(&s->gb, 5) + 1;

  1434.                 if (get_bits_left(&s->gb) < 0)

  1435.                     return -1;

  1436. 

  1437.                 for (j = 0; j < num_spkrs[i]; j++) {

  1438.                     int remap_dec_ch_mask = get_bits_long(&s->gb, num_dec_ch_remaps);

  1439.                     int num_dec_ch = av_popcount(remap_dec_ch_mask);

  1440.                     skip_bits_long(&s->gb, num_dec_ch * 5); // remap codes

  1441.                 }

  1442.             }

  1443. 

  1444.         } else {

  1445.             skip_bits(&s->gb, 3); // representation type

  1446.         }

  1447.     }

  1448. 

  1449.     drc_code_present = get_bits1(&s->gb);

  1450.     if (drc_code_present)

  1451.         get_bits(&s->gb, 8); // drc code

  1452. 

  1453.     if (get_bits1(&s->gb))

  1454.         skip_bits(&s->gb, 5); // dialog normalization code

  1455. 

  1456.     if (drc_code_present && embedded_stereo)

  1457.         get_bits(&s->gb, 8); // drc stereo code

  1458. 

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

  1460.         skip_bits(&s->gb, 1); // external mix

  1461.         skip_bits(&s->gb, 6); // post mix gain code

  1462. 

  1463.         if (get_bits(&s->gb, 2) != 3) // mixer drc code

  1464.             skip_bits(&s->gb, 3); // drc limit

  1465.         else

  1466.             skip_bits(&s->gb, 8); // custom drc code

  1467. 

  1468.         if (get_bits1(&s->gb)) // channel specific scaling

  1469.             for (i = 0; i < s->num_mix_configs; i++)

  1470.                 skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6); // scale codes

  1471.         else

  1472.             skip_bits_long(&s->gb, s->num_mix_configs * 6); // scale codes

  1473. 

  1474.         for (i = 0; i < s->num_mix_configs; i++) {

  1475.             if (get_bits_left(&s->gb) < 0)

  1476.                 return -1;

  1477.             dca_exss_skip_mix_coeffs(&s->gb, channels, s->mix_config_num_ch[i]);

  1478.             if (embedded_6ch)

  1479.                 dca_exss_skip_mix_coeffs(&s->gb, 6, s->mix_config_num_ch[i]);

  1480.             if (embedded_stereo)

  1481.                 dca_exss_skip_mix_coeffs(&s->gb, 2, s->mix_config_num_ch[i]);

  1482.         }

  1483.     }

  1484. 

  1485.     switch (get_bits(&s->gb, 2)) {

  1486.     case 0: extensions_mask = get_bits(&s->gb, 12); break;

  1487.     case 1: extensions_mask = DCA_EXT_EXSS_XLL;     break;

  1488.     case 2: extensions_mask = DCA_EXT_EXSS_LBR;     break;

  1489.     case 3: extensions_mask = 0; /* aux coding */   break;

  1490.     }

  1491. 

  1492.     /* not parsed further, we were only interested in the extensions mask */

  1493. 

  1494.     if (get_bits_left(&s->gb) < 0)

  1495.         return -1;

  1496. 

  1497.     if (get_bits_count(&s->gb) - header_pos > header_size * 8) {

  1498.         av_log(s->avctx, AV_LOG_WARNING, "Asset header size mismatch.\n");

  1499.         return -1;

  1500.     }

  1501.     skip_bits_long(&s->gb, header_pos + header_size * 8 - get_bits_count(&s->gb));

  1502. 

  1503.     if (extensions_mask & DCA_EXT_EXSS_XLL)

  1504.         s->profile = FF_PROFILE_DTS_HD_MA;

  1505.     else if (extensions_mask & (DCA_EXT_EXSS_XBR | DCA_EXT_EXSS_X96 |

  1506.                                 DCA_EXT_EXSS_XXCH))

  1507.         s->profile = FF_PROFILE_DTS_HD_HRA;

  1508. 

  1509.     if (!(extensions_mask & DCA_EXT_CORE))

  1510.         av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n");

  1511.     if (!!(extensions_mask & DCA_EXT_XCH) != s->xch_present)

  1512.         av_log(s->avctx, AV_LOG_WARNING, "DTS XCh detection mismatch.\n");

  1513.     if (!!(extensions_mask & DCA_EXT_XXCH) != s->xxch_present)

  1514.         av_log(s->avctx, AV_LOG_WARNING, "DTS XXCh detection mismatch.\n");

  1515.     if (!!(extensions_mask & DCA_EXT_X96) != s->x96_present)

  1516.         av_log(s->avctx, AV_LOG_WARNING, "DTS X96 detection mismatch.\n");

  1517. 

  1518.     return 0;

  1519. }

  1520. 

  1521. /**

  1522.  * Parse extension substream header (HD)

  1523.  */

  1524. static void dca_exss_parse_header(DCAContext *s)

  1525. {

  1526.     int ss_index;

  1527.     int blownup;

  1528.     int header_size;

  1529.     int hd_size;

  1530.     int num_audiop = 1;

  1531.     int num_assets = 1;

  1532.     int active_ss_mask[8];

  1533.     int i, j;

  1534. 

  1535.     if (get_bits_left(&s->gb) < 52)

  1536.         return;

  1537. 

  1538.     skip_bits(&s->gb, 8); // user data

  1539.     ss_index = get_bits(&s->gb, 2);

  1540. 

  1541.     blownup = get_bits1(&s->gb);

  1542.     header_size = get_bits(&s->gb, 8 + 4 * blownup) + 1;

  1543.     hd_size = get_bits_long(&s->gb, 16 + 4 * blownup) + 1;

  1544. 

  1545.     s->static_fields = get_bits1(&s->gb);

  1546.     if (s->static_fields) {

  1547.         skip_bits(&s->gb, 2); // reference clock code

  1548.         skip_bits(&s->gb, 3); // frame duration code

  1549. 

  1550.         if (get_bits1(&s->gb))

  1551.             skip_bits_long(&s->gb, 36); // timestamp

  1552. 

  1553.         /* a single stream can contain multiple audio assets that can be

  1554.          * combined to form multiple audio presentations */

  1555. 

  1556.         num_audiop = get_bits(&s->gb, 3) + 1;

  1557.         if (num_audiop > 1) {

  1558.             av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio presentations.");

  1559.             /* ignore such streams for now */

  1560.             return;

  1561.         }

  1562. 

  1563.         num_assets = get_bits(&s->gb, 3) + 1;

  1564.         if (num_assets > 1) {

  1565.             av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio assets.");

  1566.             /* ignore such streams for now */

  1567.             return;

  1568.         }

  1569. 

  1570.         for (i = 0; i < num_audiop; i++)

  1571.             active_ss_mask[i] = get_bits(&s->gb, ss_index + 1);

  1572. 

  1573.         for (i = 0; i < num_audiop; i++)

  1574.             for (j = 0; j <= ss_index; j++)

  1575.                 if (active_ss_mask[i] & (1 << j))

  1576.                     skip_bits(&s->gb, 8); // active asset mask

  1577. 

  1578.         s->mix_metadata = get_bits1(&s->gb);

  1579.         if (s->mix_metadata) {

  1580.             int mix_out_mask_size;

  1581. 

  1582.             skip_bits(&s->gb, 2); // adjustment level

  1583.             mix_out_mask_size = (get_bits(&s->gb, 2) + 1) << 2;

  1584.             s->num_mix_configs = get_bits(&s->gb, 2) + 1;

  1585. 

  1586.             for (i = 0; i < s->num_mix_configs; i++) {

  1587.                 int mix_out_mask = get_bits(&s->gb, mix_out_mask_size);

  1588.                 s->mix_config_num_ch[i] = dca_exss_mask2count(mix_out_mask);

  1589.             }

  1590.         }

  1591.     }

  1592. 

  1593.     for (i = 0; i < num_assets; i++)

  1594.         skip_bits_long(&s->gb, 16 + 4 * blownup); // asset size

  1595. 

  1596.     for (i = 0; i < num_assets; i++) {

  1597.         if (dca_exss_parse_asset_header(s))

  1598.             return;

  1599.     }

  1600. 

  1601.     /* not parsed further, we were only interested in the extensions mask

  1602.      * from the asset header */

  1603. }

  1604. 

  1605. /**

  1606.  * Main frame decoding function

  1607.  * FIXME add arguments

  1608.  */

  1609. static int dca_decode_frame(AVCodecContext * avctx,

  1610.                             void *data, int *data_size,

  1611.                             AVPacket *avpkt)

  1612. {

  1613.     const uint8_t *buf = avpkt->data;

  1614.     int buf_size = avpkt->size;

  1615. 

  1616.     int lfe_samples;

  1617.     int num_core_channels = 0;

  1618.     int i;

  1619.     int16_t *samples = data;

  1620.     DCAContext *s = avctx->priv_data;

  1621.     int channels;

  1622.     int core_ss_end;

  1623. 

  1624. 

  1625.     s->xch_present = 0;

  1626.     s->x96_present = 0;

  1627.     s->xxch_present = 0;

  1628. 

  1629.     s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer,

  1630.                                                DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);

  1631.     if (s->dca_buffer_size == -1) {

  1632.         av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");

  1633.         return -1;

  1634.     }

  1635. 

  1636.     init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);

  1637.     if (dca_parse_frame_header(s) < 0) {

  1638.         //seems like the frame is corrupt, try with the next one

  1639.         *data_size=0;

  1640.         return buf_size;

  1641.     }

  1642.     //set AVCodec values with parsed data

  1643.     avctx->sample_rate = s->sample_rate;

  1644.     avctx->bit_rate = s->bit_rate;

  1645. 

  1646.     s->profile = FF_PROFILE_DTS;

  1647. 

  1648.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1649.         dca_decode_block(s, 0, i);

  1650.     }

  1651. 

  1652.     /* record number of core channels incase less than max channels are requested */

  1653.     num_core_channels = s->prim_channels;

  1654. 

  1655.     /* extensions start at 32-bit boundaries into bitstream */

  1656.     skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1657. 

  1658.     core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;

  1659. 

  1660.     while(core_ss_end - get_bits_count(&s->gb) >= 32) {

  1661.         uint32_t bits = get_bits_long(&s->gb, 32);

  1662. 

  1663.         switch(bits) {

  1664.         case 0x5a5a5a5a: {

  1665.             int ext_amode, xch_fsize;

  1666. 

  1667.             s->xch_base_channel = s->prim_channels;

  1668. 

  1669.             /* validate sync word using XCHFSIZE field */

  1670.             xch_fsize = show_bits(&s->gb, 10);

  1671.             if((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&

  1672.                (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))

  1673.                 continue;

  1674. 

  1675.             /* skip length-to-end-of-frame field for the moment */

  1676.             skip_bits(&s->gb, 10);

  1677. 

  1678.             s->profile = FFMAX(s->profile, FF_PROFILE_DTS_ES);

  1679. 

  1680.             /* extension amode should == 1, number of channels in extension */

  1681.             /* AFAIK XCh is not used for more channels */

  1682.             if ((ext_amode = get_bits(&s->gb, 4)) != 1) {

  1683.                 av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"

  1684.                        " supported!\n",ext_amode);

  1685.                 continue;

  1686.             }

  1687. 

  1688.             /* much like core primary audio coding header */

  1689.             dca_parse_audio_coding_header(s, s->xch_base_channel);

  1690. 

  1691.             for (i = 0; i < (s->sample_blocks / 8); i++) {

  1692.                 dca_decode_block(s, s->xch_base_channel, i);

  1693.             }

  1694. 

  1695.             s->xch_present = 1;

  1696.             break;

  1697.         }

  1698.         case 0x47004a03:

  1699.             /* XXCh: extended channels */

  1700.             /* usually found either in core or HD part in DTS-HD HRA streams,

  1701.              * but not in DTS-ES which contains XCh extensions instead */

  1702.             s->xxch_present = 1;

  1703.             s->profile = FFMAX(s->profile, FF_PROFILE_DTS_ES);

  1704.             break;

  1705. 

  1706.         case 0x1d95f262: {

  1707.             int fsize96 = show_bits(&s->gb, 12) + 1;

  1708.             if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)

  1709.                 continue;

  1710. 

  1711.             av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n", get_bits_count(&s->gb));

  1712.             skip_bits(&s->gb, 12);

  1713.             av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);

  1714.             av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));

  1715. 

  1716.             s->x96_present = 1;

  1717.             s->profile = FFMAX(s->profile, FF_PROFILE_DTS_96_24);

  1718.             break;

  1719.         }

  1720.         }

  1721. 

  1722.         skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1723.     }

  1724. 

  1725.     /* check for ExSS (HD part) */

  1726.     if (s->dca_buffer_size - s->frame_size > 32

  1727.         && get_bits_long(&s->gb, 32) == DCA_HD_MARKER)

  1728.         dca_exss_parse_header(s);

  1729. 

  1730.     avctx->profile = s->profile;

  1731. 

  1732.     channels = s->prim_channels + !!s->lfe;

  1733. 

  1734.     if (s->amode<16) {

  1735.         avctx->channel_layout = dca_core_channel_layout[s->amode];

  1736. 

  1737.         if (s->xch_present && (!avctx->request_channels ||

  1738.                                avctx->request_channels > num_core_channels + !!s->lfe)) {

  1739.             avctx->channel_layout |= AV_CH_BACK_CENTER;

  1740.             if (s->lfe) {

  1741.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1742.                 s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode];

  1743.             } else {

  1744.                 s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];

  1745.             }

  1746.         } else {

  1747.             channels = num_core_channels + !!s->lfe;

  1748.             s->xch_present = 0; /* disable further xch processing */

  1749.             if (s->lfe) {

  1750.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1751.                 s->channel_order_tab = dca_channel_reorder_lfe[s->amode];

  1752.             } else

  1753.                 s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];

  1754.         }

  1755. 

  1756.         if (channels > !!s->lfe &&

  1757.             s->channel_order_tab[channels - 1 - !!s->lfe] < 0)

  1758.             return -1;

  1759. 

  1760.         if (avctx->request_channels == 2 && s->prim_channels > 2) {

  1761.             channels = 2;

  1762.             s->output = DCA_STEREO;

  1763.             avctx->channel_layout = AV_CH_LAYOUT_STEREO;

  1764.         }

  1765.     } else {

  1766.         av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n",s->amode);

  1767.         return -1;

  1768.     }

  1769. 

  1770. 

  1771.     /* There is nothing that prevents a dts frame to change channel configuration

  1772.        but FFmpeg doesn't support that so only set the channels if it is previously

  1773.        unset. Ideally during the first probe for channels the crc should be checked

  1774.        and only set avctx->channels when the crc is ok. Right now the decoder could

  1775.        set the channels based on a broken first frame.*/

  1776.     if (s->is_channels_set == 0) {

  1777.         s->is_channels_set = 1;

  1778.         avctx->channels = channels;

  1779.     }

  1780.     if (avctx->channels != channels) {

  1781.         av_log(avctx, AV_LOG_ERROR, "DCA decoder does not support number of "

  1782.                "channels changing in stream. Skipping frame.\n");

  1783.         return -1;

  1784.     }

  1785. 

  1786.     if (*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)

  1787.         return -1;

  1788.     *data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels;

  1789. 

  1790.     /* filter to get final output */

  1791.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1792.         dca_filter_channels(s, i);

  1793. 

  1794.         /* If this was marked as a DTS-ES stream we need to subtract back- */

  1795.         /* channel from SL & SR to remove matrixed back-channel signal */

  1796.         if((s->source_pcm_res & 1) && s->xch_present) {

  1797.             float* back_chan = s->samples + s->channel_order_tab[s->xch_base_channel] * 256;

  1798.             float* lt_chan   = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256;

  1799.             float* rt_chan   = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;

  1800.             int j;

  1801.             for(j = 0; j < 256; ++j) {

  1802.                 lt_chan[j] -= back_chan[j] * M_SQRT1_2;

  1803.                 rt_chan[j] -= back_chan[j] * M_SQRT1_2;

  1804.             }

  1805.         }

  1806. 

  1807.         s->fmt_conv.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels);

  1808.         samples += 256 * channels;

  1809.     }

  1810. 

  1811.     /* update lfe history */

  1812.     lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);

  1813.     for (i = 0; i < 2 * s->lfe * 4; i++) {

  1814.         s->lfe_data[i] = s->lfe_data[i + lfe_samples];

  1815.     }

  1816. 

  1817.     return buf_size;

  1818. }

  1819. 

  1820. 

  1821. 

  1822. /**

  1823.  * DCA initialization

  1824.  *

  1825.  * @param avctx     pointer to the AVCodecContext

  1826.  */

  1827. 

  1828. static av_cold int dca_decode_init(AVCodecContext * avctx)

  1829. {

  1830.     DCAContext *s = avctx->priv_data;

  1831.     int i;

  1832. 

  1833.     s->avctx = avctx;

  1834.     dca_init_vlcs();

  1835. 

  1836.     dsputil_init(&s->dsp, avctx);

  1837.     ff_mdct_init(&s->imdct, 6, 1, 1.0);

  1838.     ff_synth_filter_init(&s->synth);

  1839.     ff_dcadsp_init(&s->dcadsp);

  1840.     ff_fmt_convert_init(&s->fmt_conv, avctx);

  1841. 

  1842.     for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)

  1843.         s->samples_chanptr[i] = s->samples + i * 256;

  1844.     avctx->sample_fmt = AV_SAMPLE_FMT_S16;

  1845. 

  1846.     s->scale_bias = 1.0;

  1847. 

  1848.     /* allow downmixing to stereo */

  1849.     if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&

  1850.         avctx->request_channels == 2) {

  1851.         avctx->channels = avctx->request_channels;

  1852.     }

  1853. 

  1854.     return 0;

  1855. }

  1856. 

  1857. static av_cold int dca_decode_end(AVCodecContext * avctx)

  1858. {

  1859.     DCAContext *s = avctx->priv_data;

  1860.     ff_mdct_end(&s->imdct);

  1861.     return 0;

  1862. }

  1863. 

  1864. static const AVProfile profiles[] = {

  1865.     { FF_PROFILE_DTS,        "DTS"        },

  1866.     { FF_PROFILE_DTS_ES,     "DTS-ES"     },

  1867.     { FF_PROFILE_DTS_96_24,  "DTS 96/24"  },

  1868.     { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },

  1869.     { FF_PROFILE_DTS_HD_MA,  "DTS-HD MA"  },

  1870.     { FF_PROFILE_UNKNOWN },

  1871. };

  1872. 

  1873. AVCodec ff_dca_decoder = {

  1874.     .name = "dca",

  1875.     .type = AVMEDIA_TYPE_AUDIO,

  1876.     .id = CODEC_ID_DTS,

  1877.     .priv_data_size = sizeof(DCAContext),

  1878.     .init = dca_decode_init,

  1879.     .decode = dca_decode_frame,

  1880.     .close = dca_decode_end,

  1881.     .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),

  1882.     .capabilities = CODEC_CAP_CHANNEL_CONF,

  1883.     .profiles = NULL_IF_CONFIG_SMALL(profiles),

  1884. };