falkTX
/
FFmpeg
mirror of https://github.com/falkTX/FFmpeg.git
1
0
Fork 0
Code Issues 0 Releases 338 Wiki Activity
You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
35865 Commits
32 Branches
296MB
Tree: b56adc7842
Branches Tags
${ item.name }
Create branch ${ searchTerm }
from 'b56adc7842'
${ noResults }
FFmpeg/libavcodec/dca.c

1957 lines
71KB
Raw Blame History 

  1. /*

  2.  * DCA compatible decoder

  3.  * Copyright (C) 2004 Gildas Bazin

  4.  * Copyright (C) 2004 Benjamin Zores

  5.  * Copyright (C) 2006 Benjamin Larsson

  6.  * Copyright (C) 2007 Konstantin Shishkov

  7.  *

  8.  * This file is part of FFmpeg.

  9.  *

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

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

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

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

  14.  *

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

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

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

  18.  * Lesser General Public License for more details.

  19.  *

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

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

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

  23.  */

  24. 

  25. #include <math.h>

  26. #include <stddef.h>

  27. #include <stdio.h>

  28. 

  29. #include "libavutil/common.h"

  30. #include "libavutil/intmath.h"

  31. #include "libavutil/intreadwrite.h"

  32. #include "libavutil/audioconvert.h"

  33. #include "avcodec.h"

  34. #include "dsputil.h"

  35. #include "fft.h"

  36. #include "get_bits.h"

  37. #include "put_bits.h"

  38. #include "dcadata.h"

  39. #include "dcahuff.h"

  40. #include "dca.h"

  41. #include "synth_filter.h"

  42. #include "dcadsp.h"

  43. #include "fmtconvert.h"

  44. 

  45. #if ARCH_ARM

  46. #   include "arm/dca.h"

  47. #endif

  48. 

  49. //#define TRACE

  50. 

  51. #define DCA_PRIM_CHANNELS_MAX (7)

  52. #define DCA_SUBBANDS (32)

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

  54. #define DCA_SUBSUBFRAMES_MAX (4)

  55. #define DCA_SUBFRAMES_MAX (16)

  56. #define DCA_BLOCKS_MAX (16)

  57. #define DCA_LFE_MAX (3)

  58. 

  59. enum DCAMode {

  60.     DCA_MONO = 0,

  61.     DCA_CHANNEL,

  62.     DCA_STEREO,

  63.     DCA_STEREO_SUMDIFF,

  64.     DCA_STEREO_TOTAL,

  65.     DCA_3F,

  66.     DCA_2F1R,

  67.     DCA_3F1R,

  68.     DCA_2F2R,

  69.     DCA_3F2R,

  70.     DCA_4F2R

  71. };

  72. 

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

  74. enum DCAExSSSpeakerMask {

  75.     DCA_EXSS_FRONT_CENTER          = 0x0001,

  76.     DCA_EXSS_FRONT_LEFT_RIGHT      = 0x0002,

  77.     DCA_EXSS_SIDE_REAR_LEFT_RIGHT  = 0x0004,

  78.     DCA_EXSS_LFE                   = 0x0008,

  79.     DCA_EXSS_REAR_CENTER           = 0x0010,

  80.     DCA_EXSS_FRONT_HIGH_LEFT_RIGHT = 0x0020,

  81.     DCA_EXSS_REAR_LEFT_RIGHT       = 0x0040,

  82.     DCA_EXSS_FRONT_HIGH_CENTER     = 0x0080,

  83.     DCA_EXSS_OVERHEAD              = 0x0100,

  84.     DCA_EXSS_CENTER_LEFT_RIGHT     = 0x0200,

  85.     DCA_EXSS_WIDE_LEFT_RIGHT       = 0x0400,

  86.     DCA_EXSS_SIDE_LEFT_RIGHT       = 0x0800,

  87.     DCA_EXSS_LFE2                  = 0x1000,

  88.     DCA_EXSS_SIDE_HIGH_LEFT_RIGHT  = 0x2000,

  89.     DCA_EXSS_REAR_HIGH_CENTER      = 0x4000,

  90.     DCA_EXSS_REAR_HIGH_LEFT_RIGHT  = 0x8000,

  91. };

  92. 

  93. enum DCAExtensionMask {

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

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

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

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

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

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

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

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

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

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

  104. };

  105. 

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

  107. static const int dca_ext_audio_descr_mask[] = {

  108.     DCA_EXT_XCH,

  109.     -1,

  110.     DCA_EXT_X96,

  111.     DCA_EXT_XCH | DCA_EXT_X96,

  112.     -1,

  113.     -1,

  114.     DCA_EXT_XXCH,

  115.     -1,

  116. };

  117. 

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

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

  120. 

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

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

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

  124.  *

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

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

  127.  * OV -> center back

  128.  * All 2 channel configurations -> AV_CH_LAYOUT_STEREO

  129.  */

  130. 

  131. static const uint64_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.     AVFrame frame;

  265.     /* Frame header */

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

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

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

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

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

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

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

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

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

  275. 

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

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

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

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

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

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

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

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

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

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

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

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

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

  289.     int copy_history;           ///< copy history

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

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

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

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

  294. 

  295.     /* Primary audio coding header */

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

  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.     skip_bits_long(&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 AVERROR_INVALIDDATA;

  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 AVERROR_INVALIDDATA;

  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 AVERROR_INVALIDDATA;

  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 AVERROR_INVALIDDATA;

  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 AVERROR_INVALIDDATA;

  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 AVERROR_INVALIDDATA;

  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 AVERROR_INVALIDDATA;

  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 AVERROR_INVALIDDATA;

  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. #ifndef decode_blockcodes

  1042. /* Very compact version of the block code decoder that does not use table

  1043.  * look-up but is slightly slower */

  1044. static int decode_blockcode(int code, int levels, int *values)

  1045. {

  1046.     int i;

  1047.     int offset = (levels - 1) >> 1;

  1048. 

  1049.     for (i = 0; i < 4; i++) {

  1050.         int div = FASTDIV(code, levels);

  1051.         values[i] = code - offset - div*levels;

  1052.         code = div;

  1053.     }

  1054. 

  1055.     return code;

  1056. }

  1057. 

  1058. static int decode_blockcodes(int code1, int code2, int levels, int *values)

  1059. {

  1060.     return decode_blockcode(code1, levels, values) |

  1061.            decode_blockcode(code2, levels, values + 4);

  1062. }

  1063. #endif

  1064. 

  1065. static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };

  1066. static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };

  1067. 

  1068. #ifndef int8x8_fmul_int32

  1069. static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)

  1070. {

  1071.     float fscale = scale / 16.0;

  1072.     int i;

  1073.     for (i = 0; i < 8; i++)

  1074.         dst[i] = src[i] * fscale;

  1075. }

  1076. #endif

  1077. 

  1078. static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)

  1079. {

  1080.     int k, l;

  1081.     int subsubframe = s->current_subsubframe;

  1082. 

  1083.     const float *quant_step_table;

  1084. 

  1085.     /* FIXME */

  1086.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1087.     LOCAL_ALIGNED_16(int, block, [8]);

  1088. 

  1089.     /*

  1090.      * Audio data

  1091.      */

  1092. 

  1093.     /* Select quantization step size table */

  1094.     if (s->bit_rate_index == 0x1f)

  1095.         quant_step_table = lossless_quant_d;

  1096.     else

  1097.         quant_step_table = lossy_quant_d;

  1098. 

  1099.     for (k = base_channel; k < s->prim_channels; k++) {

  1100.         if (get_bits_left(&s->gb) < 0)

  1101.             return AVERROR_INVALIDDATA;

  1102. 

  1103.         for (l = 0; l < s->vq_start_subband[k]; l++) {

  1104.             int m;

  1105. 

  1106.             /* Select the mid-tread linear quantizer */

  1107.             int abits = s->bitalloc[k][l];

  1108. 

  1109.             float quant_step_size = quant_step_table[abits];

  1110. 

  1111.             /*

  1112.              * Determine quantization index code book and its type

  1113.              */

  1114. 

  1115.             /* Select quantization index code book */

  1116.             int sel = s->quant_index_huffman[k][abits];

  1117. 

  1118.             /*

  1119.              * Extract bits from the bit stream

  1120.              */

  1121.             if (!abits){

  1122.                 memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));

  1123.             } else {

  1124.                 /* Deal with transients */

  1125.                 int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];

  1126.                 float rscale = quant_step_size * s->scale_factor[k][l][sfi] * s->scalefactor_adj[k][sel];

  1127. 

  1128.                 if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){

  1129.                     if (abits <= 7){

  1130.                         /* Block code */

  1131.                         int block_code1, block_code2, size, levels, err;

  1132. 

  1133.                         size = abits_sizes[abits-1];

  1134.                         levels = abits_levels[abits-1];

  1135. 

  1136.                         block_code1 = get_bits(&s->gb, size);

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

  1138.                         err = decode_blockcodes(block_code1, block_code2,

  1139.                                                 levels, block);

  1140.                         if (err) {

  1141.                             av_log(s->avctx, AV_LOG_ERROR,

  1142.                                    "ERROR: block code look-up failed\n");

  1143.                             return AVERROR_INVALIDDATA;

  1144.                         }

  1145.                     }else{

  1146.                         /* no coding */

  1147.                         for (m = 0; m < 8; m++)

  1148.                             block[m] = get_sbits(&s->gb, abits - 3);

  1149.                     }

  1150.                 }else{

  1151.                     /* Huffman coded */

  1152.                     for (m = 0; m < 8; m++)

  1153.                         block[m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);

  1154.                 }

  1155. 

  1156.                 s->fmt_conv.int32_to_float_fmul_scalar(subband_samples[k][l],

  1157.                                                   block, rscale, 8);

  1158.             }

  1159. 

  1160.             /*

  1161.              * Inverse ADPCM if in prediction mode

  1162.              */

  1163.             if (s->prediction_mode[k][l]) {

  1164.                 int n;

  1165.                 for (m = 0; m < 8; m++) {

  1166.                     for (n = 1; n <= 4; n++)

  1167.                         if (m >= n)

  1168.                             subband_samples[k][l][m] +=

  1169.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1170.                                  subband_samples[k][l][m - n] / 8192);

  1171.                         else if (s->predictor_history)

  1172.                             subband_samples[k][l][m] +=

  1173.                                 (adpcm_vb[s->prediction_vq[k][l]][n - 1] *

  1174.                                  s->subband_samples_hist[k][l][m - n +

  1175.                                                                4] / 8192);

  1176.                 }

  1177.             }

  1178.         }

  1179. 

  1180.         /*

  1181.          * Decode VQ encoded high frequencies

  1182.          */

  1183.         for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {

  1184.             /* 1 vector -> 32 samples but we only need the 8 samples

  1185.              * for this subsubframe. */

  1186.             int hfvq = s->high_freq_vq[k][l];

  1187. 

  1188.             if (!s->debug_flag & 0x01) {

  1189.                 av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n");

  1190.                 s->debug_flag |= 0x01;

  1191.             }

  1192. 

  1193.             int8x8_fmul_int32(subband_samples[k][l],

  1194.                               &high_freq_vq[hfvq][subsubframe * 8],

  1195.                               s->scale_factor[k][l][0]);

  1196.         }

  1197.     }

  1198. 

  1199.     /* Check for DSYNC after subsubframe */

  1200.     if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {

  1201.         if (0xFFFF == get_bits(&s->gb, 16)) {   /* 0xFFFF */

  1202. #ifdef TRACE

  1203.             av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");

  1204. #endif

  1205.         } else {

  1206.             av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");

  1207.         }

  1208.     }

  1209. 

  1210.     /* Backup predictor history for adpcm */

  1211.     for (k = base_channel; k < s->prim_channels; k++)

  1212.         for (l = 0; l < s->vq_start_subband[k]; l++)

  1213.             memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4],

  1214.                         4 * sizeof(subband_samples[0][0][0]));

  1215. 

  1216.     return 0;

  1217. }

  1218. 

  1219. static int dca_filter_channels(DCAContext * s, int block_index)

  1220. {

  1221.     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];

  1222.     int k;

  1223. 

  1224.     /* 32 subbands QMF */

  1225.     for (k = 0; k < s->prim_channels; k++) {

  1226. /*        static float pcm_to_double[8] =

  1227.             {32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/

  1228.          qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * s->channel_order_tab[k]],

  1229.                          M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ );

  1230.     }

  1231. 

  1232.     /* Down mixing */

  1233.     if (s->avctx->request_channels == 2 && s->prim_channels > 2) {

  1234.         dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab);

  1235.     }

  1236. 

  1237.     /* Generate LFE samples for this subsubframe FIXME!!! */

  1238.     if (s->output & DCA_LFE) {

  1239.         lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,

  1240.                               s->lfe_data + 2 * s->lfe * (block_index + 4),

  1241.                               &s->samples[256 * dca_lfe_index[s->amode]],

  1242.                               (1.0/256.0)*s->scale_bias);

  1243.         /* Outputs 20bits pcm samples */

  1244.     }

  1245. 

  1246.     return 0;

  1247. }

  1248. 

  1249. 

  1250. static int dca_subframe_footer(DCAContext * s, int base_channel)

  1251. {

  1252.     int aux_data_count = 0, i;

  1253. 

  1254.     /*

  1255.      * Unpack optional information

  1256.      */

  1257. 

  1258.     /* presumably optional information only appears in the core? */

  1259.     if (!base_channel) {

  1260.         if (s->timestamp)

  1261.             skip_bits_long(&s->gb, 32);

  1262. 

  1263.         if (s->aux_data)

  1264.             aux_data_count = get_bits(&s->gb, 6);

  1265. 

  1266.         for (i = 0; i < aux_data_count; i++)

  1267.             get_bits(&s->gb, 8);

  1268. 

  1269.         if (s->crc_present && (s->downmix || s->dynrange))

  1270.             get_bits(&s->gb, 16);

  1271.     }

  1272. 

  1273.     return 0;

  1274. }

  1275. 

  1276. /**

  1277.  * Decode a dca frame block

  1278.  *

  1279.  * @param s     pointer to the DCAContext

  1280.  */

  1281. 

  1282. static int dca_decode_block(DCAContext * s, int base_channel, int block_index)

  1283. {

  1284.     int ret;

  1285. 

  1286.     /* Sanity check */

  1287.     if (s->current_subframe >= s->subframes) {

  1288.         av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",

  1289.                s->current_subframe, s->subframes);

  1290.         return AVERROR_INVALIDDATA;

  1291.     }

  1292. 

  1293.     if (!s->current_subsubframe) {

  1294. #ifdef TRACE

  1295.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");

  1296. #endif

  1297.         /* Read subframe header */

  1298.         if ((ret = dca_subframe_header(s, base_channel, block_index)))

  1299.             return ret;

  1300.     }

  1301. 

  1302.     /* Read subsubframe */

  1303. #ifdef TRACE

  1304.     av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");

  1305. #endif

  1306.     if ((ret = dca_subsubframe(s, base_channel, block_index)))

  1307.         return ret;

  1308. 

  1309.     /* Update state */

  1310.     s->current_subsubframe++;

  1311.     if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {

  1312.         s->current_subsubframe = 0;

  1313.         s->current_subframe++;

  1314.     }

  1315.     if (s->current_subframe >= s->subframes) {

  1316. #ifdef TRACE

  1317.         av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");

  1318. #endif

  1319.         /* Read subframe footer */

  1320.         if ((ret = dca_subframe_footer(s, base_channel)))

  1321.             return ret;

  1322.     }

  1323. 

  1324.     return 0;

  1325. }

  1326. 

  1327. /**

  1328.  * Convert bitstream to one representation based on sync marker

  1329.  */

  1330. static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * dst,

  1331.                           int max_size)

  1332. {

  1333.     uint32_t mrk;

  1334.     int i, tmp;

  1335.     const uint16_t *ssrc = (const uint16_t *) src;

  1336.     uint16_t *sdst = (uint16_t *) dst;

  1337.     PutBitContext pb;

  1338. 

  1339.     if ((unsigned)src_size > (unsigned)max_size) {

  1340. //        av_log(NULL, AV_LOG_ERROR, "Input frame size larger than DCA_MAX_FRAME_SIZE!\n");

  1341. //        return -1;

  1342.         src_size = max_size;

  1343.     }

  1344. 

  1345.     mrk = AV_RB32(src);

  1346.     switch (mrk) {

  1347.     case DCA_MARKER_RAW_BE:

  1348.         memcpy(dst, src, src_size);

  1349.         return src_size;

  1350.     case DCA_MARKER_RAW_LE:

  1351.         for (i = 0; i < (src_size + 1) >> 1; i++)

  1352.             *sdst++ = av_bswap16(*ssrc++);

  1353.         return src_size;

  1354.     case DCA_MARKER_14B_BE:

  1355.     case DCA_MARKER_14B_LE:

  1356.         init_put_bits(&pb, dst, max_size);

  1357.         for (i = 0; i < (src_size + 1) >> 1; i++, src += 2) {

  1358.             tmp = ((mrk == DCA_MARKER_14B_BE) ? AV_RB16(src) : AV_RL16(src)) & 0x3FFF;

  1359.             put_bits(&pb, 14, tmp);

  1360.         }

  1361.         flush_put_bits(&pb);

  1362.         return (put_bits_count(&pb) + 7) >> 3;

  1363.     default:

  1364.         return AVERROR_INVALIDDATA;

  1365.     }

  1366. }

  1367. 

  1368. /**

  1369.  * Return the number of channels in an ExSS speaker mask (HD)

  1370.  */

  1371. static int dca_exss_mask2count(int mask)

  1372. {

  1373.     /* count bits that mean speaker pairs twice */

  1374.     return av_popcount(mask)

  1375.         + av_popcount(mask & (

  1376.             DCA_EXSS_CENTER_LEFT_RIGHT

  1377.           | DCA_EXSS_FRONT_LEFT_RIGHT

  1378.           | DCA_EXSS_FRONT_HIGH_LEFT_RIGHT

  1379.           | DCA_EXSS_WIDE_LEFT_RIGHT

  1380.           | DCA_EXSS_SIDE_LEFT_RIGHT

  1381.           | DCA_EXSS_SIDE_HIGH_LEFT_RIGHT

  1382.           | DCA_EXSS_SIDE_REAR_LEFT_RIGHT

  1383.           | DCA_EXSS_REAR_LEFT_RIGHT

  1384.           | DCA_EXSS_REAR_HIGH_LEFT_RIGHT

  1385.           ));

  1386. }

  1387. 

  1388. /**

  1389.  * Skip mixing coefficients of a single mix out configuration (HD)

  1390.  */

  1391. static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch)

  1392. {

  1393.     int i;

  1394. 

  1395.     for (i = 0; i < channels; i++) {

  1396.         int mix_map_mask = get_bits(gb, out_ch);

  1397.         int num_coeffs = av_popcount(mix_map_mask);

  1398.         skip_bits_long(gb, num_coeffs * 6);

  1399.     }

  1400. }

  1401. 

  1402. /**

  1403.  * Parse extension substream asset header (HD)

  1404.  */

  1405. static int dca_exss_parse_asset_header(DCAContext *s)

  1406. {

  1407.     int header_pos = get_bits_count(&s->gb);

  1408.     int header_size;

  1409.     int channels;

  1410.     int embedded_stereo = 0;

  1411.     int embedded_6ch = 0;

  1412.     int drc_code_present;

  1413.     int extensions_mask;

  1414.     int i, j;

  1415. 

  1416.     if (get_bits_left(&s->gb) < 16)

  1417.         return -1;

  1418. 

  1419.     /* We will parse just enough to get to the extensions bitmask with which

  1420.      * we can set the profile value. */

  1421. 

  1422.     header_size = get_bits(&s->gb, 9) + 1;

  1423.     skip_bits(&s->gb, 3); // asset index

  1424. 

  1425.     if (s->static_fields) {

  1426.         if (get_bits1(&s->gb))

  1427.             skip_bits(&s->gb, 4); // asset type descriptor

  1428.         if (get_bits1(&s->gb))

  1429.             skip_bits_long(&s->gb, 24); // language descriptor

  1430. 

  1431.         if (get_bits1(&s->gb)) {

  1432.             /* How can one fit 1024 bytes of text here if the maximum value

  1433.              * for the asset header size field above was 512 bytes? */

  1434.             int text_length = get_bits(&s->gb, 10) + 1;

  1435.             if (get_bits_left(&s->gb) < text_length * 8)

  1436.                 return -1;

  1437.             skip_bits_long(&s->gb, text_length * 8); // info text

  1438.         }

  1439. 

  1440.         skip_bits(&s->gb, 5); // bit resolution - 1

  1441.         skip_bits(&s->gb, 4); // max sample rate code

  1442.         channels = get_bits(&s->gb, 8) + 1;

  1443. 

  1444.         if (get_bits1(&s->gb)) { // 1-to-1 channels to speakers

  1445.             int spkr_remap_sets;

  1446.             int spkr_mask_size = 16;

  1447.             int num_spkrs[7];

  1448. 

  1449.             if (channels > 2)

  1450.                 embedded_stereo = get_bits1(&s->gb);

  1451.             if (channels > 6)

  1452.                 embedded_6ch = get_bits1(&s->gb);

  1453. 

  1454.             if (get_bits1(&s->gb)) {

  1455.                 spkr_mask_size = (get_bits(&s->gb, 2) + 1) << 2;

  1456.                 skip_bits(&s->gb, spkr_mask_size); // spkr activity mask

  1457.             }

  1458. 

  1459.             spkr_remap_sets = get_bits(&s->gb, 3);

  1460. 

  1461.             for (i = 0; i < spkr_remap_sets; i++) {

  1462.                 /* std layout mask for each remap set */

  1463.                 num_spkrs[i] = dca_exss_mask2count(get_bits(&s->gb, spkr_mask_size));

  1464.             }

  1465. 

  1466.             for (i = 0; i < spkr_remap_sets; i++) {

  1467.                 int num_dec_ch_remaps = get_bits(&s->gb, 5) + 1;

  1468.                 if (get_bits_left(&s->gb) < 0)

  1469.                     return -1;

  1470. 

  1471.                 for (j = 0; j < num_spkrs[i]; j++) {

  1472.                     int remap_dec_ch_mask = get_bits_long(&s->gb, num_dec_ch_remaps);

  1473.                     int num_dec_ch = av_popcount(remap_dec_ch_mask);

  1474.                     skip_bits_long(&s->gb, num_dec_ch * 5); // remap codes

  1475.                 }

  1476.             }

  1477. 

  1478.         } else {

  1479.             skip_bits(&s->gb, 3); // representation type

  1480.         }

  1481.     }

  1482. 

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

  1484.     if (drc_code_present)

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

  1486. 

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

  1488.         skip_bits(&s->gb, 5); // dialog normalization code

  1489. 

  1490.     if (drc_code_present && embedded_stereo)

  1491.         get_bits(&s->gb, 8); // drc stereo code

  1492. 

  1493.     if (s->mix_metadata && get_bits1(&s->gb)) {

  1494.         skip_bits(&s->gb, 1); // external mix

  1495.         skip_bits(&s->gb, 6); // post mix gain code

  1496. 

  1497.         if (get_bits(&s->gb, 2) != 3) // mixer drc code

  1498.             skip_bits(&s->gb, 3); // drc limit

  1499.         else

  1500.             skip_bits(&s->gb, 8); // custom drc code

  1501. 

  1502.         if (get_bits1(&s->gb)) // channel specific scaling

  1503.             for (i = 0; i < s->num_mix_configs; i++)

  1504.                 skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6); // scale codes

  1505.         else

  1506.             skip_bits_long(&s->gb, s->num_mix_configs * 6); // scale codes

  1507. 

  1508.         for (i = 0; i < s->num_mix_configs; i++) {

  1509.             if (get_bits_left(&s->gb) < 0)

  1510.                 return -1;

  1511.             dca_exss_skip_mix_coeffs(&s->gb, channels, s->mix_config_num_ch[i]);

  1512.             if (embedded_6ch)

  1513.                 dca_exss_skip_mix_coeffs(&s->gb, 6, s->mix_config_num_ch[i]);

  1514.             if (embedded_stereo)

  1515.                 dca_exss_skip_mix_coeffs(&s->gb, 2, s->mix_config_num_ch[i]);

  1516.         }

  1517.     }

  1518. 

  1519.     switch (get_bits(&s->gb, 2)) {

  1520.     case 0: extensions_mask = get_bits(&s->gb, 12); break;

  1521.     case 1: extensions_mask = DCA_EXT_EXSS_XLL;     break;

  1522.     case 2: extensions_mask = DCA_EXT_EXSS_LBR;     break;

  1523.     case 3: extensions_mask = 0; /* aux coding */   break;

  1524.     }

  1525. 

  1526.     /* not parsed further, we were only interested in the extensions mask */

  1527. 

  1528.     if (get_bits_left(&s->gb) < 0)

  1529.         return -1;

  1530. 

  1531.     if (get_bits_count(&s->gb) - header_pos > header_size * 8) {

  1532.         av_log(s->avctx, AV_LOG_WARNING, "Asset header size mismatch.\n");

  1533.         return -1;

  1534.     }

  1535.     skip_bits_long(&s->gb, header_pos + header_size * 8 - get_bits_count(&s->gb));

  1536. 

  1537.     if (extensions_mask & DCA_EXT_EXSS_XLL)

  1538.         s->profile = FF_PROFILE_DTS_HD_MA;

  1539.     else if (extensions_mask & (DCA_EXT_EXSS_XBR | DCA_EXT_EXSS_X96 |

  1540.                                 DCA_EXT_EXSS_XXCH))

  1541.         s->profile = FF_PROFILE_DTS_HD_HRA;

  1542. 

  1543.     if (!(extensions_mask & DCA_EXT_CORE))

  1544.         av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n");

  1545.     if ((extensions_mask & DCA_CORE_EXTS) != s->core_ext_mask)

  1546.         av_log(s->avctx, AV_LOG_WARNING, "DTS extensions detection mismatch (%d, %d)\n",

  1547.                extensions_mask & DCA_CORE_EXTS, s->core_ext_mask);

  1548. 

  1549.     return 0;

  1550. }

  1551. 

  1552. /**

  1553.  * Parse extension substream header (HD)

  1554.  */

  1555. static void dca_exss_parse_header(DCAContext *s)

  1556. {

  1557.     int ss_index;

  1558.     int blownup;

  1559.     int num_audiop = 1;

  1560.     int num_assets = 1;

  1561.     int active_ss_mask[8];

  1562.     int i, j;

  1563. 

  1564.     if (get_bits_left(&s->gb) < 52)

  1565.         return;

  1566. 

  1567.     skip_bits(&s->gb, 8); // user data

  1568.     ss_index = get_bits(&s->gb, 2);

  1569. 

  1570.     blownup = get_bits1(&s->gb);

  1571.     skip_bits(&s->gb, 8 + 4 * blownup); // header_size

  1572.     skip_bits(&s->gb, 16 + 4 * blownup); // hd_size

  1573. 

  1574.     s->static_fields = get_bits1(&s->gb);

  1575.     if (s->static_fields) {

  1576.         skip_bits(&s->gb, 2); // reference clock code

  1577.         skip_bits(&s->gb, 3); // frame duration code

  1578. 

  1579.         if (get_bits1(&s->gb))

  1580.             skip_bits_long(&s->gb, 36); // timestamp

  1581. 

  1582.         /* a single stream can contain multiple audio assets that can be

  1583.          * combined to form multiple audio presentations */

  1584. 

  1585.         num_audiop = get_bits(&s->gb, 3) + 1;

  1586.         if (num_audiop > 1) {

  1587.             av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio presentations.");

  1588.             /* ignore such streams for now */

  1589.             return;

  1590.         }

  1591. 

  1592.         num_assets = get_bits(&s->gb, 3) + 1;

  1593.         if (num_assets > 1) {

  1594.             av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio assets.");

  1595.             /* ignore such streams for now */

  1596.             return;

  1597.         }

  1598. 

  1599.         for (i = 0; i < num_audiop; i++)

  1600.             active_ss_mask[i] = get_bits(&s->gb, ss_index + 1);

  1601. 

  1602.         for (i = 0; i < num_audiop; i++)

  1603.             for (j = 0; j <= ss_index; j++)

  1604.                 if (active_ss_mask[i] & (1 << j))

  1605.                     skip_bits(&s->gb, 8); // active asset mask

  1606. 

  1607.         s->mix_metadata = get_bits1(&s->gb);

  1608.         if (s->mix_metadata) {

  1609.             int mix_out_mask_size;

  1610. 

  1611.             skip_bits(&s->gb, 2); // adjustment level

  1612.             mix_out_mask_size = (get_bits(&s->gb, 2) + 1) << 2;

  1613.             s->num_mix_configs = get_bits(&s->gb, 2) + 1;

  1614. 

  1615.             for (i = 0; i < s->num_mix_configs; i++) {

  1616.                 int mix_out_mask = get_bits(&s->gb, mix_out_mask_size);

  1617.                 s->mix_config_num_ch[i] = dca_exss_mask2count(mix_out_mask);

  1618.             }

  1619.         }

  1620.     }

  1621. 

  1622.     for (i = 0; i < num_assets; i++)

  1623.         skip_bits_long(&s->gb, 16 + 4 * blownup); // asset size

  1624. 

  1625.     for (i = 0; i < num_assets; i++) {

  1626.         if (dca_exss_parse_asset_header(s))

  1627.             return;

  1628.     }

  1629. 

  1630.     /* not parsed further, we were only interested in the extensions mask

  1631.      * from the asset header */

  1632. }

  1633. 

  1634. /**

  1635.  * Main frame decoding function

  1636.  * FIXME add arguments

  1637.  */

  1638. static int dca_decode_frame(AVCodecContext *avctx, void *data,

  1639.                             int *got_frame_ptr, AVPacket *avpkt)

  1640. {

  1641.     const uint8_t *buf = avpkt->data;

  1642.     int buf_size = avpkt->size;

  1643. 

  1644.     int lfe_samples;

  1645.     int num_core_channels = 0;

  1646.     int i, ret;

  1647.     float   *samples_flt;

  1648.     int16_t *samples_s16;

  1649.     DCAContext *s = avctx->priv_data;

  1650.     int channels;

  1651.     int core_ss_end;

  1652. 

  1653. 

  1654.     s->xch_present = 0;

  1655. 

  1656.     s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer,

  1657.                                                DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);

  1658.     if (s->dca_buffer_size == AVERROR_INVALIDDATA) {

  1659.         av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");

  1660.         return AVERROR_INVALIDDATA;

  1661.     }

  1662. 

  1663.     init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);

  1664.     if ((ret = dca_parse_frame_header(s)) < 0) {

  1665.         //seems like the frame is corrupt, try with the next one

  1666.         return ret;

  1667.     }

  1668.     //set AVCodec values with parsed data

  1669.     avctx->sample_rate = s->sample_rate;

  1670.     avctx->bit_rate = s->bit_rate;

  1671.     avctx->frame_size = s->sample_blocks * 32;

  1672. 

  1673.     s->profile = FF_PROFILE_DTS;

  1674. 

  1675.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1676.         if ((ret = dca_decode_block(s, 0, i))) {

  1677.             av_log(avctx, AV_LOG_ERROR, "error decoding block\n");

  1678.             return ret;

  1679.         }

  1680.     }

  1681. 

  1682.     /* record number of core channels incase less than max channels are requested */

  1683.     num_core_channels = s->prim_channels;

  1684. 

  1685.     if (s->ext_coding)

  1686.         s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];

  1687.     else

  1688.         s->core_ext_mask = 0;

  1689. 

  1690.     core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;

  1691. 

  1692.     /* only scan for extensions if ext_descr was unknown or indicated a

  1693.      * supported XCh extension */

  1694.     if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {

  1695. 

  1696.         /* if ext_descr was unknown, clear s->core_ext_mask so that the

  1697.          * extensions scan can fill it up */

  1698.         s->core_ext_mask = FFMAX(s->core_ext_mask, 0);

  1699. 

  1700.         /* extensions start at 32-bit boundaries into bitstream */

  1701.         skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1702. 

  1703.     while(core_ss_end - get_bits_count(&s->gb) >= 32) {

  1704.         uint32_t bits = get_bits_long(&s->gb, 32);

  1705. 

  1706.         switch(bits) {

  1707.         case 0x5a5a5a5a: {

  1708.             int ext_amode, xch_fsize;

  1709. 

  1710.             s->xch_base_channel = s->prim_channels;

  1711. 

  1712.             /* validate sync word using XCHFSIZE field */

  1713.             xch_fsize = show_bits(&s->gb, 10);

  1714.             if((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&

  1715.                (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))

  1716.                 continue;

  1717. 

  1718.             /* skip length-to-end-of-frame field for the moment */

  1719.             skip_bits(&s->gb, 10);

  1720. 

  1721.             s->core_ext_mask |= DCA_EXT_XCH;

  1722. 

  1723.             /* extension amode should == 1, number of channels in extension */

  1724.             /* AFAIK XCh is not used for more channels */

  1725.             if ((ext_amode = get_bits(&s->gb, 4)) != 1) {

  1726.                 av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"

  1727.                        " supported!\n",ext_amode);

  1728.                 continue;

  1729.             }

  1730. 

  1731.             /* much like core primary audio coding header */

  1732.             dca_parse_audio_coding_header(s, s->xch_base_channel);

  1733. 

  1734.             for (i = 0; i < (s->sample_blocks / 8); i++) {

  1735.                 if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {

  1736.                     av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");

  1737.                     continue;

  1738.                 }

  1739.             }

  1740. 

  1741.             s->xch_present = 1;

  1742.             break;

  1743.         }

  1744.         case 0x47004a03:

  1745.             /* XXCh: extended channels */

  1746.             /* usually found either in core or HD part in DTS-HD HRA streams,

  1747.              * but not in DTS-ES which contains XCh extensions instead */

  1748.             s->core_ext_mask |= DCA_EXT_XXCH;

  1749.             break;

  1750. 

  1751.         case 0x1d95f262: {

  1752.             int fsize96 = show_bits(&s->gb, 12) + 1;

  1753.             if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)

  1754.                 continue;

  1755. 

  1756.             av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n", get_bits_count(&s->gb));

  1757.             skip_bits(&s->gb, 12);

  1758.             av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);

  1759.             av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));

  1760. 

  1761.             s->core_ext_mask |= DCA_EXT_X96;

  1762.             break;

  1763.         }

  1764.         }

  1765. 

  1766.         skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

  1767.     }

  1768. 

  1769.     } else {

  1770.         /* no supported extensions, skip the rest of the core substream */

  1771.         skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));

  1772.     }

  1773. 

  1774.     if (s->core_ext_mask & DCA_EXT_X96)

  1775.         s->profile = FF_PROFILE_DTS_96_24;

  1776.     else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))

  1777.         s->profile = FF_PROFILE_DTS_ES;

  1778. 

  1779.     /* check for ExSS (HD part) */

  1780.     if (s->dca_buffer_size - s->frame_size > 32

  1781.         && get_bits_long(&s->gb, 32) == DCA_HD_MARKER)

  1782.         dca_exss_parse_header(s);

  1783. 

  1784.     avctx->profile = s->profile;

  1785. 

  1786.     channels = s->prim_channels + !!s->lfe;

  1787. 

  1788.     if (s->amode<16) {

  1789.         avctx->channel_layout = dca_core_channel_layout[s->amode];

  1790. 

  1791.         if (s->xch_present && (!avctx->request_channels ||

  1792.                                avctx->request_channels > num_core_channels + !!s->lfe)) {

  1793.             avctx->channel_layout |= AV_CH_BACK_CENTER;

  1794.             if (s->lfe) {

  1795.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

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

  1797.             } else {

  1798.                 s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];

  1799.             }

  1800.         } else {

  1801.             channels = num_core_channels + !!s->lfe;

  1802.             s->xch_present = 0; /* disable further xch processing */

  1803.             if (s->lfe) {

  1804.                 avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

  1805.                 s->channel_order_tab = dca_channel_reorder_lfe[s->amode];

  1806.             } else

  1807.                 s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];

  1808.         }

  1809. 

  1810.         if (channels > !!s->lfe &&

  1811.             s->channel_order_tab[channels - 1 - !!s->lfe] < 0)

  1812.             return AVERROR_INVALIDDATA;

  1813. 

  1814.         if (avctx->request_channels == 2 && s->prim_channels > 2) {

  1815.             channels = 2;

  1816.             s->output = DCA_STEREO;

  1817.             avctx->channel_layout = AV_CH_LAYOUT_STEREO;

  1818.         }

  1819.         else if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) {

  1820.             static const int8_t dca_channel_order_native[9] = { 0, 1, 2, 3, 4, 5, 6, 7, 8 };

  1821.             s->channel_order_tab = dca_channel_order_native;

  1822.         }

  1823.     } else {

  1824.         av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n",s->amode);

  1825.         return AVERROR_INVALIDDATA;

  1826.     }

  1827. 

  1828.     if (avctx->channels != channels) {

  1829.         if (avctx->channels)

  1830.             av_log(avctx, AV_LOG_INFO, "Number of channels changed in DCA decoder (%d -> %d)\n", avctx->channels, channels);

  1831.         avctx->channels = channels;

  1832.     }

  1833. 

  1834.     /* get output buffer */

  1835.     s->frame.nb_samples = 256 * (s->sample_blocks / 8);

  1836.     if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {

  1837.         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");

  1838.         return ret;

  1839.     }

  1840.     samples_flt = (float   *)s->frame.data[0];

  1841.     samples_s16 = (int16_t *)s->frame.data[0];

  1842. 

  1843.     /* filter to get final output */

  1844.     for (i = 0; i < (s->sample_blocks / 8); i++) {

  1845.         dca_filter_channels(s, i);

  1846. 

  1847.         /* If this was marked as a DTS-ES stream we need to subtract back- */

  1848.         /* channel from SL & SR to remove matrixed back-channel signal */

  1849.         if((s->source_pcm_res & 1) && s->xch_present) {

  1850.             float* back_chan = s->samples + s->channel_order_tab[s->xch_base_channel] * 256;

  1851.             float* lt_chan   = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256;

  1852.             float* rt_chan   = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;

  1853.             s->dsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);

  1854.             s->dsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);

  1855.         }

  1856. 

  1857.         if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {

  1858.             s->fmt_conv.float_interleave(samples_flt, s->samples_chanptr, 256,

  1859.                                          channels);

  1860.             samples_flt += 256 * channels;

  1861.         } else {

  1862.             s->fmt_conv.float_to_int16_interleave(samples_s16,

  1863.                                                   s->samples_chanptr, 256,

  1864.                                                   channels);

  1865.             samples_s16 += 256 * channels;

  1866.         }

  1867.     }

  1868. 

  1869.     /* update lfe history */

  1870.     lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);

  1871.     for (i = 0; i < 2 * s->lfe * 4; i++) {

  1872.         s->lfe_data[i] = s->lfe_data[i + lfe_samples];

  1873.     }

  1874. 

  1875.     *got_frame_ptr   = 1;

  1876.     *(AVFrame *)data = s->frame;

  1877. 

  1878.     return buf_size;

  1879. }

  1880. 

  1881. 

  1882. 

  1883. /**

  1884.  * DCA initialization

  1885.  *

  1886.  * @param avctx     pointer to the AVCodecContext

  1887.  */

  1888. 

  1889. static av_cold int dca_decode_init(AVCodecContext * avctx)

  1890. {

  1891.     DCAContext *s = avctx->priv_data;

  1892.     int i;

  1893. 

  1894.     s->avctx = avctx;

  1895.     dca_init_vlcs();

  1896. 

  1897.     dsputil_init(&s->dsp, avctx);

  1898.     ff_mdct_init(&s->imdct, 6, 1, 1.0);

  1899.     ff_synth_filter_init(&s->synth);

  1900.     ff_dcadsp_init(&s->dcadsp);

  1901.     ff_fmt_convert_init(&s->fmt_conv, avctx);

  1902. 

  1903.     for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)

  1904.         s->samples_chanptr[i] = s->samples + i * 256;

  1905. 

  1906.     if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {

  1907.         avctx->sample_fmt = AV_SAMPLE_FMT_FLT;

  1908.         s->scale_bias = 1.0 / 32768.0;

  1909.     } else {

  1910.         avctx->sample_fmt = AV_SAMPLE_FMT_S16;

  1911.         s->scale_bias = 1.0;

  1912.     }

  1913. 

  1914.     /* allow downmixing to stereo */

  1915.     if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&

  1916.         avctx->request_channels == 2) {

  1917.         avctx->channels = avctx->request_channels;

  1918.     }

  1919. 

  1920.     avcodec_get_frame_defaults(&s->frame);

  1921.     avctx->coded_frame = &s->frame;

  1922. 

  1923.     return 0;

  1924. }

  1925. 

  1926. static av_cold int dca_decode_end(AVCodecContext * avctx)

  1927. {

  1928.     DCAContext *s = avctx->priv_data;

  1929.     ff_mdct_end(&s->imdct);

  1930.     return 0;

  1931. }

  1932. 

  1933. static const AVProfile profiles[] = {

  1934.     { FF_PROFILE_DTS,        "DTS"        },

  1935.     { FF_PROFILE_DTS_ES,     "DTS-ES"     },

  1936.     { FF_PROFILE_DTS_96_24,  "DTS 96/24"  },

  1937.     { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },

  1938.     { FF_PROFILE_DTS_HD_MA,  "DTS-HD MA"  },

  1939.     { FF_PROFILE_UNKNOWN },

  1940. };

  1941. 

  1942. AVCodec ff_dca_decoder = {

  1943.     .name = "dca",

  1944.     .type = AVMEDIA_TYPE_AUDIO,

  1945.     .id = CODEC_ID_DTS,

  1946.     .priv_data_size = sizeof(DCAContext),

  1947.     .init = dca_decode_init,

  1948.     .decode = dca_decode_frame,

  1949.     .close = dca_decode_end,

  1950.     .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),

  1951.     .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,

  1952.     .sample_fmts = (const enum AVSampleFormat[]) {

  1953.         AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE

  1954.     },

  1955.     .profiles = NULL_IF_CONFIG_SMALL(profiles),

  1956. };