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FFmpeg/libavcodec/dca.c

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  1. /*

  2.  * DCA compatible decoder

  3.  * Copyright (C) 2004 Gildas Bazin

  4.  * Copyright (C) 2004 Benjamin Zores

  5.  * Copyright (C) 2006 Benjamin Larsson

  6.  * Copyright (C) 2007 Konstantin Shishkov

  7.  *

  8.  * This file is part of FFmpeg.

  9.  *

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

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

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

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

  14.  *

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

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

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

  18.  * Lesser General Public License for more details.

  19.  *

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

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

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

  23.  */

  24. 

  25. #include <math.h>

  26. #include <stddef.h>

  27. #include <stdio.h>

  28. 

  29. #include "libavutil/common.h"

  30. #include "libavutil/intmath.h"

  31. #include "libavutil/intreadwrite.h"

  32. #include "libavutil/audioconvert.h"

  33. #include "avcodec.h"

  34. #include "dsputil.h"

  35. #include "fft.h"

  36. #include "get_bits.h"

  37. #include "put_bits.h"

  38. #include "dcadata.h"

  39. #include "dcahuff.h"

  40. #include "dca.h"

  41. #include "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. 

  131. static const int64_t dca_core_channel_layout[] = {

  132.     AV_CH_FRONT_CENTER,                                                      ///< 1, A

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

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

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

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

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

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

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

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

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

  142.     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

  143.     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

  144.     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

  145.     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

  146.     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

  147.     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

  148. };

  149. 

  150. static const int8_t dca_lfe_index[] = {

  151.     1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3

  152. };

  153. 

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

  171. };

  172. 

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

  174.     { 0,  2, -1, -1, -1, -1, -1, -1, -1},

  175.     { 0,  1,  3, -1, -1, -1, -1, -1, -1},

  176.     { 0,  1,  3, -1, -1, -1, -1, -1, -1},

  177.     { 0,  1,  3, -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.     { 0,  1,  4,  5,  3, -1, -1, -1, -1},

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

  184.     { 3,  4,  0,  1,  6,  7,  5, -1, -1},

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

  186.     { 0,  6,  4,  5,  2,  3,  7, -1, -1},

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

  188.     { 5,  6,  0,  1,  8,  3,  9,  4,  7},

  189.     { 4,  2,  5,  0,  1,  6,  9,  8,  7},

  190. };

  191. 

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

  209. };

  210. 

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

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

  213.     { 0,  1,  2, -1, -1, -1, -1, -1, -1},

  214.     { 0,  1,  2, -1, -1, -1, -1, -1, -1},

  215.     { 0,  1,  2, -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.     { 0,  1,  3,  4,  2, -1, -1, -1, -1},

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

  222.     { 2,  3,  0,  1,  5,  6,  4, -1, -1},

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

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

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

  226.     { 4,  5,  0,  1,  7,  2,  8,  3,  6},

  227.     { 3,  2,  4,  0,  1,  5,  8,  7,  6},

  228. };

  229. 

  230. #define DCA_DOLBY 101           /* FIXME */

  231. 

  232. #define DCA_CHANNEL_BITS 6

  233. #define DCA_CHANNEL_MASK 0x3F

  234. 

  235. #define DCA_LFE 0x80

  236. 

  237. #define HEADER_SIZE 14

  238. 

  239. #define DCA_MAX_FRAME_SIZE 16384

  240. #define DCA_MAX_EXSS_HEADER_SIZE 4096

  241. 

  242. #define DCA_BUFFER_PADDING_SIZE 1024

  243. 

  244. /** Bit allocation */

  245. typedef struct {

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

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

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

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

  250. } BitAlloc;

  251. 

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

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

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

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

  256. 

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

  258. {

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

  260. }

  261. 

  262. typedef struct {

  263.     AVCodecContext *avctx;

  264.     /* Frame header */

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

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

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

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

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

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

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

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

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

  274. 

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

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

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

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

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

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

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

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

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

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

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

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

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

  288.     int copy_history;           ///< copy history

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

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

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

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

  293. 

  294.     /* Primary audio coding header */

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

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

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

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

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

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

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

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

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

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

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

  306. 

  307.     /* Primary audio coding side information */

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

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

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

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

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

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

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

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

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

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

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

  319. 

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

  321. 

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

  323.     int lfe_scale_factor;

  324. 

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

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

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

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

  329.     int hist_index[DCA_PRIM_CHANNELS_MAX];

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

  331. 

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

  333.     float scale_bias;           ///< output scale

  334. 

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

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

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

  338. 

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

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

  341. 

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

  343.     GetBitContext gb;

  344.     /* Current position in DCA frame */

  345.     int current_subframe;

  346.     int current_subsubframe;

  347. 

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

  349. 

  350.     /* XCh extension information */

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

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

  353. 

  354.     /* ExSS header parser */

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

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

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

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

  359. 

  360.     int profile;

  361. 

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

  363.     DSPContext dsp;

  364.     FFTContext imdct;

  365.     SynthFilterContext synth;

  366.     DCADSPContext dcadsp;

  367.     FmtConvertContext fmt_conv;

  368. } DCAContext;

  369. 

  370. static const uint16_t dca_vlc_offs[] = {

  371.         0,   512,   640,   768,  1282,  1794,  2436,  3080,  3770,  4454,  5364,

  372.      5372,  5380,  5388,  5392,  5396,  5412,  5420,  5428,  5460,  5492,  5508,

  373.      5572,  5604,  5668,  5796,  5860,  5892,  6412,  6668,  6796,  7308,  7564,

  374.      7820,  8076,  8620,  9132,  9388,  9910, 10166, 10680, 11196, 11726, 12240,

  375.     12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264,

  376.     18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622,

  377. };

  378. 

  379. static av_cold void dca_init_vlcs(void)

  380. {

  381.     static int vlcs_initialized = 0;

  382.     int i, j, c = 14;

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

  384. 

  385.     if (vlcs_initialized)

  386.         return;

  387. 

  388.     dca_bitalloc_index.offset = 1;

  389.     dca_bitalloc_index.wrap = 2;

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

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

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

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

  394.                  bitalloc_12_bits[i], 1, 1,

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

  396.     }

  397.     dca_scalefactor.offset = -64;

  398.     dca_scalefactor.wrap = 2;

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

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

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

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

  403.                  scales_bits[i], 1, 1,

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

  405.     }

  406.     dca_tmode.offset = 0;

  407.     dca_tmode.wrap = 1;

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

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

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

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

  412.                  tmode_bits[i], 1, 1,

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

  414.     }

  415. 

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

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

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

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

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

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

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

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

  424.                      bitalloc_sizes[i],

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

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

  427.             c++;

  428.         }

  429.     vlcs_initialized = 1;

  430. }

  431. 

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

  433. {

  434.     while(len--)

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

  436. }

  437. 

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

  439. {

  440.     int i, j;

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

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

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

  444. 

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

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

  447. 

  448.     if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)

  449.         s->prim_channels = DCA_PRIM_CHANNELS_MAX;

  450. 

  451. 

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

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

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

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

  456.     }

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

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

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

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

  461.     }

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

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

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

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

  466. 

  467.     /* Get codebooks quantization indexes */

  468.     if (!base_channel)

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

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

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

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

  473. 

  474.     /* Get scale factor adjustment */

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

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

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

  478. 

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

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

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

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

  483. 

  484.     if (s->crc_present) {

  485.         /* Audio header CRC check */

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

  487.     }

  488. 

  489.     s->current_subframe = 0;

  490.     s->current_subsubframe = 0;

  491. 

  492. #ifdef TRACE

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

  511.     }

  512. #endif

  513. 

  514.   return 0;

  515. }

  516. 

  517. static int dca_parse_frame_header(DCAContext * s)

  518. {

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

  520. 

  521.     /* Sync code */

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

  523. 

  524.     /* Frame header */

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

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

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

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

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

  530.     if (s->frame_size < 95)

  531.         return AVERROR_INVALIDDATA;

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

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

  534.     if (!s->sample_rate)

  535.         return AVERROR_INVALIDDATA;

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

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

  538.     if (!s->bit_rate)

  539.         return AVERROR_INVALIDDATA;

  540. 

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

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

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

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

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

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

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

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

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

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

  551. 

  552.     /* TODO: check CRC */

  553.     if (s->crc_present)

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

  555. 

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

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

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

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

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

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

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

  563. 

  564.     /* FIXME: channels mixing levels */

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

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

  567. 

  568. #ifdef TRACE

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

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

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

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

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

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

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

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

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

  578.            s->sample_rate);

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

  580.            s->bit_rate);

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

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

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

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

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

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

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

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

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

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

  591.            s->predictor_history);

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

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

  594.            s->multirate_inter);

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

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

  597.     av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

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

  604. #endif

  605. 

  606.     /* Primary audio coding header */

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

  608. 

  609.     return dca_parse_audio_coding_header(s, 0);

  610. }

  611. 

  612. 

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

  614. {

  615.    if (level < 5) {

  616.        /* huffman encoded */

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

  618.    } else if (level < 8)

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

  620.    return value;

  621. }

  622. 

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

  624. {

  625.     /* Primary audio coding side information */

  626.     int j, k;

  627. 

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

  629.         return AVERROR_INVALIDDATA;

  630. 

  631.     if (!base_channel) {

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

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

  634.     }

  635. 

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

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

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

  639.     }

  640. 

  641.     /* Get prediction codebook */

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

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

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

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

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

  647.             }

  648.         }

  649.     }

  650. 

  651.     /* Bit allocation index */

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

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

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

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

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

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

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

  659.                 av_log(s->avctx, AV_LOG_ERROR,

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

  661.                 return AVERROR_INVALIDDATA;

  662.             } else {

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

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

  665.             }

  666. 

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

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

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

  670.                 return AVERROR_INVALIDDATA;

  671.             }

  672.         }

  673.     }

  674. 

  675.     /* Transition mode */

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

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

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

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

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

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

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

  683.             }

  684.         }

  685.     }

  686. 

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

  688.         return AVERROR_INVALIDDATA;

  689. 

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

  691.         const uint32_t *scale_table;

  692.         int scale_sum;

  693. 

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

  695. 

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

  697.             scale_table = scale_factor_quant7;

  698.         else

  699.             scale_table = scale_factor_quant6;

  700. 

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

  702.         scale_sum = 0;

  703. 

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

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

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

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

  708.             }

  709. 

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

  711.                 /* Get second scale factor */

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

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

  714.             }

  715.         }

  716.     }

  717. 

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

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

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

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

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

  723.     }

  724. 

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

  726.         return AVERROR_INVALIDDATA;

  727. 

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

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

  730.         int source_channel;

  731. 

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

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

  734.             int scale = 0;

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

  736. 

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

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

  739. 

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

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

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

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

  744.             }

  745. 

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

  747.                 av_log(s->avctx, AV_LOG_DEBUG,

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

  749.                 s->debug_flag |= 0x02;

  750.             }

  751.         }

  752.     }

  753. 

  754.     /* Stereo downmix coefficients */

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

  756.         if (s->downmix) {

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

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

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

  760.             }

  761.         } else {

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

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

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

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

  766.             }

  767.         }

  768.     }

  769. 

  770.     /* Dynamic range coefficient */

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

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

  773. 

  774.     /* Side information CRC check word */

  775.     if (s->crc_present) {

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

  777.     }

  778. 

  779.     /*

  780.      * Primary audio data arrays

  781.      */

  782. 

  783.     /* VQ encoded high frequency subbands */

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

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

  786.             /* 1 vector -> 32 samples */

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

  788. 

  789.     /* Low frequency effect data */

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

  791.         /* LFE samples */

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

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

  794.         float lfe_scale;

  795. 

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

  797.             /* Signed 8 bits int */

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

  799.         }

  800. 

  801.         /* Scale factor index */

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

  803. 

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

  805.         lfe_scale = 0.035 * s->lfe_scale_factor;

  806. 

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

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

  809.     }

  810. 

  811. #ifdef TRACE

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

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

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

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

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

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

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

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

  820.     }

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

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

  823.                 av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

  829.     }

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

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

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

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

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

  835.     }

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

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

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

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

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

  841.     }

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

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

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

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

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

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

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

  849.         }

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

  851.     }

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

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

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

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

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

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

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

  859.         }

  860.     }

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

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

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

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

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

  866.         }

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

  868.     }

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

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

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

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

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

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

  875. 

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

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

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

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

  880.     }

  881. #endif

  882. 

  883.     return 0;

  884. }

  885. 

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

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

  888.                             float scale)

  889. {

  890.     const float *prCoeff;

  891.     int i;

  892. 

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

  894.     int subindex;

  895. 

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

  897. 

  898.     /* Select filter */

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

  900.         prCoeff = fir_32bands_nonperfect;

  901.     else                        /* Perfect reconstruction */

  902.         prCoeff = fir_32bands_perfect;

  903. 

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

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

  906. 

  907.     /* Reconstructed channel sample index */

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

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

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

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

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

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

  914.         }

  915. 

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

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

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

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

  920.         samples_out+= 32;

  921. 

  922.     }

  923. }

  924. 

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

  926.                                   int num_deci_sample, float *samples_in,

  927.                                   float *samples_out, float scale)

  928. {

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

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

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

  932.      *   from last subframe as history.

  933.      *

  934.      * samples_out: An array holding interpolated samples

  935.      */

  936. 

  937.     int decifactor;

  938.     const float *prCoeff;

  939.     int deciindex;

  940. 

  941.     /* Select decimation filter */

  942.     if (decimation_select == 1) {

  943.         decifactor = 64;

  944.         prCoeff = lfe_fir_128;

  945.     } else {

  946.         decifactor = 32;

  947.         prCoeff = lfe_fir_64;

  948.     }

  949.     /* Interpolation */

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

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

  952.                           scale);

  953.         samples_in++;

  954.         samples_out += 2 * decifactor;

  955.     }

  956. }

  957. 

  958. /* downmixing routines */

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

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

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

  962. 

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

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

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

  966. 

  967. #define MIX_FRONT3(samples, coef) \

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

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

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

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

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

  973. 

  974. #define DOWNMIX_TO_STEREO(op1, op2) \

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

  976.         op1 \

  977.         op2 \

  978.     }

  979. 

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

  981.                         int downmix_coef[DCA_PRIM_CHANNELS_MAX][2],

  982.                         const int8_t *channel_mapping)

  983. {

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

  985.     int i;

  986.     float t, u, v;

  987.     float coef[DCA_PRIM_CHANNELS_MAX][2];

  988. 

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

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

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

  992.     }

  993. 

  994.     switch (srcfmt) {

  995.     case DCA_MONO:

  996.     case DCA_CHANNEL:

  997.     case DCA_STEREO_TOTAL:

  998.     case DCA_STEREO_SUMDIFF:

  999.     case DCA_4F2R:

  1000.         av_log(NULL, 0, "Not implemented!\n");

  1001.         break;

  1002.     case DCA_STEREO:

  1003.         break;

  1004.     case DCA_3F:

  1005.         c = channel_mapping[0] * 256;

  1006.         l = channel_mapping[1] * 256;

  1007.         r = channel_mapping[2] * 256;

  1008.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),);

  1009.         break;

  1010.     case DCA_2F1R:

  1011.         s = channel_mapping[2] * 256;

  1012.         DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + s, 2, coef),);

  1013.         break;

  1014.     case DCA_3F1R:

  1015.         c = channel_mapping[0] * 256;

  1016.         l = channel_mapping[1] * 256;

  1017.         r = channel_mapping[2] * 256;

  1018.         s = channel_mapping[3] * 256;

  1019.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1020.                           MIX_REAR1(samples, i + s, 3, coef));

  1021.         break;

  1022.     case DCA_2F2R:

  1023.         sl = channel_mapping[2] * 256;

  1024.         sr = channel_mapping[3] * 256;

  1025.         DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + sl, i + sr, 2, coef),);

  1026.         break;

  1027.     case DCA_3F2R:

  1028.         c =  channel_mapping[0] * 256;

  1029.         l =  channel_mapping[1] * 256;

  1030.         r =  channel_mapping[2] * 256;

  1031.         sl = channel_mapping[3] * 256;

  1032.         sr = channel_mapping[4] * 256;

  1033.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1034.                           MIX_REAR2(samples, i + sl, i + sr, 3, coef));

  1035.         break;

  1036.     }

  1037. }

  1038. 

  1039. 

  1040. #ifndef decode_blockcode

  1041. /* Very compact version of the block code decoder that does not use table

  1042.  * look-up but is slightly slower */

  1043. static int decode_blockcode(int code, int levels, int *values)

  1044. {

  1045.     int i;

  1046.     int offset = (levels - 1) >> 1;

  1047. 

  1048.     for (i = 0; i < 4; i++) {

  1049.         int div = FASTDIV(code, levels);

  1050.         values[i] = code - offset - div*levels;

  1051.         code = div;

  1052.     }

  1053. 

  1054.     if (code == 0)

  1055.         return 0;

  1056.     else {

  1057.         av_log(NULL, AV_LOG_ERROR, "ERROR: block code look-up failed\n");

  1058.         return AVERROR_INVALIDDATA;

  1059.     }

  1060. }

  1061. #endif

  1062. 

  1063. static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };

  1064. static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };

  1065. 

  1066. #ifndef int8x8_fmul_int32

  1067. static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)

  1068. {

  1069.     float fscale = scale / 16.0;

  1070.     int i;

  1071.     for (i = 0; i < 8; i++)

  1072.         dst[i] = src[i] * fscale;

  1073. }

  1074. #endif

  1075. 

  1076. static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)

  1077. {

  1078.     int k, l;

  1079.     int subsubframe = s->current_subsubframe;

  1080. 

  1081.     const float *quant_step_table;

  1082. 

  1083.     /* FIXME */

  1084.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1085.     LOCAL_ALIGNED_16(int, block, [8]);

  1086. 

  1087.     /*

  1088.      * Audio data

  1089.      */

  1090. 

  1091.     /* Select quantization step size table */

  1092.     if (s->bit_rate_index == 0x1f)

  1093.         quant_step_table = lossless_quant_d;

  1094.     else

  1095.         quant_step_table = lossy_quant_d;

  1096. 

  1097.     for (k = base_channel; k < s->prim_channels; k++) {

  1098.         if (get_bits_left(&s->gb) < 0)

  1099.             return AVERROR_INVALIDDATA;

  1100. 

  1101.         for (l = 0; l < s->vq_start_subband[k]; l++) {

  1102.             int m;

  1103. 

  1104.             /* Select the mid-tread linear quantizer */

  1105.             int abits = s->bitalloc[k][l];

  1106. 

  1107.             float quant_step_size = quant_step_table[abits];

  1108. 

  1109.             /*

  1110.              * Determine quantization index code book and its type

  1111.              */

  1112. 

  1113.             /* Select quantization index code book */

  1114.             int sel = s->quant_index_huffman[k][abits];

  1115. 

  1116.             /*

  1117.              * Extract bits from the bit stream

  1118.              */

  1119.             if (!abits){

  1120.                 memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));

  1121.             } else {

  1122.                 /* Deal with transients */

  1123.                 int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];

  1124.                 float rscale = quant_step_size * s->scale_factor[k][l][sfi] * s->scalefactor_adj[k][sel];

  1125. 

  1126.                 if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){

  1127.                     if (abits <= 7){

  1128.                         /* Block code */

  1129.                         int block_code1, block_code2, size, levels;

  1130. 

  1131.                         size = abits_sizes[abits-1];

  1132.                         levels = abits_levels[abits-1];

  1133. 

  1134.                         block_code1 = get_bits(&s->gb, size);

  1135.                         /* FIXME Should test return value */

  1136.                         decode_blockcode(block_code1, levels, block);

  1137.                         block_code2 = get_bits(&s->gb, size);

  1138.                         decode_blockcode(block_code2, levels, &block[4]);

  1139.                     }else{

  1140.                         /* no coding */

  1141.                         for (m = 0; m < 8; m++)

  1142.                             block[m] = get_sbits(&s->gb, abits - 3);

  1143.                     }

  1144.                 }else{

  1145.                     /* Huffman coded */

  1146.                     for (m = 0; m < 8; m++)

  1147.                         block[m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);

  1148.                 }

  1149. 

  1150.                 s->fmt_conv.int32_to_float_fmul_scalar(subband_samples[k][l],

  1151.                                                   block, rscale, 8);

  1152.             }

  1153. 

  1154.             /*

  1155.              * Inverse ADPCM if in prediction mode

  1156.              */

  1157.             if (s->prediction_mode[k][l]) {

  1158.                 int n;

  1159.                 for (m = 0; m < 8; m++) {

  1160.                     for (n = 1; n <= 4; n++)

  1161.                         if (m >= n)

  1162.                             subband_samples[k][l][m] +=

  1163.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1164.                                  subband_samples[k][l][m - n] / 8192);

  1165.                         else if (s->predictor_history)

  1166.                             subband_samples[k][l][m] +=

  1167.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1168.                                  s->subband_samples_hist[k][l][m - n +

  1169.                                                                4] / 8192);

  1170.                 }

  1171.             }

  1172.         }

  1173. 

  1174.         /*

  1175.          * Decode VQ encoded high frequencies

  1176.          */

  1177.         for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {

  1178.             /* 1 vector -> 32 samples but we only need the 8 samples

  1179.              * for this subsubframe. */

  1180.             int hfvq = s->high_freq_vq[k][l];

  1181. 

  1182.             if (!s->debug_flag & 0x01) {

  1183.                 av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n");

  1184.                 s->debug_flag |= 0x01;

  1185.             }

  1186. 

  1187.             int8x8_fmul_int32(subband_samples[k][l],

  1188.                               &high_freq_vq[hfvq][subsubframe * 8],

  1189.                               s->scale_factor[k][l][0]);

  1190.         }

  1191.     }

  1192. 

  1193.     /* Check for DSYNC after subsubframe */

  1194.     if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {

  1195.         if (0xFFFF == get_bits(&s->gb, 16)) {   /* 0xFFFF */

  1196. #ifdef TRACE

  1197.             av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");

  1198. #endif

  1199.         } else {

  1200.             av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");

  1201.         }

  1202.     }

  1203. 

  1204.     /* Backup predictor history for adpcm */

  1205.     for (k = base_channel; k < s->prim_channels; k++)

  1206.         for (l = 0; l < s->vq_start_subband[k]; l++)

  1207.             memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4],

  1208.                         4 * sizeof(subband_samples[0][0][0]));

  1209. 

  1210.     return 0;

  1211. }

  1212. 

  1213. static int dca_filter_channels(DCAContext * s, int block_index)

  1214. {

  1215.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1216.     int k;

  1217. 

  1218.     /* 32 subbands QMF */

  1219.     for (k = 0; k < s->prim_channels; k++) {

  1220. /*        static float pcm_to_double[8] =

  1221.             {32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/

  1222.          qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * s->channel_order_tab[k]],

  1223.                          M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ );

  1224.     }

  1225. 

  1226.     /* Down mixing */

  1227.     if (s->avctx->request_channels == 2 && s->prim_channels > 2) {

  1228.         dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab);

  1229.     }

  1230. 

  1231.     /* Generate LFE samples for this subsubframe FIXME!!! */

  1232.     if (s->output & DCA_LFE) {

  1233.         lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,

  1234.                               s->lfe_data + 2 * s->lfe * (block_index + 4),

  1235.                               &s->samples[256 * dca_lfe_index[s->amode]],

  1236.                               (1.0/256.0)*s->scale_bias);

  1237.         /* Outputs 20bits pcm samples */

  1238.     }

  1239. 

  1240.     return 0;

  1241. }

  1242. 

  1243. 

  1244. static int dca_subframe_footer(DCAContext * s, int base_channel)

  1245. {

  1246.     int aux_data_count = 0, i;

  1247. 

  1248.     /*

  1249.      * Unpack optional information

  1250.      */

  1251. 

  1252.     /* presumably optional information only appears in the core? */

  1253.     if (!base_channel) {

  1254.         if (s->timestamp)

  1255.             get_bits(&s->gb, 32);

  1256. 

  1257.         if (s->aux_data)

  1258.             aux_data_count = get_bits(&s->gb, 6);

  1259. 

  1260.         for (i = 0; i < aux_data_count; i++)

  1261.             get_bits(&s->gb, 8);

  1262. 

  1263.         if (s->crc_present && (s->downmix || s->dynrange))

  1264.             get_bits(&s->gb, 16);

  1265.     }

  1266. 

  1267.     return 0;

  1268. }

  1269. 

  1270. /**

  1271.  * Decode a dca frame block

  1272.  *

  1273.  * @param s     pointer to the DCAContext

  1274.  */

  1275. 

  1276. static int dca_decode_block(DCAContext * s, int base_channel, int block_index)

  1277. {

  1278.     int ret;

  1279. 

  1280.     /* Sanity check */

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

  1282.         av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",

  1283.                s->current_subframe, s->subframes);

  1284.         return AVERROR_INVALIDDATA;

  1285.     }

  1286. 

  1287.     if (!s->current_subsubframe) {

  1288. #ifdef TRACE

  1289.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");

  1290. #endif

  1291.         /* Read subframe header */

  1292.         if ((ret = dca_subframe_header(s, base_channel, block_index)))

  1293.             return ret;

  1294.     }

  1295. 

  1296.     /* Read subsubframe */

  1297. #ifdef TRACE

  1298.     av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");

  1299. #endif

  1300.     if ((ret = dca_subsubframe(s, base_channel, block_index)))

  1301.         return ret;

  1302. 

  1303.     /* Update state */

  1304.     s->current_subsubframe++;

  1305.     if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {

  1306.         s->current_subsubframe = 0;

  1307.         s->current_subframe++;

  1308.     }

  1309.     if (s->current_subframe >= s->subframes) {

  1310. #ifdef TRACE

  1311.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");

  1312. #endif

  1313.         /* Read subframe footer */

  1314.         if ((ret = dca_subframe_footer(s, base_channel)))

  1315.             return ret;

  1316.     }

  1317. 

  1318.     return 0;

  1319. }

  1320. 

  1321. /**

  1322.  * Convert bitstream to one representation based on sync marker

  1323.  */

  1324. static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * dst,

  1325.                           int max_size)

  1326. {

  1327.     uint32_t mrk;

  1328.     int i, tmp;

  1329.     const uint16_t *ssrc = (const uint16_t *) src;

  1330.     uint16_t *sdst = (uint16_t *) dst;

  1331.     PutBitContext pb;

  1332. 

  1333.     if ((unsigned)src_size > (unsigned)max_size) {

  1334. //        av_log(NULL, AV_LOG_ERROR, "Input frame size larger than DCA_MAX_FRAME_SIZE!\n");

  1335. //        return -1;

  1336.         src_size = max_size;

  1337.     }

  1338. 

  1339.     mrk = AV_RB32(src);

  1340.     switch (mrk) {

  1341.     case DCA_MARKER_RAW_BE:

  1342.         memcpy(dst, src, src_size);

  1343.         return src_size;

  1344.     case DCA_MARKER_RAW_LE:

  1345.         for (i = 0; i < (src_size + 1) >> 1; i++)

  1346.             *sdst++ = av_bswap16(*ssrc++);

  1347.         return src_size;

  1348.     case DCA_MARKER_14B_BE:

  1349.     case DCA_MARKER_14B_LE:

  1350.         init_put_bits(&pb, dst, max_size);

  1351.         for (i = 0; i < (src_size + 1) >> 1; i++, src += 2) {

  1352.             tmp = ((mrk == DCA_MARKER_14B_BE) ? AV_RB16(src) : AV_RL16(src)) & 0x3FFF;

  1353.             put_bits(&pb, 14, tmp);

  1354.         }

  1355.         flush_put_bits(&pb);

  1356.         return (put_bits_count(&pb) + 7) >> 3;

  1357.     default:

  1358.         return AVERROR_INVALIDDATA;

  1359.     }

  1360. }

  1361. 

  1362. /**

  1363.  * Return the number of channels in an ExSS speaker mask (HD)

  1364.  */

  1365. static int dca_exss_mask2count(int mask)

  1366. {

  1367.     /* count bits that mean speaker pairs twice */

  1368.     return av_popcount(mask)

  1369.         + av_popcount(mask & (

  1370.             DCA_EXSS_CENTER_LEFT_RIGHT

  1371.           | DCA_EXSS_FRONT_LEFT_RIGHT

  1372.           | DCA_EXSS_FRONT_HIGH_LEFT_RIGHT

  1373.           | DCA_EXSS_WIDE_LEFT_RIGHT

  1374.           | DCA_EXSS_SIDE_LEFT_RIGHT

  1375.           | DCA_EXSS_SIDE_HIGH_LEFT_RIGHT

  1376.           | DCA_EXSS_SIDE_REAR_LEFT_RIGHT

  1377.           | DCA_EXSS_REAR_LEFT_RIGHT

  1378.           | DCA_EXSS_REAR_HIGH_LEFT_RIGHT

  1379.           ));

  1380. }

  1381. 

  1382. /**

  1383.  * Skip mixing coefficients of a single mix out configuration (HD)

  1384.  */

  1385. static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch)

  1386. {

  1387.     int i;

  1388. 

  1389.     for (i = 0; i < channels; i++) {

  1390.         int mix_map_mask = get_bits(gb, out_ch);

  1391.         int num_coeffs = av_popcount(mix_map_mask);

  1392.         skip_bits_long(gb, num_coeffs * 6);

  1393.     }

  1394. }

  1395. 

  1396. /**

  1397.  * Parse extension substream asset header (HD)

  1398.  */

  1399. static int dca_exss_parse_asset_header(DCAContext *s)

  1400. {

  1401.     int header_pos = get_bits_count(&s->gb);

  1402.     int header_size;

  1403.     int channels;

  1404.     int embedded_stereo = 0;

  1405.     int embedded_6ch = 0;

  1406.     int drc_code_present;

  1407.     int extensions_mask;

  1408.     int i, j;

  1409. 

  1410.     if (get_bits_left(&s->gb) < 16)

  1411.         return -1;

  1412. 

  1413.     /* We will parse just enough to get to the extensions bitmask with which

  1414.      * we can set the profile value. */

  1415. 

  1416.     header_size = get_bits(&s->gb, 9) + 1;

  1417.     skip_bits(&s->gb, 3); // asset index

  1418. 

  1419.     if (s->static_fields) {

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

  1421.             skip_bits(&s->gb, 4); // asset type descriptor

  1422.         if (get_bits1(&s->gb))

  1423.             skip_bits_long(&s->gb, 24); // language descriptor

  1424. 

  1425.         if (get_bits1(&s->gb)) {

  1426.             /* How can one fit 1024 bytes of text here if the maximum value

  1427.              * for the asset header size field above was 512 bytes? */

  1428.             int text_length = get_bits(&s->gb, 10) + 1;

  1429.             if (get_bits_left(&s->gb) < text_length * 8)

  1430.                 return -1;

  1431.             skip_bits_long(&s->gb, text_length * 8); // info text

  1432.         }

  1433. 

  1434.         skip_bits(&s->gb, 5); // bit resolution - 1

  1435.         skip_bits(&s->gb, 4); // max sample rate code

  1436.         channels = get_bits(&s->gb, 8) + 1;

  1437. 

  1438.         if (get_bits1(&s->gb)) { // 1-to-1 channels to speakers

  1439.             int spkr_remap_sets;

  1440.             int spkr_mask_size = 16;

  1441.             int num_spkrs[7];

  1442. 

  1443.             if (channels > 2)

  1444.                 embedded_stereo = get_bits1(&s->gb);

  1445.             if (channels > 6)

  1446.                 embedded_6ch = get_bits1(&s->gb);

  1447. 

  1448.             if (get_bits1(&s->gb)) {

  1449.                 spkr_mask_size = (get_bits(&s->gb, 2) + 1) << 2;

  1450.                 skip_bits(&s->gb, spkr_mask_size); // spkr activity mask

  1451.             }

  1452. 

  1453.             spkr_remap_sets = get_bits(&s->gb, 3);

  1454. 

  1455.             for (i = 0; i < spkr_remap_sets; i++) {

  1456.                 /* std layout mask for each remap set */

  1457.                 num_spkrs[i] = dca_exss_mask2count(get_bits(&s->gb, spkr_mask_size));

  1458.             }

  1459. 

  1460.             for (i = 0; i < spkr_remap_sets; i++) {

  1461.                 int num_dec_ch_remaps = get_bits(&s->gb, 5) + 1;

  1462.                 if (get_bits_left(&s->gb) < 0)

  1463.                     return -1;

  1464. 

  1465.                 for (j = 0; j < num_spkrs[i]; j++) {

  1466.                     int remap_dec_ch_mask = get_bits_long(&s->gb, num_dec_ch_remaps);

  1467.                     int num_dec_ch = av_popcount(remap_dec_ch_mask);

  1468.                     skip_bits_long(&s->gb, num_dec_ch * 5); // remap codes

  1469.                 }

  1470.             }

  1471. 

  1472.         } else {

  1473.             skip_bits(&s->gb, 3); // representation type

  1474.         }

  1475.     }

  1476. 

  1477.     drc_code_present = get_bits1(&s->gb);

  1478.     if (drc_code_present)

  1479.         get_bits(&s->gb, 8); // drc code

  1480. 

  1481.     if (get_bits1(&s->gb))

  1482.         skip_bits(&s->gb, 5); // dialog normalization code

  1483. 

  1484.     if (drc_code_present && embedded_stereo)

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

  1486. 

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

  1488.         skip_bits(&s->gb, 1); // external mix

  1489.         skip_bits(&s->gb, 6); // post mix gain code

  1490. 

  1491.         if (get_bits(&s->gb, 2) != 3) // mixer drc code

  1492.             skip_bits(&s->gb, 3); // drc limit

  1493.         else

  1494.             skip_bits(&s->gb, 8); // custom drc code

  1495. 

  1496.         if (get_bits1(&s->gb)) // channel specific scaling

  1497.             for (i = 0; i < s->num_mix_configs; i++)

  1498.                 skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6); // scale codes

  1499.         else

  1500.             skip_bits_long(&s->gb, s->num_mix_configs * 6); // scale codes

  1501. 

  1502.         for (i = 0; i < s->num_mix_configs; i++) {

  1503.             if (get_bits_left(&s->gb) < 0)

  1504.                 return -1;

  1505.             dca_exss_skip_mix_coeffs(&s->gb, channels, s->mix_config_num_ch[i]);

  1506.             if (embedded_6ch)

  1507.                 dca_exss_skip_mix_coeffs(&s->gb, 6, s->mix_config_num_ch[i]);

  1508.             if (embedded_stereo)

  1509.                 dca_exss_skip_mix_coeffs(&s->gb, 2, s->mix_config_num_ch[i]);

  1510.         }

  1511.     }

  1512. 

  1513.     switch (get_bits(&s->gb, 2)) {

  1514.     case 0: extensions_mask = get_bits(&s->gb, 12); break;

  1515.     case 1: extensions_mask = DCA_EXT_EXSS_XLL;     break;

  1516.     case 2: extensions_mask = DCA_EXT_EXSS_LBR;     break;

  1517.     case 3: extensions_mask = 0; /* aux coding */   break;

  1518.     }

  1519. 

  1520.     /* not parsed further, we were only interested in the extensions mask */

  1521. 

  1522.     if (get_bits_left(&s->gb) < 0)

  1523.         return -1;

  1524. 

  1525.     if (get_bits_count(&s->gb) - header_pos > header_size * 8) {

  1526.         av_log(s->avctx, AV_LOG_WARNING, "Asset header size mismatch.\n");

  1527.         return -1;

  1528.     }

  1529.     skip_bits_long(&s->gb, header_pos + header_size * 8 - get_bits_count(&s->gb));

  1530. 

  1531.     if (extensions_mask & DCA_EXT_EXSS_XLL)

  1532.         s->profile = FF_PROFILE_DTS_HD_MA;

  1533.     else if (extensions_mask & (DCA_EXT_EXSS_XBR | DCA_EXT_EXSS_X96 |

  1534.                                 DCA_EXT_EXSS_XXCH))

  1535.         s->profile = FF_PROFILE_DTS_HD_HRA;

  1536. 

  1537.     if (!(extensions_mask & DCA_EXT_CORE))

  1538.         av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n");

  1539.     if ((extensions_mask & DCA_CORE_EXTS) != s->core_ext_mask)

  1540.         av_log(s->avctx, AV_LOG_WARNING, "DTS extensions detection mismatch (%d, %d)\n",

  1541.                extensions_mask & DCA_CORE_EXTS, s->core_ext_mask);

  1542. 

  1543.     return 0;

  1544. }

  1545. 

  1546. /**

  1547.  * Parse extension substream header (HD)

  1548.  */

  1549. static void dca_exss_parse_header(DCAContext *s)

  1550. {

  1551.     int ss_index;

  1552.     int blownup;

  1553.     int num_audiop = 1;

  1554.     int num_assets = 1;

  1555.     int active_ss_mask[8];

  1556.     int i, j;

  1557. 

  1558.     if (get_bits_left(&s->gb) < 52)

  1559.         return;

  1560. 

  1561.     skip_bits(&s->gb, 8); // user data

  1562.     ss_index = get_bits(&s->gb, 2);

  1563. 

  1564.     blownup = get_bits1(&s->gb);

  1565.     skip_bits(&s->gb, 8 + 4 * blownup); // header_size

  1566.     skip_bits(&s->gb, 16 + 4 * blownup); // hd_size

  1567. 

  1568.     s->static_fields = get_bits1(&s->gb);

  1569.     if (s->static_fields) {

  1570.         skip_bits(&s->gb, 2); // reference clock code

  1571.         skip_bits(&s->gb, 3); // frame duration code

  1572. 

  1573.         if (get_bits1(&s->gb))

  1574.             skip_bits_long(&s->gb, 36); // timestamp

  1575. 

  1576.         /* a single stream can contain multiple audio assets that can be

  1577.          * combined to form multiple audio presentations */

  1578. 

  1579.         num_audiop = get_bits(&s->gb, 3) + 1;

  1580.         if (num_audiop > 1) {

  1581.             av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio presentations.");

  1582.             /* ignore such streams for now */

  1583.             return;

  1584.         }

  1585. 

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

  1587.         if (num_assets > 1) {

  1588.             av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio assets.");

  1589.             /* ignore such streams for now */

  1590.             return;

  1591.         }

  1592. 

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

  1594.             active_ss_mask[i] = get_bits(&s->gb, ss_index + 1);

  1595. 

  1596.         for (i = 0; i < num_audiop; i++)

  1597.             for (j = 0; j <= ss_index; j++)

  1598.                 if (active_ss_mask[i] & (1 << j))

  1599.                     skip_bits(&s->gb, 8); // active asset mask

  1600. 

  1601.         s->mix_metadata = get_bits1(&s->gb);

  1602.         if (s->mix_metadata) {

  1603.             int mix_out_mask_size;

  1604. 

  1605.             skip_bits(&s->gb, 2); // adjustment level

  1606.             mix_out_mask_size = (get_bits(&s->gb, 2) + 1) << 2;

  1607.             s->num_mix_configs = get_bits(&s->gb, 2) + 1;

  1608. 

  1609.             for (i = 0; i < s->num_mix_configs; i++) {

  1610.                 int mix_out_mask = get_bits(&s->gb, mix_out_mask_size);

  1611.                 s->mix_config_num_ch[i] = dca_exss_mask2count(mix_out_mask);

  1612.             }

  1613.         }

  1614.     }

  1615. 

  1616.     for (i = 0; i < num_assets; i++)

  1617.         skip_bits_long(&s->gb, 16 + 4 * blownup); // asset size

  1618. 

  1619.     for (i = 0; i < num_assets; i++) {

  1620.         if (dca_exss_parse_asset_header(s))

  1621.             return;

  1622.     }

  1623. 

  1624.     /* not parsed further, we were only interested in the extensions mask

  1625.      * from the asset header */

  1626. }

  1627. 

  1628. /**

  1629.  * Main frame decoding function

  1630.  * FIXME add arguments

  1631.  */

  1632. static int dca_decode_frame(AVCodecContext * avctx,

  1633.                             void *data, int *data_size,

  1634.                             AVPacket *avpkt)

  1635. {

  1636.     const uint8_t *buf = avpkt->data;

  1637.     int buf_size = avpkt->size;

  1638. 

  1639.     int lfe_samples;

  1640.     int num_core_channels = 0;

  1641.     int i, ret;

  1642.     float   *samples_flt = data;

  1643.     int16_t *samples_s16 = data;

  1644.     int out_size;

  1645.     DCAContext *s = avctx->priv_data;

  1646.     int channels;

  1647.     int core_ss_end;

  1648. 

  1649. 

  1650.     s->xch_present = 0;

  1651. 

  1652.     s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer,

  1653.                                                DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);

  1654.     if (s->dca_buffer_size == AVERROR_INVALIDDATA) {

  1655.         av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");

  1656.         return AVERROR_INVALIDDATA;

  1657.     }

  1658. 

  1659.     init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);

  1660.     if ((ret = dca_parse_frame_header(s)) < 0) {

  1661.         //seems like the frame is corrupt, try with the next one

  1662.         return ret;

  1663.     }

  1664.     //set AVCodec values with parsed data

  1665.     avctx->sample_rate = s->sample_rate;

  1666.     avctx->bit_rate = s->bit_rate;

  1667.     avctx->frame_size = s->sample_blocks * 32;

  1668. 

  1669.     s->profile = FF_PROFILE_DTS;

  1670. 

  1671.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1672.         if ((ret = dca_decode_block(s, 0, i))) {

  1673.             av_log(avctx, AV_LOG_ERROR, "error decoding block\n");

  1674.             return ret;

  1675.         }

  1676.     }

  1677. 

  1678.     /* record number of core channels incase less than max channels are requested */

  1679.     num_core_channels = s->prim_channels;

  1680. 

  1681.     if (s->ext_coding)

  1682.         s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];

  1683.     else

  1684.         s->core_ext_mask = 0;

  1685. 

  1686.     core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;

  1687. 

  1688.     /* only scan for extensions if ext_descr was unknown or indicated a

  1689.      * supported XCh extension */

  1690.     if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {

  1691. 

  1692.         /* if ext_descr was unknown, clear s->core_ext_mask so that the

  1693.          * extensions scan can fill it up */

  1694.         s->core_ext_mask = FFMAX(s->core_ext_mask, 0);

  1695. 

  1696.         /* extensions start at 32-bit boundaries into bitstream */

  1697.         skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1698. 

  1699.     while(core_ss_end - get_bits_count(&s->gb) >= 32) {

  1700.         uint32_t bits = get_bits_long(&s->gb, 32);

  1701. 

  1702.         switch(bits) {

  1703.         case 0x5a5a5a5a: {

  1704.             int ext_amode, xch_fsize;

  1705. 

  1706.             s->xch_base_channel = s->prim_channels;

  1707. 

  1708.             /* validate sync word using XCHFSIZE field */

  1709.             xch_fsize = show_bits(&s->gb, 10);

  1710.             if((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&

  1711.                (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))

  1712.                 continue;

  1713. 

  1714.             /* skip length-to-end-of-frame field for the moment */

  1715.             skip_bits(&s->gb, 10);

  1716. 

  1717.             s->core_ext_mask |= DCA_EXT_XCH;

  1718. 

  1719.             /* extension amode should == 1, number of channels in extension */

  1720.             /* AFAIK XCh is not used for more channels */

  1721.             if ((ext_amode = get_bits(&s->gb, 4)) != 1) {

  1722.                 av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"

  1723.                        " supported!\n",ext_amode);

  1724.                 continue;

  1725.             }

  1726. 

  1727.             /* much like core primary audio coding header */

  1728.             dca_parse_audio_coding_header(s, s->xch_base_channel);

  1729. 

  1730.             for (i = 0; i < (s->sample_blocks / 8); i++) {

  1731.                 if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {

  1732.                     av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");

  1733.                     continue;

  1734.                 }

  1735.             }

  1736. 

  1737.             s->xch_present = 1;

  1738.             break;

  1739.         }

  1740.         case 0x47004a03:

  1741.             /* XXCh: extended channels */

  1742.             /* usually found either in core or HD part in DTS-HD HRA streams,

  1743.              * but not in DTS-ES which contains XCh extensions instead */

  1744.             s->core_ext_mask |= DCA_EXT_XXCH;

  1745.             break;

  1746. 

  1747.         case 0x1d95f262: {

  1748.             int fsize96 = show_bits(&s->gb, 12) + 1;

  1749.             if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)

  1750.                 continue;

  1751. 

  1752.             av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n", get_bits_count(&s->gb));

  1753.             skip_bits(&s->gb, 12);

  1754.             av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);

  1755.             av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));

  1756. 

  1757.             s->core_ext_mask |= DCA_EXT_X96;

  1758.             break;

  1759.         }

  1760.         }

  1761. 

  1762.         skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1763.     }

  1764. 

  1765.     } else {

  1766.         /* no supported extensions, skip the rest of the core substream */

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

  1768.     }

  1769. 

  1770.     if (s->core_ext_mask & DCA_EXT_X96)

  1771.         s->profile = FF_PROFILE_DTS_96_24;

  1772.     else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))

  1773.         s->profile = FF_PROFILE_DTS_ES;

  1774. 

  1775.     /* check for ExSS (HD part) */

  1776.     if (s->dca_buffer_size - s->frame_size > 32

  1777.         && get_bits_long(&s->gb, 32) == DCA_HD_MARKER)

  1778.         dca_exss_parse_header(s);

  1779. 

  1780.     avctx->profile = s->profile;

  1781. 

  1782.     channels = s->prim_channels + !!s->lfe;

  1783. 

  1784.     if (s->amode<16) {

  1785.         avctx->channel_layout = dca_core_channel_layout[s->amode];

  1786. 

  1787.         if (s->xch_present && (!avctx->request_channels ||

  1788.                                avctx->request_channels > num_core_channels + !!s->lfe)) {

  1789.             avctx->channel_layout |= AV_CH_BACK_CENTER;

  1790.             if (s->lfe) {

  1791.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1792.                 s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode];

  1793.             } else {

  1794.                 s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];

  1795.             }

  1796.         } else {

  1797.             channels = num_core_channels + !!s->lfe;

  1798.             s->xch_present = 0; /* disable further xch processing */

  1799.             if (s->lfe) {

  1800.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1801.                 s->channel_order_tab = dca_channel_reorder_lfe[s->amode];

  1802.             } else

  1803.                 s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];

  1804.         }

  1805. 

  1806.         if (channels > !!s->lfe &&

  1807.             s->channel_order_tab[channels - 1 - !!s->lfe] < 0)

  1808.             return AVERROR_INVALIDDATA;

  1809. 

  1810.         if (avctx->request_channels == 2 && s->prim_channels > 2) {

  1811.             channels = 2;

  1812.             s->output = DCA_STEREO;

  1813.             avctx->channel_layout = AV_CH_LAYOUT_STEREO;

  1814.         }

  1815.         else if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) {

  1816.             static const int8_t dca_channel_order_native[9] = { 0, 1, 2, 3, 4, 5, 6, 7, 8 };

  1817.             s->channel_order_tab = dca_channel_order_native;

  1818.         }

  1819.     } else {

  1820.         av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n",s->amode);

  1821.         return AVERROR_INVALIDDATA;

  1822.     }

  1823. 

  1824.     if (avctx->channels != channels) {

  1825.         if (avctx->channels)

  1826.             av_log(avctx, AV_LOG_INFO, "Number of channels changed in DCA decoder (%d -> %d)\n", avctx->channels, channels);

  1827.         avctx->channels = channels;

  1828.     }

  1829. 

  1830.     out_size = 256 / 8 * s->sample_blocks * channels *

  1831.                av_get_bytes_per_sample(avctx->sample_fmt);

  1832.     if (*data_size < out_size)

  1833.         return AVERROR(EINVAL);

  1834.     *data_size = out_size;

  1835. 

  1836.     /* filter to get final output */

  1837.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1838.         dca_filter_channels(s, i);

  1839. 

  1840.         /* If this was marked as a DTS-ES stream we need to subtract back- */

  1841.         /* channel from SL & SR to remove matrixed back-channel signal */

  1842.         if((s->source_pcm_res & 1) && s->xch_present) {

  1843.             float* back_chan = s->samples + s->channel_order_tab[s->xch_base_channel] * 256;

  1844.             float* lt_chan   = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256;

  1845.             float* rt_chan   = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;

  1846.             s->dsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);

  1847.             s->dsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);

  1848.         }

  1849. 

  1850.         if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {

  1851.             s->fmt_conv.float_interleave(samples_flt, s->samples_chanptr, 256,

  1852.                                          channels);

  1853.             samples_flt += 256 * channels;

  1854.         } else {

  1855.             s->fmt_conv.float_to_int16_interleave(samples_s16,

  1856.                                                   s->samples_chanptr, 256,

  1857.                                                   channels);

  1858.             samples_s16 += 256 * channels;

  1859.         }

  1860.     }

  1861. 

  1862.     /* update lfe history */

  1863.     lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);

  1864.     for (i = 0; i < 2 * s->lfe * 4; i++) {

  1865.         s->lfe_data[i] = s->lfe_data[i + lfe_samples];

  1866.     }

  1867. 

  1868.     return buf_size;

  1869. }

  1870. 

  1871. 

  1872. 

  1873. /**

  1874.  * DCA initialization

  1875.  *

  1876.  * @param avctx     pointer to the AVCodecContext

  1877.  */

  1878. 

  1879. static av_cold int dca_decode_init(AVCodecContext * avctx)

  1880. {

  1881.     DCAContext *s = avctx->priv_data;

  1882.     int i;

  1883. 

  1884.     s->avctx = avctx;

  1885.     dca_init_vlcs();

  1886. 

  1887.     dsputil_init(&s->dsp, avctx);

  1888.     ff_mdct_init(&s->imdct, 6, 1, 1.0);

  1889.     ff_synth_filter_init(&s->synth);

  1890.     ff_dcadsp_init(&s->dcadsp);

  1891.     ff_fmt_convert_init(&s->fmt_conv, avctx);

  1892. 

  1893.     for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)

  1894.         s->samples_chanptr[i] = s->samples + i * 256;

  1895. 

  1896.     if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {

  1897.         avctx->sample_fmt = AV_SAMPLE_FMT_FLT;

  1898.         s->scale_bias = 1.0 / 32768.0;

  1899.     } else {

  1900.         avctx->sample_fmt = AV_SAMPLE_FMT_S16;

  1901.         s->scale_bias = 1.0;

  1902.     }

  1903. 

  1904.     /* allow downmixing to stereo */

  1905.     if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&

  1906.         avctx->request_channels == 2) {

  1907.         avctx->channels = avctx->request_channels;

  1908.     }

  1909. 

  1910.     return 0;

  1911. }

  1912. 

  1913. static av_cold int dca_decode_end(AVCodecContext * avctx)

  1914. {

  1915.     DCAContext *s = avctx->priv_data;

  1916.     ff_mdct_end(&s->imdct);

  1917.     return 0;

  1918. }

  1919. 

  1920. static const AVProfile profiles[] = {

  1921.     { FF_PROFILE_DTS,        "DTS"        },

  1922.     { FF_PROFILE_DTS_ES,     "DTS-ES"     },

  1923.     { FF_PROFILE_DTS_96_24,  "DTS 96/24"  },

  1924.     { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },

  1925.     { FF_PROFILE_DTS_HD_MA,  "DTS-HD MA"  },

  1926.     { FF_PROFILE_UNKNOWN },

  1927. };

  1928. 

  1929. AVCodec ff_dca_decoder = {

  1930.     .name = "dca",

  1931.     .type = AVMEDIA_TYPE_AUDIO,

  1932.     .id = CODEC_ID_DTS,

  1933.     .priv_data_size = sizeof(DCAContext),

  1934.     .init = dca_decode_init,

  1935.     .decode = dca_decode_frame,

  1936.     .close = dca_decode_end,

  1937.     .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),

  1938.     .capabilities = CODEC_CAP_CHANNEL_CONF,

  1939.     .sample_fmts = (const enum AVSampleFormat[]) {

  1940.         AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE

  1941.     },

  1942.     .profiles = NULL_IF_CONFIG_SMALL(profiles),

  1943. };