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

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

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

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

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

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

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

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

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

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

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

  307. 

  308.     /* Primary audio coding side information */

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

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

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

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

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

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

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

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

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

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

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

  320. 

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

  322. 

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

  324.     int lfe_scale_factor;

  325. 

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

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

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

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

  330.     int hist_index[DCA_PRIM_CHANNELS_MAX];

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

  332. 

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

  334.     float scale_bias;           ///< output scale

  335. 

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

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

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

  339. 

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

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

  342. 

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

  344.     GetBitContext gb;

  345.     /* Current position in DCA frame */

  346.     int current_subframe;

  347.     int current_subsubframe;

  348. 

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

  350. 

  351.     /* XCh extension information */

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

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

  354. 

  355.     /* ExSS header parser */

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

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

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

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

  360. 

  361.     int profile;

  362. 

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

  364.     DSPContext dsp;

  365.     FFTContext imdct;

  366.     SynthFilterContext synth;

  367.     DCADSPContext dcadsp;

  368.     FmtConvertContext fmt_conv;

  369. } DCAContext;

  370. 

  371. static const uint16_t dca_vlc_offs[] = {

  372.         0,   512,   640,   768,  1282,  1794,  2436,  3080,  3770,  4454,  5364,

  373.      5372,  5380,  5388,  5392,  5396,  5412,  5420,  5428,  5460,  5492,  5508,

  374.      5572,  5604,  5668,  5796,  5860,  5892,  6412,  6668,  6796,  7308,  7564,

  375.      7820,  8076,  8620,  9132,  9388,  9910, 10166, 10680, 11196, 11726, 12240,

  376.     12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264,

  377.     18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622,

  378. };

  379. 

  380. static av_cold void dca_init_vlcs(void)

  381. {

  382.     static int vlcs_initialized = 0;

  383.     int i, j, c = 14;

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

  385. 

  386.     if (vlcs_initialized)

  387.         return;

  388. 

  389.     dca_bitalloc_index.offset = 1;

  390.     dca_bitalloc_index.wrap = 2;

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

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

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

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

  395.                  bitalloc_12_bits[i], 1, 1,

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

  397.     }

  398.     dca_scalefactor.offset = -64;

  399.     dca_scalefactor.wrap = 2;

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

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

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

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

  404.                  scales_bits[i], 1, 1,

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

  406.     }

  407.     dca_tmode.offset = 0;

  408.     dca_tmode.wrap = 1;

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

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

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

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

  413.                  tmode_bits[i], 1, 1,

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

  415.     }

  416. 

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

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

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

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

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

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

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

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

  425.                      bitalloc_sizes[i],

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

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

  428.             c++;

  429.         }

  430.     vlcs_initialized = 1;

  431. }

  432. 

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

  434. {

  435.     while(len--)

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

  437. }

  438. 

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

  440. {

  441.     int i, j;

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

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

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

  445. 

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

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

  448. 

  449.     if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)

  450.         s->prim_channels = DCA_PRIM_CHANNELS_MAX;

  451. 

  452. 

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

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

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

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

  457.     }

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

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

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

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

  462.     }

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

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

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

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

  467. 

  468.     /* Get codebooks quantization indexes */

  469.     if (!base_channel)

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

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

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

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

  474. 

  475.     /* Get scale factor adjustment */

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

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

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

  479. 

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

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

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

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

  484. 

  485.     if (s->crc_present) {

  486.         /* Audio header CRC check */

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

  488.     }

  489. 

  490.     s->current_subframe = 0;

  491.     s->current_subsubframe = 0;

  492. 

  493. #ifdef TRACE

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

  512.     }

  513. #endif

  514. 

  515.   return 0;

  516. }

  517. 

  518. static int dca_parse_frame_header(DCAContext * s)

  519. {

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

  521. 

  522.     /* Sync code */

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

  524. 

  525.     /* Frame header */

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

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

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

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

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

  531.     if (s->frame_size < 95)

  532.         return -1;

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

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

  535.     if (!s->sample_rate)

  536.         return -1;

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

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

  539.     if (!s->bit_rate)

  540.         return -1;

  541. 

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

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

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

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

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

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

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

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

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

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

  552. 

  553.     /* TODO: check CRC */

  554.     if (s->crc_present)

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

  556. 

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

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

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

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

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

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

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

  564. 

  565.     /* FIXME: channels mixing levels */

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

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

  568. 

  569. #ifdef TRACE

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

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

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

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

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

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

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

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

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

  579.            s->sample_rate);

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

  581.            s->bit_rate);

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

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

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

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

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

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

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

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

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

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

  592.            s->predictor_history);

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

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

  595.            s->multirate_inter);

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

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

  598.     av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

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

  605. #endif

  606. 

  607.     /* Primary audio coding header */

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

  609. 

  610.     return dca_parse_audio_coding_header(s, 0);

  611. }

  612. 

  613. 

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

  615. {

  616.    if (level < 5) {

  617.        /* huffman encoded */

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

  619.    } else if (level < 8)

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

  621.    return value;

  622. }

  623. 

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

  625. {

  626.     /* Primary audio coding side information */

  627.     int j, k;

  628. 

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

  630.         return -1;

  631. 

  632.     if (!base_channel) {

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

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

  635.     }

  636. 

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

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

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

  640.     }

  641. 

  642.     /* Get prediction codebook */

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

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

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

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

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

  648.             }

  649.         }

  650.     }

  651. 

  652.     /* Bit allocation index */

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

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

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

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

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

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

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

  660.                 av_log(s->avctx, AV_LOG_ERROR,

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

  662.                 return -1;

  663.             } else {

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

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

  666.             }

  667. 

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

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

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

  671.                 return -1;

  672.             }

  673.         }

  674.     }

  675. 

  676.     /* Transition mode */

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

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

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

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

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

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

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

  684.             }

  685.         }

  686.     }

  687. 

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

  689.         return -1;

  690. 

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

  692.         const uint32_t *scale_table;

  693.         int scale_sum;

  694. 

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

  696. 

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

  698.             scale_table = scale_factor_quant7;

  699.         else

  700.             scale_table = scale_factor_quant6;

  701. 

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

  703.         scale_sum = 0;

  704. 

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

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

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

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

  709.             }

  710. 

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

  712.                 /* Get second scale factor */

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

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

  715.             }

  716.         }

  717.     }

  718. 

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

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

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

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

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

  724.     }

  725. 

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

  727.         return -1;

  728. 

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

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

  731.         int source_channel;

  732. 

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

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

  735.             int scale = 0;

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

  737. 

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

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

  740. 

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

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

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

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

  745.             }

  746. 

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

  748.                 av_log(s->avctx, AV_LOG_DEBUG,

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

  750.                 s->debug_flag |= 0x02;

  751.             }

  752.         }

  753.     }

  754. 

  755.     /* Stereo downmix coefficients */

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

  757.         if (s->downmix) {

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

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

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

  761.             }

  762.         } else {

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

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

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

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

  767.             }

  768.         }

  769.     }

  770. 

  771.     /* Dynamic range coefficient */

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

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

  774. 

  775.     /* Side information CRC check word */

  776.     if (s->crc_present) {

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

  778.     }

  779. 

  780.     /*

  781.      * Primary audio data arrays

  782.      */

  783. 

  784.     /* VQ encoded high frequency subbands */

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

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

  787.             /* 1 vector -> 32 samples */

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

  789. 

  790.     /* Low frequency effect data */

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

  792.         /* LFE samples */

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

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

  795.         float lfe_scale;

  796. 

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

  798.             /* Signed 8 bits int */

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

  800.         }

  801. 

  802.         /* Scale factor index */

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

  804. 

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

  806.         lfe_scale = 0.035 * s->lfe_scale_factor;

  807. 

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

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

  810.     }

  811. 

  812. #ifdef TRACE

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

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

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

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

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

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

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

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

  821.     }

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

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

  824.                 av_log(s->avctx, AV_LOG_DEBUG,

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

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

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

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

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

  830.     }

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

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

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

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

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

  836.     }

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

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

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

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

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

  842.     }

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

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

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

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

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

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

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

  850.         }

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

  852.     }

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

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

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

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

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

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

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

  860.         }

  861.     }

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

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

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

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

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

  867.         }

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

  869.     }

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

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

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

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

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

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

  876. 

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

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

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

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

  881.     }

  882. #endif

  883. 

  884.     return 0;

  885. }

  886. 

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

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

  889.                             float scale)

  890. {

  891.     const float *prCoeff;

  892.     int i;

  893. 

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

  895.     int subindex;

  896. 

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

  898. 

  899.     /* Select filter */

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

  901.         prCoeff = fir_32bands_nonperfect;

  902.     else                        /* Perfect reconstruction */

  903.         prCoeff = fir_32bands_perfect;

  904. 

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

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

  907. 

  908.     /* Reconstructed channel sample index */

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

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

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

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

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

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

  915.         }

  916. 

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

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

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

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

  921.         samples_out+= 32;

  922. 

  923.     }

  924. }

  925. 

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

  927.                                   int num_deci_sample, float *samples_in,

  928.                                   float *samples_out, float scale)

  929. {

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

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

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

  933.      *   from last subframe as history.

  934.      *

  935.      * samples_out: An array holding interpolated samples

  936.      */

  937. 

  938.     int decifactor;

  939.     const float *prCoeff;

  940.     int deciindex;

  941. 

  942.     /* Select decimation filter */

  943.     if (decimation_select == 1) {

  944.         decifactor = 64;

  945.         prCoeff = lfe_fir_128;

  946.     } else {

  947.         decifactor = 32;

  948.         prCoeff = lfe_fir_64;

  949.     }

  950.     /* Interpolation */

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

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

  953.                           scale);

  954.         samples_in++;

  955.         samples_out += 2 * decifactor;

  956.     }

  957. }

  958. 

  959. /* downmixing routines */

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

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

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

  963. 

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

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

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

  967. 

  968. #define MIX_FRONT3(samples, coef) \

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

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

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

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

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

  974. 

  975. #define DOWNMIX_TO_STEREO(op1, op2) \

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

  977.         op1 \

  978.         op2 \

  979.     }

  980. 

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

  982.                         int downmix_coef[DCA_PRIM_CHANNELS_MAX][2],

  983.                         const int8_t *channel_mapping)

  984. {

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

  986.     int i;

  987.     float t, u, v;

  988.     float coef[DCA_PRIM_CHANNELS_MAX][2];

  989. 

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

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

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

  993.     }

  994. 

  995.     switch (srcfmt) {

  996.     case DCA_MONO:

  997.     case DCA_CHANNEL:

  998.     case DCA_STEREO_TOTAL:

  999.     case DCA_STEREO_SUMDIFF:

  1000.     case DCA_4F2R:

  1001.         av_log(NULL, 0, "Not implemented!\n");

  1002.         break;

  1003.     case DCA_STEREO:

  1004.         break;

  1005.     case DCA_3F:

  1006.         c = channel_mapping[0] * 256;

  1007.         l = channel_mapping[1] * 256;

  1008.         r = channel_mapping[2] * 256;

  1009.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),);

  1010.         break;

  1011.     case DCA_2F1R:

  1012.         s = channel_mapping[2] * 256;

  1013.         DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + s, 2, coef),);

  1014.         break;

  1015.     case DCA_3F1R:

  1016.         c = channel_mapping[0] * 256;

  1017.         l = channel_mapping[1] * 256;

  1018.         r = channel_mapping[2] * 256;

  1019.         s = channel_mapping[3] * 256;

  1020.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1021.                           MIX_REAR1(samples, i + s, 3, coef));

  1022.         break;

  1023.     case DCA_2F2R:

  1024.         sl = channel_mapping[2] * 256;

  1025.         sr = channel_mapping[3] * 256;

  1026.         DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + sl, i + sr, 2, coef),);

  1027.         break;

  1028.     case DCA_3F2R:

  1029.         c =  channel_mapping[0] * 256;

  1030.         l =  channel_mapping[1] * 256;

  1031.         r =  channel_mapping[2] * 256;

  1032.         sl = channel_mapping[3] * 256;

  1033.         sr = channel_mapping[4] * 256;

  1034.         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

  1035.                           MIX_REAR2(samples, i + sl, i + sr, 3, coef));

  1036.         break;

  1037.     }

  1038. }

  1039. 

  1040. 

  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 -1;

  1059.     }

  1060. }

  1061. 

  1062. static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };

  1063. static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };

  1064. 

  1065. #ifndef int8x8_fmul_int32

  1066. static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)

  1067. {

  1068.     float fscale = scale / 16.0;

  1069.     int i;

  1070.     for (i = 0; i < 8; i++)

  1071.         dst[i] = src[i] * fscale;

  1072. }

  1073. #endif

  1074. 

  1075. static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)

  1076. {

  1077.     int k, l;

  1078.     int subsubframe = s->current_subsubframe;

  1079. 

  1080.     const float *quant_step_table;

  1081. 

  1082.     /* FIXME */

  1083.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1084.     LOCAL_ALIGNED_16(int, block, [8]);

  1085. 

  1086.     /*

  1087.      * Audio data

  1088.      */

  1089. 

  1090.     /* Select quantization step size table */

  1091.     if (s->bit_rate_index == 0x1f)

  1092.         quant_step_table = lossless_quant_d;

  1093.     else

  1094.         quant_step_table = lossy_quant_d;

  1095. 

  1096.     for (k = base_channel; k < s->prim_channels; k++) {

  1097.         if (get_bits_left(&s->gb) < 0)

  1098.             return -1;

  1099. 

  1100.         for (l = 0; l < s->vq_start_subband[k]; l++) {

  1101.             int m;

  1102. 

  1103.             /* Select the mid-tread linear quantizer */

  1104.             int abits = s->bitalloc[k][l];

  1105. 

  1106.             float quant_step_size = quant_step_table[abits];

  1107. 

  1108.             /*

  1109.              * Determine quantization index code book and its type

  1110.              */

  1111. 

  1112.             /* Select quantization index code book */

  1113.             int sel = s->quant_index_huffman[k][abits];

  1114. 

  1115.             /*

  1116.              * Extract bits from the bit stream

  1117.              */

  1118.             if (!abits){

  1119.                 memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));

  1120.             } else {

  1121.                 /* Deal with transients */

  1122.                 int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];

  1123.                 float rscale = quant_step_size * s->scale_factor[k][l][sfi] * s->scalefactor_adj[k][sel];

  1124. 

  1125.                 if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){

  1126.                     if (abits <= 7){

  1127.                         /* Block code */

  1128.                         int block_code1, block_code2, size, levels;

  1129. 

  1130.                         size = abits_sizes[abits-1];

  1131.                         levels = abits_levels[abits-1];

  1132. 

  1133.                         block_code1 = get_bits(&s->gb, size);

  1134.                         /* FIXME Should test return value */

  1135.                         decode_blockcode(block_code1, levels, block);

  1136.                         block_code2 = get_bits(&s->gb, size);

  1137.                         decode_blockcode(block_code2, levels, &block[4]);

  1138.                     }else{

  1139.                         /* no coding */

  1140.                         for (m = 0; m < 8; m++)

  1141.                             block[m] = get_sbits(&s->gb, abits - 3);

  1142.                     }

  1143.                 }else{

  1144.                     /* Huffman coded */

  1145.                     for (m = 0; m < 8; m++)

  1146.                         block[m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);

  1147.                 }

  1148. 

  1149.                 s->fmt_conv.int32_to_float_fmul_scalar(subband_samples[k][l],

  1150.                                                   block, rscale, 8);

  1151.             }

  1152. 

  1153.             /*

  1154.              * Inverse ADPCM if in prediction mode

  1155.              */

  1156.             if (s->prediction_mode[k][l]) {

  1157.                 int n;

  1158.                 for (m = 0; m < 8; m++) {

  1159.                     for (n = 1; n <= 4; n++)

  1160.                         if (m >= n)

  1161.                             subband_samples[k][l][m] +=

  1162.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1163.                                  subband_samples[k][l][m - n] / 8192);

  1164.                         else if (s->predictor_history)

  1165.                             subband_samples[k][l][m] +=

  1166.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1167.                                  s->subband_samples_hist[k][l][m - n +

  1168.                                                                4] / 8192);

  1169.                 }

  1170.             }

  1171.         }

  1172. 

  1173.         /*

  1174.          * Decode VQ encoded high frequencies

  1175.          */

  1176.         for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {

  1177.             /* 1 vector -> 32 samples but we only need the 8 samples

  1178.              * for this subsubframe. */

  1179.             int hfvq = s->high_freq_vq[k][l];

  1180. 

  1181.             if (!s->debug_flag & 0x01) {

  1182.                 av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n");

  1183.                 s->debug_flag |= 0x01;

  1184.             }

  1185. 

  1186.             int8x8_fmul_int32(subband_samples[k][l],

  1187.                               &high_freq_vq[hfvq][subsubframe * 8],

  1188.                               s->scale_factor[k][l][0]);

  1189.         }

  1190.     }

  1191. 

  1192.     /* Check for DSYNC after subsubframe */

  1193.     if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {

  1194.         if (0xFFFF == get_bits(&s->gb, 16)) {   /* 0xFFFF */

  1195. #ifdef TRACE

  1196.             av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");

  1197. #endif

  1198.         } else {

  1199.             av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");

  1200.         }

  1201.     }

  1202. 

  1203.     /* Backup predictor history for adpcm */

  1204.     for (k = base_channel; k < s->prim_channels; k++)

  1205.         for (l = 0; l < s->vq_start_subband[k]; l++)

  1206.             memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4],

  1207.                         4 * sizeof(subband_samples[0][0][0]));

  1208. 

  1209.     return 0;

  1210. }

  1211. 

  1212. static int dca_filter_channels(DCAContext * s, int block_index)

  1213. {

  1214.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1215.     int k;

  1216. 

  1217.     /* 32 subbands QMF */

  1218.     for (k = 0; k < s->prim_channels; k++) {

  1219. /*        static float pcm_to_double[8] =

  1220.             {32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/

  1221.          qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * s->channel_order_tab[k]],

  1222.                          M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ );

  1223.     }

  1224. 

  1225.     /* Down mixing */

  1226.     if (s->avctx->request_channels == 2 && s->prim_channels > 2) {

  1227.         dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab);

  1228.     }

  1229. 

  1230.     /* Generate LFE samples for this subsubframe FIXME!!! */

  1231.     if (s->output & DCA_LFE) {

  1232.         lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,

  1233.                               s->lfe_data + 2 * s->lfe * (block_index + 4),

  1234.                               &s->samples[256 * dca_lfe_index[s->amode]],

  1235.                               (1.0/256.0)*s->scale_bias);

  1236.         /* Outputs 20bits pcm samples */

  1237.     }

  1238. 

  1239.     return 0;

  1240. }

  1241. 

  1242. 

  1243. static int dca_subframe_footer(DCAContext * s, int base_channel)

  1244. {

  1245.     int aux_data_count = 0, i;

  1246. 

  1247.     /*

  1248.      * Unpack optional information

  1249.      */

  1250. 

  1251.     /* presumably optional information only appears in the core? */

  1252.     if (!base_channel) {

  1253.         if (s->timestamp)

  1254.             get_bits(&s->gb, 32);

  1255. 

  1256.         if (s->aux_data)

  1257.             aux_data_count = get_bits(&s->gb, 6);

  1258. 

  1259.         for (i = 0; i < aux_data_count; i++)

  1260.             get_bits(&s->gb, 8);

  1261. 

  1262.         if (s->crc_present && (s->downmix || s->dynrange))

  1263.             get_bits(&s->gb, 16);

  1264.     }

  1265. 

  1266.     return 0;

  1267. }

  1268. 

  1269. /**

  1270.  * Decode a dca frame block

  1271.  *

  1272.  * @param s     pointer to the DCAContext

  1273.  */

  1274. 

  1275. static int dca_decode_block(DCAContext * s, int base_channel, int block_index)

  1276. {

  1277. 

  1278.     /* Sanity check */

  1279.     if (s->current_subframe >= s->subframes) {

  1280.         av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",

  1281.                s->current_subframe, s->subframes);

  1282.         return -1;

  1283.     }

  1284. 

  1285.     if (!s->current_subsubframe) {

  1286. #ifdef TRACE

  1287.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");

  1288. #endif

  1289.         /* Read subframe header */

  1290.         if (dca_subframe_header(s, base_channel, block_index))

  1291.             return -1;

  1292.     }

  1293. 

  1294.     /* Read subsubframe */

  1295. #ifdef TRACE

  1296.     av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");

  1297. #endif

  1298.     if (dca_subsubframe(s, base_channel, block_index))

  1299.         return -1;

  1300. 

  1301.     /* Update state */

  1302.     s->current_subsubframe++;

  1303.     if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {

  1304.         s->current_subsubframe = 0;

  1305.         s->current_subframe++;

  1306.     }

  1307.     if (s->current_subframe >= s->subframes) {

  1308. #ifdef TRACE

  1309.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");

  1310. #endif

  1311.         /* Read subframe footer */

  1312.         if (dca_subframe_footer(s, base_channel))

  1313.             return -1;

  1314.     }

  1315. 

  1316.     return 0;

  1317. }

  1318. 

  1319. /**

  1320.  * Convert bitstream to one representation based on sync marker

  1321.  */

  1322. static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * dst,

  1323.                           int max_size)

  1324. {

  1325.     uint32_t mrk;

  1326.     int i, tmp;

  1327.     const uint16_t *ssrc = (const uint16_t *) src;

  1328.     uint16_t *sdst = (uint16_t *) dst;

  1329.     PutBitContext pb;

  1330. 

  1331.     if ((unsigned)src_size > (unsigned)max_size) {

  1332. //        av_log(NULL, AV_LOG_ERROR, "Input frame size larger than DCA_MAX_FRAME_SIZE!\n");

  1333. //        return -1;

  1334.         src_size = max_size;

  1335.     }

  1336. 

  1337.     mrk = AV_RB32(src);

  1338.     switch (mrk) {

  1339.     case DCA_MARKER_RAW_BE:

  1340.         memcpy(dst, src, src_size);

  1341.         return src_size;

  1342.     case DCA_MARKER_RAW_LE:

  1343.         for (i = 0; i < (src_size + 1) >> 1; i++)

  1344.             *sdst++ = av_bswap16(*ssrc++);

  1345.         return src_size;

  1346.     case DCA_MARKER_14B_BE:

  1347.     case DCA_MARKER_14B_LE:

  1348.         init_put_bits(&pb, dst, max_size);

  1349.         for (i = 0; i < (src_size + 1) >> 1; i++, src += 2) {

  1350.             tmp = ((mrk == DCA_MARKER_14B_BE) ? AV_RB16(src) : AV_RL16(src)) & 0x3FFF;

  1351.             put_bits(&pb, 14, tmp);

  1352.         }

  1353.         flush_put_bits(&pb);

  1354.         return (put_bits_count(&pb) + 7) >> 3;

  1355.     default:

  1356.         return -1;

  1357.     }

  1358. }

  1359. 

  1360. /**

  1361.  * Return the number of channels in an ExSS speaker mask (HD)

  1362.  */

  1363. static int dca_exss_mask2count(int mask)

  1364. {

  1365.     /* count bits that mean speaker pairs twice */

  1366.     return av_popcount(mask)

  1367.         + av_popcount(mask & (

  1368.             DCA_EXSS_CENTER_LEFT_RIGHT

  1369.           | DCA_EXSS_FRONT_LEFT_RIGHT

  1370.           | DCA_EXSS_FRONT_HIGH_LEFT_RIGHT

  1371.           | DCA_EXSS_WIDE_LEFT_RIGHT

  1372.           | DCA_EXSS_SIDE_LEFT_RIGHT

  1373.           | DCA_EXSS_SIDE_HIGH_LEFT_RIGHT

  1374.           | DCA_EXSS_SIDE_REAR_LEFT_RIGHT

  1375.           | DCA_EXSS_REAR_LEFT_RIGHT

  1376.           | DCA_EXSS_REAR_HIGH_LEFT_RIGHT

  1377.           ));

  1378. }

  1379. 

  1380. /**

  1381.  * Skip mixing coefficients of a single mix out configuration (HD)

  1382.  */

  1383. static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch)

  1384. {

  1385.     int i;

  1386. 

  1387.     for (i = 0; i < channels; i++) {

  1388.         int mix_map_mask = get_bits(gb, out_ch);

  1389.         int num_coeffs = av_popcount(mix_map_mask);

  1390.         skip_bits_long(gb, num_coeffs * 6);

  1391.     }

  1392. }

  1393. 

  1394. /**

  1395.  * Parse extension substream asset header (HD)

  1396.  */

  1397. static int dca_exss_parse_asset_header(DCAContext *s)

  1398. {

  1399.     int header_pos = get_bits_count(&s->gb);

  1400.     int header_size;

  1401.     int channels;

  1402.     int embedded_stereo = 0;

  1403.     int embedded_6ch = 0;

  1404.     int drc_code_present;

  1405.     int extensions_mask;

  1406.     int i, j;

  1407. 

  1408.     if (get_bits_left(&s->gb) < 16)

  1409.         return -1;

  1410. 

  1411.     /* We will parse just enough to get to the extensions bitmask with which

  1412.      * we can set the profile value. */

  1413. 

  1414.     header_size = get_bits(&s->gb, 9) + 1;

  1415.     skip_bits(&s->gb, 3); // asset index

  1416. 

  1417.     if (s->static_fields) {

  1418.         if (get_bits1(&s->gb))

  1419.             skip_bits(&s->gb, 4); // asset type descriptor

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

  1421.             skip_bits_long(&s->gb, 24); // language descriptor

  1422. 

  1423.         if (get_bits1(&s->gb)) {

  1424.             /* How can one fit 1024 bytes of text here if the maximum value

  1425.              * for the asset header size field above was 512 bytes? */

  1426.             int text_length = get_bits(&s->gb, 10) + 1;

  1427.             if (get_bits_left(&s->gb) < text_length * 8)

  1428.                 return -1;

  1429.             skip_bits_long(&s->gb, text_length * 8); // info text

  1430.         }

  1431. 

  1432.         skip_bits(&s->gb, 5); // bit resolution - 1

  1433.         skip_bits(&s->gb, 4); // max sample rate code

  1434.         channels = get_bits(&s->gb, 8) + 1;

  1435. 

  1436.         if (get_bits1(&s->gb)) { // 1-to-1 channels to speakers

  1437.             int spkr_remap_sets;

  1438.             int spkr_mask_size = 16;

  1439.             int num_spkrs[7];

  1440. 

  1441.             if (channels > 2)

  1442.                 embedded_stereo = get_bits1(&s->gb);

  1443.             if (channels > 6)

  1444.                 embedded_6ch = get_bits1(&s->gb);

  1445. 

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

  1447.                 spkr_mask_size = (get_bits(&s->gb, 2) + 1) << 2;

  1448.                 skip_bits(&s->gb, spkr_mask_size); // spkr activity mask

  1449.             }

  1450. 

  1451.             spkr_remap_sets = get_bits(&s->gb, 3);

  1452. 

  1453.             for (i = 0; i < spkr_remap_sets; i++) {

  1454.                 /* std layout mask for each remap set */

  1455.                 num_spkrs[i] = dca_exss_mask2count(get_bits(&s->gb, spkr_mask_size));

  1456.             }

  1457. 

  1458.             for (i = 0; i < spkr_remap_sets; i++) {

  1459.                 int num_dec_ch_remaps = get_bits(&s->gb, 5) + 1;

  1460.                 if (get_bits_left(&s->gb) < 0)

  1461.                     return -1;

  1462. 

  1463.                 for (j = 0; j < num_spkrs[i]; j++) {

  1464.                     int remap_dec_ch_mask = get_bits_long(&s->gb, num_dec_ch_remaps);

  1465.                     int num_dec_ch = av_popcount(remap_dec_ch_mask);

  1466.                     skip_bits_long(&s->gb, num_dec_ch * 5); // remap codes

  1467.                 }

  1468.             }

  1469. 

  1470.         } else {

  1471.             skip_bits(&s->gb, 3); // representation type

  1472.         }

  1473.     }

  1474. 

  1475.     drc_code_present = get_bits1(&s->gb);

  1476.     if (drc_code_present)

  1477.         get_bits(&s->gb, 8); // drc code

  1478. 

  1479.     if (get_bits1(&s->gb))

  1480.         skip_bits(&s->gb, 5); // dialog normalization code

  1481. 

  1482.     if (drc_code_present && embedded_stereo)

  1483.         get_bits(&s->gb, 8); // drc stereo code

  1484. 

  1485.     if (s->mix_metadata && get_bits1(&s->gb)) {

  1486.         skip_bits(&s->gb, 1); // external mix

  1487.         skip_bits(&s->gb, 6); // post mix gain code

  1488. 

  1489.         if (get_bits(&s->gb, 2) != 3) // mixer drc code

  1490.             skip_bits(&s->gb, 3); // drc limit

  1491.         else

  1492.             skip_bits(&s->gb, 8); // custom drc code

  1493. 

  1494.         if (get_bits1(&s->gb)) // channel specific scaling

  1495.             for (i = 0; i < s->num_mix_configs; i++)

  1496.                 skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6); // scale codes

  1497.         else

  1498.             skip_bits_long(&s->gb, s->num_mix_configs * 6); // scale codes

  1499. 

  1500.         for (i = 0; i < s->num_mix_configs; i++) {

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

  1502.                 return -1;

  1503.             dca_exss_skip_mix_coeffs(&s->gb, channels, s->mix_config_num_ch[i]);

  1504.             if (embedded_6ch)

  1505.                 dca_exss_skip_mix_coeffs(&s->gb, 6, s->mix_config_num_ch[i]);

  1506.             if (embedded_stereo)

  1507.                 dca_exss_skip_mix_coeffs(&s->gb, 2, s->mix_config_num_ch[i]);

  1508.         }

  1509.     }

  1510. 

  1511.     switch (get_bits(&s->gb, 2)) {

  1512.     case 0: extensions_mask = get_bits(&s->gb, 12); break;

  1513.     case 1: extensions_mask = DCA_EXT_EXSS_XLL;     break;

  1514.     case 2: extensions_mask = DCA_EXT_EXSS_LBR;     break;

  1515.     case 3: extensions_mask = 0; /* aux coding */   break;

  1516.     }

  1517. 

  1518.     /* not parsed further, we were only interested in the extensions mask */

  1519. 

  1520.     if (get_bits_left(&s->gb) < 0)

  1521.         return -1;

  1522. 

  1523.     if (get_bits_count(&s->gb) - header_pos > header_size * 8) {

  1524.         av_log(s->avctx, AV_LOG_WARNING, "Asset header size mismatch.\n");

  1525.         return -1;

  1526.     }

  1527.     skip_bits_long(&s->gb, header_pos + header_size * 8 - get_bits_count(&s->gb));

  1528. 

  1529.     if (extensions_mask & DCA_EXT_EXSS_XLL)

  1530.         s->profile = FF_PROFILE_DTS_HD_MA;

  1531.     else if (extensions_mask & (DCA_EXT_EXSS_XBR | DCA_EXT_EXSS_X96 |

  1532.                                 DCA_EXT_EXSS_XXCH))

  1533.         s->profile = FF_PROFILE_DTS_HD_HRA;

  1534. 

  1535.     if (!(extensions_mask & DCA_EXT_CORE))

  1536.         av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n");

  1537.     if ((extensions_mask & DCA_CORE_EXTS) != s->core_ext_mask)

  1538.         av_log(s->avctx, AV_LOG_WARNING, "DTS extensions detection mismatch (%d, %d)\n",

  1539.                extensions_mask & DCA_CORE_EXTS, s->core_ext_mask);

  1540. 

  1541.     return 0;

  1542. }

  1543. 

  1544. /**

  1545.  * Parse extension substream header (HD)

  1546.  */

  1547. static void dca_exss_parse_header(DCAContext *s)

  1548. {

  1549.     int ss_index;

  1550.     int blownup;

  1551.     int num_audiop = 1;

  1552.     int num_assets = 1;

  1553.     int active_ss_mask[8];

  1554.     int i, j;

  1555. 

  1556.     if (get_bits_left(&s->gb) < 52)

  1557.         return;

  1558. 

  1559.     skip_bits(&s->gb, 8); // user data

  1560.     ss_index = get_bits(&s->gb, 2);

  1561. 

  1562.     blownup = get_bits1(&s->gb);

  1563.     skip_bits(&s->gb, 8 + 4 * blownup); // header_size

  1564.     skip_bits(&s->gb, 16 + 4 * blownup); // hd_size

  1565. 

  1566.     s->static_fields = get_bits1(&s->gb);

  1567.     if (s->static_fields) {

  1568.         skip_bits(&s->gb, 2); // reference clock code

  1569.         skip_bits(&s->gb, 3); // frame duration code

  1570. 

  1571.         if (get_bits1(&s->gb))

  1572.             skip_bits_long(&s->gb, 36); // timestamp

  1573. 

  1574.         /* a single stream can contain multiple audio assets that can be

  1575.          * combined to form multiple audio presentations */

  1576. 

  1577.         num_audiop = get_bits(&s->gb, 3) + 1;

  1578.         if (num_audiop > 1) {

  1579.             av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio presentations.");

  1580.             /* ignore such streams for now */

  1581.             return;

  1582.         }

  1583. 

  1584.         num_assets = get_bits(&s->gb, 3) + 1;

  1585.         if (num_assets > 1) {

  1586.             av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio assets.");

  1587.             /* ignore such streams for now */

  1588.             return;

  1589.         }

  1590. 

  1591.         for (i = 0; i < num_audiop; i++)

  1592.             active_ss_mask[i] = get_bits(&s->gb, ss_index + 1);

  1593. 

  1594.         for (i = 0; i < num_audiop; i++)

  1595.             for (j = 0; j <= ss_index; j++)

  1596.                 if (active_ss_mask[i] & (1 << j))

  1597.                     skip_bits(&s->gb, 8); // active asset mask

  1598. 

  1599.         s->mix_metadata = get_bits1(&s->gb);

  1600.         if (s->mix_metadata) {

  1601.             int mix_out_mask_size;

  1602. 

  1603.             skip_bits(&s->gb, 2); // adjustment level

  1604.             mix_out_mask_size = (get_bits(&s->gb, 2) + 1) << 2;

  1605.             s->num_mix_configs = get_bits(&s->gb, 2) + 1;

  1606. 

  1607.             for (i = 0; i < s->num_mix_configs; i++) {

  1608.                 int mix_out_mask = get_bits(&s->gb, mix_out_mask_size);

  1609.                 s->mix_config_num_ch[i] = dca_exss_mask2count(mix_out_mask);

  1610.             }

  1611.         }

  1612.     }

  1613. 

  1614.     for (i = 0; i < num_assets; i++)

  1615.         skip_bits_long(&s->gb, 16 + 4 * blownup); // asset size

  1616. 

  1617.     for (i = 0; i < num_assets; i++) {

  1618.         if (dca_exss_parse_asset_header(s))

  1619.             return;

  1620.     }

  1621. 

  1622.     /* not parsed further, we were only interested in the extensions mask

  1623.      * from the asset header */

  1624. }

  1625. 

  1626. /**

  1627.  * Main frame decoding function

  1628.  * FIXME add arguments

  1629.  */

  1630. static int dca_decode_frame(AVCodecContext * avctx,

  1631.                             void *data, int *data_size,

  1632.                             AVPacket *avpkt)

  1633. {

  1634.     const uint8_t *buf = avpkt->data;

  1635.     int buf_size = avpkt->size;

  1636. 

  1637.     int lfe_samples;

  1638.     int num_core_channels = 0;

  1639.     int i;

  1640.     float   *samples_flt = data;

  1641.     int16_t *samples_s16 = data;

  1642.     int out_size;

  1643.     DCAContext *s = avctx->priv_data;

  1644.     int channels;

  1645.     int core_ss_end;

  1646. 

  1647. 

  1648.     s->xch_present = 0;

  1649. 

  1650.     s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer,

  1651.                                                DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);

  1652.     if (s->dca_buffer_size == -1) {

  1653.         av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");

  1654.         return -1;

  1655.     }

  1656. 

  1657.     init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);

  1658.     if (dca_parse_frame_header(s) < 0) {

  1659.         //seems like the frame is corrupt, try with the next one

  1660.         *data_size=0;

  1661.         return buf_size;

  1662.     }

  1663.     //set AVCodec values with parsed data

  1664.     avctx->sample_rate = s->sample_rate;

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

  1666.     avctx->frame_size = s->sample_blocks * 32;

  1667. 

  1668.     s->profile = FF_PROFILE_DTS;

  1669. 

  1670.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1671.         dca_decode_block(s, 0, i);

  1672.     }

  1673. 

  1674.     /* record number of core channels incase less than max channels are requested */

  1675.     num_core_channels = s->prim_channels;

  1676. 

  1677.     if (s->ext_coding)

  1678.         s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];

  1679.     else

  1680.         s->core_ext_mask = 0;

  1681. 

  1682.     core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;

  1683. 

  1684.     /* only scan for extensions if ext_descr was unknown or indicated a

  1685.      * supported XCh extension */

  1686.     if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {

  1687. 

  1688.         /* if ext_descr was unknown, clear s->core_ext_mask so that the

  1689.          * extensions scan can fill it up */

  1690.         s->core_ext_mask = FFMAX(s->core_ext_mask, 0);

  1691. 

  1692.         /* extensions start at 32-bit boundaries into bitstream */

  1693.         skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1694. 

  1695.     while(core_ss_end - get_bits_count(&s->gb) >= 32) {

  1696.         uint32_t bits = get_bits_long(&s->gb, 32);

  1697. 

  1698.         switch(bits) {

  1699.         case 0x5a5a5a5a: {

  1700.             int ext_amode, xch_fsize;

  1701. 

  1702.             s->xch_base_channel = s->prim_channels;

  1703. 

  1704.             /* validate sync word using XCHFSIZE field */

  1705.             xch_fsize = show_bits(&s->gb, 10);

  1706.             if((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&

  1707.                (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))

  1708.                 continue;

  1709. 

  1710.             /* skip length-to-end-of-frame field for the moment */

  1711.             skip_bits(&s->gb, 10);

  1712. 

  1713.             s->core_ext_mask |= DCA_EXT_XCH;

  1714. 

  1715.             /* extension amode should == 1, number of channels in extension */

  1716.             /* AFAIK XCh is not used for more channels */

  1717.             if ((ext_amode = get_bits(&s->gb, 4)) != 1) {

  1718.                 av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"

  1719.                        " supported!\n",ext_amode);

  1720.                 continue;

  1721.             }

  1722. 

  1723.             /* much like core primary audio coding header */

  1724.             dca_parse_audio_coding_header(s, s->xch_base_channel);

  1725. 

  1726.             for (i = 0; i < (s->sample_blocks / 8); i++) {

  1727.                 dca_decode_block(s, s->xch_base_channel, i);

  1728.             }

  1729. 

  1730.             s->xch_present = 1;

  1731.             break;

  1732.         }

  1733.         case 0x47004a03:

  1734.             /* XXCh: extended channels */

  1735.             /* usually found either in core or HD part in DTS-HD HRA streams,

  1736.              * but not in DTS-ES which contains XCh extensions instead */

  1737.             s->core_ext_mask |= DCA_EXT_XXCH;

  1738.             break;

  1739. 

  1740.         case 0x1d95f262: {

  1741.             int fsize96 = show_bits(&s->gb, 12) + 1;

  1742.             if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)

  1743.                 continue;

  1744. 

  1745.             av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n", get_bits_count(&s->gb));

  1746.             skip_bits(&s->gb, 12);

  1747.             av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);

  1748.             av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));

  1749. 

  1750.             s->core_ext_mask |= DCA_EXT_X96;

  1751.             break;

  1752.         }

  1753.         }

  1754. 

  1755.         skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1756.     }

  1757. 

  1758.     } else {

  1759.         /* no supported extensions, skip the rest of the core substream */

  1760.         skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));

  1761.     }

  1762. 

  1763.     if (s->core_ext_mask & DCA_EXT_X96)

  1764.         s->profile = FF_PROFILE_DTS_96_24;

  1765.     else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))

  1766.         s->profile = FF_PROFILE_DTS_ES;

  1767. 

  1768.     /* check for ExSS (HD part) */

  1769.     if (s->dca_buffer_size - s->frame_size > 32

  1770.         && get_bits_long(&s->gb, 32) == DCA_HD_MARKER)

  1771.         dca_exss_parse_header(s);

  1772. 

  1773.     avctx->profile = s->profile;

  1774. 

  1775.     channels = s->prim_channels + !!s->lfe;

  1776. 

  1777.     if (s->amode<16) {

  1778.         avctx->channel_layout = dca_core_channel_layout[s->amode];

  1779. 

  1780.         if (s->xch_present && (!avctx->request_channels ||

  1781.                                avctx->request_channels > num_core_channels + !!s->lfe)) {

  1782.             avctx->channel_layout |= AV_CH_BACK_CENTER;

  1783.             if (s->lfe) {

  1784.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1785.                 s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode];

  1786.             } else {

  1787.                 s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];

  1788.             }

  1789.         } else {

  1790.             channels = num_core_channels + !!s->lfe;

  1791.             s->xch_present = 0; /* disable further xch processing */

  1792.             if (s->lfe) {

  1793.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

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

  1795.             } else

  1796.                 s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];

  1797.         }

  1798. 

  1799.         if (channels > !!s->lfe &&

  1800.             s->channel_order_tab[channels - 1 - !!s->lfe] < 0)

  1801.             return -1;

  1802. 

  1803.         if (avctx->request_channels == 2 && s->prim_channels > 2) {

  1804.             channels = 2;

  1805.             s->output = DCA_STEREO;

  1806.             avctx->channel_layout = AV_CH_LAYOUT_STEREO;

  1807.         }

  1808.         else if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) {

  1809.             static const int8_t dca_channel_order_native[9] = { 0, 1, 2, 3, 4, 5, 6, 7, 8 };

  1810.             s->channel_order_tab = dca_channel_order_native;

  1811.         }

  1812.     } else {

  1813.         av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n",s->amode);

  1814.         return -1;

  1815.     }

  1816. 

  1817. 

  1818.     /* There is nothing that prevents a dts frame to change channel configuration

  1819.        but FFmpeg doesn't support that so only set the channels if it is previously

  1820.        unset. Ideally during the first probe for channels the crc should be checked

  1821.        and only set avctx->channels when the crc is ok. Right now the decoder could

  1822.        set the channels based on a broken first frame.*/

  1823.     if (s->is_channels_set == 0) {

  1824.         s->is_channels_set = 1;

  1825.         avctx->channels = channels;

  1826.     }

  1827.     if (avctx->channels != channels) {

  1828.         av_log(avctx, AV_LOG_ERROR, "DCA decoder does not support number of "

  1829.                "channels changing in stream. Skipping frame.\n");

  1830.         return -1;

  1831.     }

  1832. 

  1833.     out_size = 256 / 8 * s->sample_blocks * channels *

  1834.                av_get_bytes_per_sample(avctx->sample_fmt);

  1835.     if (*data_size < out_size)

  1836.         return -1;

  1837.     *data_size = out_size;

  1838. 

  1839.     /* filter to get final output */

  1840.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1841.         dca_filter_channels(s, i);

  1842. 

  1843.         /* If this was marked as a DTS-ES stream we need to subtract back- */

  1844.         /* channel from SL & SR to remove matrixed back-channel signal */

  1845.         if((s->source_pcm_res & 1) && s->xch_present) {

  1846.             float* back_chan = s->samples + s->channel_order_tab[s->xch_base_channel] * 256;

  1847.             float* lt_chan   = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256;

  1848.             float* rt_chan   = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;

  1849.             s->dsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);

  1850.             s->dsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);

  1851.         }

  1852. 

  1853.         if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {

  1854.             s->fmt_conv.float_interleave(samples_flt, s->samples_chanptr, 256,

  1855.                                          channels);

  1856.             samples_flt += 256 * channels;

  1857.         } else {

  1858.             s->fmt_conv.float_to_int16_interleave(samples_s16,

  1859.                                                   s->samples_chanptr, 256,

  1860.                                                   channels);

  1861.             samples_s16 += 256 * channels;

  1862.         }

  1863.     }

  1864. 

  1865.     /* update lfe history */

  1866.     lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);

  1867.     for (i = 0; i < 2 * s->lfe * 4; i++) {

  1868.         s->lfe_data[i] = s->lfe_data[i + lfe_samples];

  1869.     }

  1870. 

  1871.     return buf_size;

  1872. }

  1873. 

  1874. 

  1875. 

  1876. /**

  1877.  * DCA initialization

  1878.  *

  1879.  * @param avctx     pointer to the AVCodecContext

  1880.  */

  1881. 

  1882. static av_cold int dca_decode_init(AVCodecContext * avctx)

  1883. {

  1884.     DCAContext *s = avctx->priv_data;

  1885.     int i;

  1886. 

  1887.     s->avctx = avctx;

  1888.     dca_init_vlcs();

  1889. 

  1890.     dsputil_init(&s->dsp, avctx);

  1891.     ff_mdct_init(&s->imdct, 6, 1, 1.0);

  1892.     ff_synth_filter_init(&s->synth);

  1893.     ff_dcadsp_init(&s->dcadsp);

  1894.     ff_fmt_convert_init(&s->fmt_conv, avctx);

  1895. 

  1896.     for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)

  1897.         s->samples_chanptr[i] = s->samples + i * 256;

  1898. 

  1899.     if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {

  1900.         avctx->sample_fmt = AV_SAMPLE_FMT_FLT;

  1901.         s->scale_bias = 1.0 / 32768.0;

  1902.     } else {

  1903.         avctx->sample_fmt = AV_SAMPLE_FMT_S16;

  1904.         s->scale_bias = 1.0;

  1905.     }

  1906. 

  1907.     /* allow downmixing to stereo */

  1908.     if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&

  1909.         avctx->request_channels == 2) {

  1910.         avctx->channels = avctx->request_channels;

  1911.     }

  1912. 

  1913.     return 0;

  1914. }

  1915. 

  1916. static av_cold int dca_decode_end(AVCodecContext * avctx)

  1917. {

  1918.     DCAContext *s = avctx->priv_data;

  1919.     ff_mdct_end(&s->imdct);

  1920.     return 0;

  1921. }

  1922. 

  1923. static const AVProfile profiles[] = {

  1924.     { FF_PROFILE_DTS,        "DTS"        },

  1925.     { FF_PROFILE_DTS_ES,     "DTS-ES"     },

  1926.     { FF_PROFILE_DTS_96_24,  "DTS 96/24"  },

  1927.     { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },

  1928.     { FF_PROFILE_DTS_HD_MA,  "DTS-HD MA"  },

  1929.     { FF_PROFILE_UNKNOWN },

  1930. };

  1931. 

  1932. AVCodec ff_dca_decoder = {

  1933.     .name = "dca",

  1934.     .type = AVMEDIA_TYPE_AUDIO,

  1935.     .id = CODEC_ID_DTS,

  1936.     .priv_data_size = sizeof(DCAContext),

  1937.     .init = dca_decode_init,

  1938.     .decode = dca_decode_frame,

  1939.     .close = dca_decode_end,

  1940.     .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),

  1941.     .capabilities = CODEC_CAP_CHANNEL_CONF,

  1942.     .sample_fmts = (const enum AVSampleFormat[]) {

  1943.         AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE

  1944.     },

  1945.     .profiles = NULL_IF_CONFIG_SMALL(profiles),

  1946. };