/*
 * DCA compatible decoder
 * Copyright (C) 2004 Gildas Bazin
 * Copyright (C) 2004 Benjamin Zores
 * Copyright (C) 2006 Benjamin Larsson
 * Copyright (C) 2007 Konstantin Shishkov
 *
 * This file is part of Libav.
 *
 * Libav is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * Libav is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with Libav; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include <math.h>
#include <stddef.h>
#include <stdio.h>

#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "libavutil/internal.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/mathematics.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "avcodec.h"
#include "fft.h"
#include "get_bits.h"
#include "put_bits.h"
#include "dcadata.h"
#include "dcahuff.h"
#include "dca.h"
#include "mathops.h"
#include "synth_filter.h"
#include "dcadsp.h"
#include "fmtconvert.h"
#include "internal.h"

#if ARCH_ARM
#   include "arm/dca.h"
#endif
#if ARCH_X86
#   include "x86/dca.h"
#endif

//#define TRACE

#define DCA_PRIM_CHANNELS_MAX  (7)
#define DCA_SUBBANDS          (32)
#define DCA_ABITS_MAX         (32)      /* Should be 28 */
#define DCA_SUBSUBFRAMES_MAX   (4)
#define DCA_SUBFRAMES_MAX     (16)
#define DCA_BLOCKS_MAX        (16)
#define DCA_LFE_MAX            (3)

enum DCAMode {
    DCA_MONO = 0,
    DCA_CHANNEL,
    DCA_STEREO,
    DCA_STEREO_SUMDIFF,
    DCA_STEREO_TOTAL,
    DCA_3F,
    DCA_2F1R,
    DCA_3F1R,
    DCA_2F2R,
    DCA_3F2R,
    DCA_4F2R
};

/* these are unconfirmed but should be mostly correct */
enum DCAExSSSpeakerMask {
    DCA_EXSS_FRONT_CENTER          = 0x0001,
    DCA_EXSS_FRONT_LEFT_RIGHT      = 0x0002,
    DCA_EXSS_SIDE_REAR_LEFT_RIGHT  = 0x0004,
    DCA_EXSS_LFE                   = 0x0008,
    DCA_EXSS_REAR_CENTER           = 0x0010,
    DCA_EXSS_FRONT_HIGH_LEFT_RIGHT = 0x0020,
    DCA_EXSS_REAR_LEFT_RIGHT       = 0x0040,
    DCA_EXSS_FRONT_HIGH_CENTER     = 0x0080,
    DCA_EXSS_OVERHEAD              = 0x0100,
    DCA_EXSS_CENTER_LEFT_RIGHT     = 0x0200,
    DCA_EXSS_WIDE_LEFT_RIGHT       = 0x0400,
    DCA_EXSS_SIDE_LEFT_RIGHT       = 0x0800,
    DCA_EXSS_LFE2                  = 0x1000,
    DCA_EXSS_SIDE_HIGH_LEFT_RIGHT  = 0x2000,
    DCA_EXSS_REAR_HIGH_CENTER      = 0x4000,
    DCA_EXSS_REAR_HIGH_LEFT_RIGHT  = 0x8000,
};

enum DCAExtensionMask {
    DCA_EXT_CORE       = 0x001, ///< core in core substream
    DCA_EXT_XXCH       = 0x002, ///< XXCh channels extension in core substream
    DCA_EXT_X96        = 0x004, ///< 96/24 extension in core substream
    DCA_EXT_XCH        = 0x008, ///< XCh channel extension in core substream
    DCA_EXT_EXSS_CORE  = 0x010, ///< core in ExSS (extension substream)
    DCA_EXT_EXSS_XBR   = 0x020, ///< extended bitrate extension in ExSS
    DCA_EXT_EXSS_XXCH  = 0x040, ///< XXCh channels extension in ExSS
    DCA_EXT_EXSS_X96   = 0x080, ///< 96/24 extension in ExSS
    DCA_EXT_EXSS_LBR   = 0x100, ///< low bitrate component in ExSS
    DCA_EXT_EXSS_XLL   = 0x200, ///< lossless extension in ExSS
};

/* -1 are reserved or unknown */
static const int dca_ext_audio_descr_mask[] = {
    DCA_EXT_XCH,
    -1,
    DCA_EXT_X96,
    DCA_EXT_XCH | DCA_EXT_X96,
    -1,
    -1,
    DCA_EXT_XXCH,
    -1,
};

/* extensions that reside in core substream */
#define DCA_CORE_EXTS (DCA_EXT_XCH | DCA_EXT_XXCH | DCA_EXT_X96)

/* Tables for mapping dts channel configurations to libavcodec multichannel api.
 * Some compromises have been made for special configurations. Most configurations
 * are never used so complete accuracy is not needed.
 *
 * L = left, R = right, C = center, S = surround, F = front, R = rear, T = total, OV = overhead.
 * S  -> side, when both rear and back are configured move one of them to the side channel
 * OV -> center back
 * All 2 channel configurations -> AV_CH_LAYOUT_STEREO
 */
static const uint64_t dca_core_channel_layout[] = {
    AV_CH_FRONT_CENTER,                                                     ///< 1, A
    AV_CH_LAYOUT_STEREO,                                                    ///< 2, A + B (dual mono)
    AV_CH_LAYOUT_STEREO,                                                    ///< 2, L + R (stereo)
    AV_CH_LAYOUT_STEREO,                                                    ///< 2, (L + R) + (L - R) (sum-difference)
    AV_CH_LAYOUT_STEREO,                                                    ///< 2, LT + RT (left and right total)
    AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER,                               ///< 3, C + L + R
    AV_CH_LAYOUT_STEREO | AV_CH_BACK_CENTER,                                ///< 3, L + R + S
    AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER,           ///< 4, C + L + R + S
    AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT,               ///< 4, L + R + SL + SR

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

    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

    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

    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

    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

    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

    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
};

static const int8_t dca_lfe_index[] = {
    1, 2, 2, 2, 2, 3, 2, 3, 2, 3, 2, 3, 1, 3, 2, 3
};

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

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

static const int8_t dca_channel_reorder_nolfe[][9] = {
    { 0, -1, -1, -1, -1, -1, -1, -1, -1},
    { 0,  1, -1, -1, -1, -1, -1, -1, -1},
    { 0,  1, -1, -1, -1, -1, -1, -1, -1},
    { 0,  1, -1, -1, -1, -1, -1, -1, -1},
    { 0,  1, -1, -1, -1, -1, -1, -1, -1},
    { 2,  0,  1, -1, -1, -1, -1, -1, -1},
    { 0,  1,  2, -1, -1, -1, -1, -1, -1},
    { 2,  0,  1,  3, -1, -1, -1, -1, -1},
    { 0,  1,  2,  3, -1, -1, -1, -1, -1},
    { 2,  0,  1,  3,  4, -1, -1, -1, -1},
    { 2,  3,  0,  1,  4,  5, -1, -1, -1},
    { 2,  0,  1,  3,  4,  5, -1, -1, -1},
    { 0,  5,  3,  4,  1,  2, -1, -1, -1},
    { 3,  2,  4,  0,  1,  5,  6, -1, -1},
    { 4,  5,  0,  1,  6,  2,  7,  3, -1},
    { 3,  2,  4,  0,  1,  5,  7,  6, -1},
};

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

#define DCA_DOLBY                  101           /* FIXME */

#define DCA_CHANNEL_BITS             6
#define DCA_CHANNEL_MASK          0x3F

#define DCA_LFE                   0x80

#define HEADER_SIZE                 14

#define DCA_MAX_FRAME_SIZE       16384
#define DCA_MAX_EXSS_HEADER_SIZE  4096

#define DCA_BUFFER_PADDING_SIZE   1024

#define DCA_NSYNCAUX        0x9A1105A0

/** Bit allocation */
typedef struct {
    int offset;                 ///< code values offset
    int maxbits[8];             ///< max bits in VLC
    int wrap;                   ///< wrap for get_vlc2()
    VLC vlc[8];                 ///< actual codes
} BitAlloc;

static BitAlloc dca_bitalloc_index;    ///< indexes for samples VLC select
static BitAlloc dca_tmode;             ///< transition mode VLCs
static BitAlloc dca_scalefactor;       ///< scalefactor VLCs
static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs

static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba,
                                         int idx)
{
    return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) +
           ba->offset;
}

typedef struct {
    AVClass *class;             ///< class for AVOptions
    AVCodecContext *avctx;
    /* Frame header */
    int frame_type;             ///< type of the current frame
    int samples_deficit;        ///< deficit sample count
    int crc_present;            ///< crc is present in the bitstream
    int sample_blocks;          ///< number of PCM sample blocks
    int frame_size;             ///< primary frame byte size
    int amode;                  ///< audio channels arrangement
    int sample_rate;            ///< audio sampling rate
    int bit_rate;               ///< transmission bit rate
    int bit_rate_index;         ///< transmission bit rate index

    int dynrange;               ///< embedded dynamic range flag
    int timestamp;              ///< embedded time stamp flag
    int aux_data;               ///< auxiliary data flag
    int hdcd;                   ///< source material is mastered in HDCD
    int ext_descr;              ///< extension audio descriptor flag
    int ext_coding;             ///< extended coding flag
    int aspf;                   ///< audio sync word insertion flag
    int lfe;                    ///< low frequency effects flag
    int predictor_history;      ///< predictor history flag
    int header_crc;             ///< header crc check bytes
    int multirate_inter;        ///< multirate interpolator switch
    int version;                ///< encoder software revision
    int copy_history;           ///< copy history
    int source_pcm_res;         ///< source pcm resolution
    int front_sum;              ///< front sum/difference flag
    int surround_sum;           ///< surround sum/difference flag
    int dialog_norm;            ///< dialog normalisation parameter

    /* Primary audio coding header */
    int subframes;              ///< number of subframes
    int total_channels;         ///< number of channels including extensions
    int prim_channels;          ///< number of primary audio channels
    int subband_activity[DCA_PRIM_CHANNELS_MAX];    ///< subband activity count
    int vq_start_subband[DCA_PRIM_CHANNELS_MAX];    ///< high frequency vq start subband
    int joint_intensity[DCA_PRIM_CHANNELS_MAX];     ///< joint intensity coding index
    int transient_huffman[DCA_PRIM_CHANNELS_MAX];   ///< transient mode code book
    int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book
    int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX];    ///< bit allocation quantizer select
    int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select
    float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX];   ///< scale factor adjustment

    /* Primary audio coding side information */
    int subsubframes[DCA_SUBFRAMES_MAX];                         ///< number of subsubframes
    int partial_samples[DCA_SUBFRAMES_MAX];                      ///< partial subsubframe samples count
    int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];    ///< prediction mode (ADPCM used or not)
    int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];      ///< prediction VQ coefs
    int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];           ///< bit allocation index
    int transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];    ///< transition mode (transients)
    int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2];    ///< scale factors (2 if transient)
    int joint_huff[DCA_PRIM_CHANNELS_MAX];                       ///< joint subband scale factors codebook
    int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors
    float downmix_coef[DCA_PRIM_CHANNELS_MAX + 1][2];            ///< stereo downmix coefficients
    int dynrange_coef;                                           ///< dynamic range coefficient

    /* Core substream's embedded downmix coefficients (cf. ETSI TS 102 114 V1.4.1)
     * Input:  primary audio channels (incl. LFE if present)
     * Output: downmix audio channels (up to 4, no LFE) */
    uint8_t  core_downmix;                                       ///< embedded downmix coefficients available
    uint8_t  core_downmix_amode;                                 ///< audio channel arrangement of embedded downmix
    uint16_t core_downmix_codes[DCA_PRIM_CHANNELS_MAX + 1][4];   ///< embedded downmix coefficients (9-bit codes)

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

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

    /* Subband samples history (for ADPCM) */
    DECLARE_ALIGNED(16, float, subband_samples_hist)[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
    DECLARE_ALIGNED(32, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512];
    DECLARE_ALIGNED(32, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32];
    int hist_index[DCA_PRIM_CHANNELS_MAX];
    DECLARE_ALIGNED(32, float, raXin)[32];

    int output;                 ///< type of output

    DECLARE_ALIGNED(32, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
    float *samples_chanptr[DCA_PRIM_CHANNELS_MAX + 1];
    float *extra_channels[DCA_PRIM_CHANNELS_MAX + 1];
    uint8_t *extra_channels_buffer;
    unsigned int extra_channels_buffer_size;

    uint8_t dca_buffer[DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE + DCA_BUFFER_PADDING_SIZE];
    int dca_buffer_size;        ///< how much data is in the dca_buffer

    const int8_t *channel_order_tab;  ///< channel reordering table, lfe and non lfe
    GetBitContext gb;
    /* Current position in DCA frame */
    int current_subframe;
    int current_subsubframe;

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

    /* XCh extension information */
    int xch_present;            ///< XCh extension present and valid
    int xch_base_channel;       ///< index of first (only) channel containing XCH data
    int xch_disable;            ///< whether the XCh extension should be decoded or not

    /* ExSS header parser */
    int static_fields;          ///< static fields present
    int mix_metadata;           ///< mixing metadata present
    int num_mix_configs;        ///< number of mix out configurations
    int mix_config_num_ch[4];   ///< number of channels in each mix out configuration

    int profile;

    int debug_flag;             ///< used for suppressing repeated error messages output
    AVFloatDSPContext fdsp;
    FFTContext imdct;
    SynthFilterContext synth;
    DCADSPContext dcadsp;
    FmtConvertContext fmt_conv;
} DCAContext;

static const uint16_t dca_vlc_offs[] = {
        0,   512,   640,   768,  1282,  1794,  2436,  3080,  3770,  4454,  5364,
     5372,  5380,  5388,  5392,  5396,  5412,  5420,  5428,  5460,  5492,  5508,
     5572,  5604,  5668,  5796,  5860,  5892,  6412,  6668,  6796,  7308,  7564,
     7820,  8076,  8620,  9132,  9388,  9910, 10166, 10680, 11196, 11726, 12240,
    12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264,
    18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622,
};

static av_cold void dca_init_vlcs(void)
{
    static int vlcs_initialized = 0;
    int i, j, c = 14;
    static VLC_TYPE dca_table[23622][2];

    if (vlcs_initialized)
        return;

    dca_bitalloc_index.offset = 1;
    dca_bitalloc_index.wrap = 2;
    for (i = 0; i < 5; i++) {
        dca_bitalloc_index.vlc[i].table = &dca_table[dca_vlc_offs[i]];
        dca_bitalloc_index.vlc[i].table_allocated = dca_vlc_offs[i + 1] - dca_vlc_offs[i];
        init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,
                 bitalloc_12_bits[i], 1, 1,
                 bitalloc_12_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
    }
    dca_scalefactor.offset = -64;
    dca_scalefactor.wrap = 2;
    for (i = 0; i < 5; i++) {
        dca_scalefactor.vlc[i].table = &dca_table[dca_vlc_offs[i + 5]];
        dca_scalefactor.vlc[i].table_allocated = dca_vlc_offs[i + 6] - dca_vlc_offs[i + 5];
        init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
                 scales_bits[i], 1, 1,
                 scales_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
    }
    dca_tmode.offset = 0;
    dca_tmode.wrap = 1;
    for (i = 0; i < 4; i++) {
        dca_tmode.vlc[i].table = &dca_table[dca_vlc_offs[i + 10]];
        dca_tmode.vlc[i].table_allocated = dca_vlc_offs[i + 11] - dca_vlc_offs[i + 10];
        init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
                 tmode_bits[i], 1, 1,
                 tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
    }

    for (i = 0; i < 10; i++)
        for (j = 0; j < 7; j++) {
            if (!bitalloc_codes[i][j])
                break;
            dca_smpl_bitalloc[i + 1].offset                 = bitalloc_offsets[i];
            dca_smpl_bitalloc[i + 1].wrap                   = 1 + (j > 4);
            dca_smpl_bitalloc[i + 1].vlc[j].table           = &dca_table[dca_vlc_offs[c]];
            dca_smpl_bitalloc[i + 1].vlc[j].table_allocated = dca_vlc_offs[c + 1] - dca_vlc_offs[c];

            init_vlc(&dca_smpl_bitalloc[i + 1].vlc[j], bitalloc_maxbits[i][j],
                     bitalloc_sizes[i],
                     bitalloc_bits[i][j], 1, 1,
                     bitalloc_codes[i][j], 2, 2, INIT_VLC_USE_NEW_STATIC);
            c++;
        }
    vlcs_initialized = 1;
}

static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
{
    while (len--)
        *dst++ = get_bits(gb, bits);
}

static int dca_parse_audio_coding_header(DCAContext *s, int base_channel)
{
    int i, j;
    static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
    static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
    static const int thr[11]    = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };

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

    if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)
        s->prim_channels = DCA_PRIM_CHANNELS_MAX;


    for (i = base_channel; i < s->prim_channels; i++) {
        s->subband_activity[i] = get_bits(&s->gb, 5) + 2;
        if (s->subband_activity[i] > DCA_SUBBANDS)
            s->subband_activity[i] = DCA_SUBBANDS;
    }
    for (i = base_channel; i < s->prim_channels; i++) {
        s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
        if (s->vq_start_subband[i] > DCA_SUBBANDS)
            s->vq_start_subband[i] = DCA_SUBBANDS;
    }
    get_array(&s->gb, s->joint_intensity + base_channel,     s->prim_channels - base_channel, 3);
    get_array(&s->gb, s->transient_huffman + base_channel,   s->prim_channels - base_channel, 2);
    get_array(&s->gb, s->scalefactor_huffman + base_channel, s->prim_channels - base_channel, 3);
    get_array(&s->gb, s->bitalloc_huffman + base_channel,    s->prim_channels - base_channel, 3);

    /* Get codebooks quantization indexes */
    if (!base_channel)
        memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
    for (j = 1; j < 11; j++)
        for (i = base_channel; i < s->prim_channels; i++)
            s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);

    /* Get scale factor adjustment */
    for (j = 0; j < 11; j++)
        for (i = base_channel; i < s->prim_channels; i++)
            s->scalefactor_adj[i][j] = 1;

    for (j = 1; j < 11; j++)
        for (i = base_channel; i < s->prim_channels; i++)
            if (s->quant_index_huffman[i][j] < thr[j])
                s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];

    if (s->crc_present) {
        /* Audio header CRC check */
        get_bits(&s->gb, 16);
    }

    s->current_subframe    = 0;
    s->current_subsubframe = 0;

#ifdef TRACE
    av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes);
    av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels);
    for (i = base_channel; i < s->prim_channels; i++) {
        av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n",
               s->subband_activity[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n",
               s->vq_start_subband[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n",
               s->joint_intensity[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n",
               s->transient_huffman[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n",
               s->scalefactor_huffman[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n",
               s->bitalloc_huffman[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:");
        for (j = 0; j < 11; j++)
            av_log(s->avctx, AV_LOG_DEBUG, " %i", s->quant_index_huffman[i][j]);
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
        av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");
        for (j = 0; j < 11; j++)
            av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]);
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
    }
#endif

    return 0;
}

static int dca_parse_frame_header(DCAContext *s)
{
    init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);

    /* Sync code */
    skip_bits_long(&s->gb, 32);

    /* Frame header */
    s->frame_type        = get_bits(&s->gb, 1);
    s->samples_deficit   = get_bits(&s->gb, 5) + 1;
    s->crc_present       = get_bits(&s->gb, 1);
    s->sample_blocks     = get_bits(&s->gb, 7) + 1;
    s->frame_size        = get_bits(&s->gb, 14) + 1;
    if (s->frame_size < 95)
        return AVERROR_INVALIDDATA;
    s->amode             = get_bits(&s->gb, 6);
    s->sample_rate       = avpriv_dca_sample_rates[get_bits(&s->gb, 4)];
    if (!s->sample_rate)
        return AVERROR_INVALIDDATA;
    s->bit_rate_index    = get_bits(&s->gb, 5);
    s->bit_rate          = dca_bit_rates[s->bit_rate_index];
    if (!s->bit_rate)
        return AVERROR_INVALIDDATA;

    skip_bits1(&s->gb); // always 0 (reserved, cf. ETSI TS 102 114 V1.4.1)
    s->dynrange          = get_bits(&s->gb, 1);
    s->timestamp         = get_bits(&s->gb, 1);
    s->aux_data          = get_bits(&s->gb, 1);
    s->hdcd              = get_bits(&s->gb, 1);
    s->ext_descr         = get_bits(&s->gb, 3);
    s->ext_coding        = get_bits(&s->gb, 1);
    s->aspf              = get_bits(&s->gb, 1);
    s->lfe               = get_bits(&s->gb, 2);
    s->predictor_history = get_bits(&s->gb, 1);

    if (s->lfe > 2) {
        av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE value: %d\n", s->lfe);
        return AVERROR_INVALIDDATA;
    }

    /* TODO: check CRC */
    if (s->crc_present)
        s->header_crc    = get_bits(&s->gb, 16);

    s->multirate_inter   = get_bits(&s->gb, 1);
    s->version           = get_bits(&s->gb, 4);
    s->copy_history      = get_bits(&s->gb, 2);
    s->source_pcm_res    = get_bits(&s->gb, 3);
    s->front_sum         = get_bits(&s->gb, 1);
    s->surround_sum      = get_bits(&s->gb, 1);
    s->dialog_norm       = get_bits(&s->gb, 4);

    /* FIXME: channels mixing levels */
    s->output = s->amode;
    if (s->lfe)
        s->output |= DCA_LFE;

#ifdef TRACE
    av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type);
    av_log(s->avctx, AV_LOG_DEBUG, "samples deficit: %i\n", s->samples_deficit);
    av_log(s->avctx, AV_LOG_DEBUG, "crc present: %i\n", s->crc_present);
    av_log(s->avctx, AV_LOG_DEBUG, "sample blocks: %i (%i samples)\n",
           s->sample_blocks, s->sample_blocks * 32);
    av_log(s->avctx, AV_LOG_DEBUG, "frame size: %i bytes\n", s->frame_size);
    av_log(s->avctx, AV_LOG_DEBUG, "amode: %i (%i channels)\n",
           s->amode, dca_channels[s->amode]);
    av_log(s->avctx, AV_LOG_DEBUG, "sample rate: %i Hz\n",
           s->sample_rate);
    av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i bits/s\n",
           s->bit_rate);
    av_log(s->avctx, AV_LOG_DEBUG, "dynrange: %i\n", s->dynrange);
    av_log(s->avctx, AV_LOG_DEBUG, "timestamp: %i\n", s->timestamp);
    av_log(s->avctx, AV_LOG_DEBUG, "aux_data: %i\n", s->aux_data);
    av_log(s->avctx, AV_LOG_DEBUG, "hdcd: %i\n", s->hdcd);
    av_log(s->avctx, AV_LOG_DEBUG, "ext descr: %i\n", s->ext_descr);
    av_log(s->avctx, AV_LOG_DEBUG, "ext coding: %i\n", s->ext_coding);
    av_log(s->avctx, AV_LOG_DEBUG, "aspf: %i\n", s->aspf);
    av_log(s->avctx, AV_LOG_DEBUG, "lfe: %i\n", s->lfe);
    av_log(s->avctx, AV_LOG_DEBUG, "predictor history: %i\n",
           s->predictor_history);
    av_log(s->avctx, AV_LOG_DEBUG, "header crc: %i\n", s->header_crc);
    av_log(s->avctx, AV_LOG_DEBUG, "multirate inter: %i\n",
           s->multirate_inter);
    av_log(s->avctx, AV_LOG_DEBUG, "version number: %i\n", s->version);
    av_log(s->avctx, AV_LOG_DEBUG, "copy history: %i\n", s->copy_history);
    av_log(s->avctx, AV_LOG_DEBUG,
           "source pcm resolution: %i (%i bits/sample)\n",
           s->source_pcm_res, dca_bits_per_sample[s->source_pcm_res]);
    av_log(s->avctx, AV_LOG_DEBUG, "front sum: %i\n", s->front_sum);
    av_log(s->avctx, AV_LOG_DEBUG, "surround sum: %i\n", s->surround_sum);
    av_log(s->avctx, AV_LOG_DEBUG, "dialog norm: %i\n", s->dialog_norm);
    av_log(s->avctx, AV_LOG_DEBUG, "\n");
#endif

    /* Primary audio coding header */
    s->subframes         = get_bits(&s->gb, 4) + 1;

    return dca_parse_audio_coding_header(s, 0);
}


static inline int get_scale(GetBitContext *gb, int level, int value, int log2range)
{
    if (level < 5) {
        /* huffman encoded */
        value += get_bitalloc(gb, &dca_scalefactor, level);
        value = av_clip(value, 0, (1 << log2range) - 1);
    } else if (level < 8) {
        if (level + 1 > log2range) {
            skip_bits(gb, level + 1 - log2range);
            value = get_bits(gb, log2range);
        } else {
            value = get_bits(gb, level + 1);
        }
    }
    return value;
}

static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
{
    /* Primary audio coding side information */
    int j, k;

    if (get_bits_left(&s->gb) < 0)
        return AVERROR_INVALIDDATA;

    if (!base_channel) {
        s->subsubframes[s->current_subframe]    = get_bits(&s->gb, 2) + 1;
        s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3);
    }

    for (j = base_channel; j < s->prim_channels; j++) {
        for (k = 0; k < s->subband_activity[j]; k++)
            s->prediction_mode[j][k] = get_bits(&s->gb, 1);
    }

    /* Get prediction codebook */
    for (j = base_channel; j < s->prim_channels; j++) {
        for (k = 0; k < s->subband_activity[j]; k++) {
            if (s->prediction_mode[j][k] > 0) {
                /* (Prediction coefficient VQ address) */
                s->prediction_vq[j][k] = get_bits(&s->gb, 12);
            }
        }
    }

    /* Bit allocation index */
    for (j = base_channel; j < s->prim_channels; j++) {
        for (k = 0; k < s->vq_start_subband[j]; k++) {
            if (s->bitalloc_huffman[j] == 6)
                s->bitalloc[j][k] = get_bits(&s->gb, 5);
            else if (s->bitalloc_huffman[j] == 5)
                s->bitalloc[j][k] = get_bits(&s->gb, 4);
            else if (s->bitalloc_huffman[j] == 7) {
                av_log(s->avctx, AV_LOG_ERROR,
                       "Invalid bit allocation index\n");
                return AVERROR_INVALIDDATA;
            } else {
                s->bitalloc[j][k] =
                    get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);
            }

            if (s->bitalloc[j][k] > 26) {
                av_dlog(s->avctx, "bitalloc index [%i][%i] too big (%i)\n",
                        j, k, s->bitalloc[j][k]);
                return AVERROR_INVALIDDATA;
            }
        }
    }

    /* Transition mode */
    for (j = base_channel; j < s->prim_channels; j++) {
        for (k = 0; k < s->subband_activity[j]; k++) {
            s->transition_mode[j][k] = 0;
            if (s->subsubframes[s->current_subframe] > 1 &&
                k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) {
                s->transition_mode[j][k] =
                    get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]);
            }
        }
    }

    if (get_bits_left(&s->gb) < 0)
        return AVERROR_INVALIDDATA;

    for (j = base_channel; j < s->prim_channels; j++) {
        const uint32_t *scale_table;
        int scale_sum, log_size;

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

        if (s->scalefactor_huffman[j] == 6) {
            scale_table = scale_factor_quant7;
            log_size = 7;
        } else {
            scale_table = scale_factor_quant6;
            log_size = 6;
        }

        /* When huffman coded, only the difference is encoded */
        scale_sum = 0;

        for (k = 0; k < s->subband_activity[j]; k++) {
            if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) {
                scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size);
                s->scale_factor[j][k][0] = scale_table[scale_sum];
            }

            if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) {
                /* Get second scale factor */
                scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size);
                s->scale_factor[j][k][1] = scale_table[scale_sum];
            }
        }
    }

    /* Joint subband scale factor codebook select */
    for (j = base_channel; j < s->prim_channels; j++) {
        /* Transmitted only if joint subband coding enabled */
        if (s->joint_intensity[j] > 0)
            s->joint_huff[j] = get_bits(&s->gb, 3);
    }

    if (get_bits_left(&s->gb) < 0)
        return AVERROR_INVALIDDATA;

    /* Scale factors for joint subband coding */
    for (j = base_channel; j < s->prim_channels; j++) {
        int source_channel;

        /* Transmitted only if joint subband coding enabled */
        if (s->joint_intensity[j] > 0) {
            int scale = 0;
            source_channel = s->joint_intensity[j] - 1;

            /* When huffman coded, only the difference is encoded
             * (is this valid as well for joint scales ???) */

            for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) {
                scale = get_scale(&s->gb, s->joint_huff[j], 64 /* bias */, 7);
                s->joint_scale_factor[j][k] = scale;    /*joint_scale_table[scale]; */
            }

            if (!(s->debug_flag & 0x02)) {
                av_log(s->avctx, AV_LOG_DEBUG,
                       "Joint stereo coding not supported\n");
                s->debug_flag |= 0x02;
            }
        }
    }

    /* Dynamic range coefficient */
    if (!base_channel && s->dynrange)
        s->dynrange_coef = get_bits(&s->gb, 8);

    /* Side information CRC check word */
    if (s->crc_present) {
        get_bits(&s->gb, 16);
    }

    /*
     * Primary audio data arrays
     */

    /* VQ encoded high frequency subbands */
    for (j = base_channel; j < s->prim_channels; j++)
        for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
            /* 1 vector -> 32 samples */
            s->high_freq_vq[j][k] = get_bits(&s->gb, 10);

    /* Low frequency effect data */
    if (!base_channel && s->lfe) {
        /* LFE samples */
        int lfe_samples = 2 * s->lfe * (4 + block_index);
        int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);
        float lfe_scale;

        for (j = lfe_samples; j < lfe_end_sample; j++) {
            /* Signed 8 bits int */
            s->lfe_data[j] = get_sbits(&s->gb, 8);
        }

        /* Scale factor index */
        skip_bits(&s->gb, 1);
        s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 7)];

        /* Quantization step size * scale factor */
        lfe_scale = 0.035 * s->lfe_scale_factor;

        for (j = lfe_samples; j < lfe_end_sample; j++)
            s->lfe_data[j] *= lfe_scale;
    }

#ifdef TRACE
    av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n",
           s->subsubframes[s->current_subframe]);
    av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n",
           s->partial_samples[s->current_subframe]);

    for (j = base_channel; j < s->prim_channels; j++) {
        av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:");
        for (k = 0; k < s->subband_activity[j]; k++)
            av_log(s->avctx, AV_LOG_DEBUG, " %i", s->prediction_mode[j][k]);
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
    }
    for (j = base_channel; j < s->prim_channels; j++) {
        for (k = 0; k < s->subband_activity[j]; k++)
            av_log(s->avctx, AV_LOG_DEBUG,
                   "prediction coefs: %f, %f, %f, %f\n",
                   (float) adpcm_vb[s->prediction_vq[j][k]][0] / 8192,
                   (float) adpcm_vb[s->prediction_vq[j][k]][1] / 8192,
                   (float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192,
                   (float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192);
    }
    for (j = base_channel; j < s->prim_channels; j++) {
        av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: ");
        for (k = 0; k < s->vq_start_subband[j]; k++)
            av_log(s->avctx, AV_LOG_DEBUG, "%2.2i ", s->bitalloc[j][k]);
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
    }
    for (j = base_channel; j < s->prim_channels; j++) {
        av_log(s->avctx, AV_LOG_DEBUG, "Transition mode:");
        for (k = 0; k < s->subband_activity[j]; k++)
            av_log(s->avctx, AV_LOG_DEBUG, " %i", s->transition_mode[j][k]);
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
    }
    for (j = base_channel; j < s->prim_channels; j++) {
        av_log(s->avctx, AV_LOG_DEBUG, "Scale factor:");
        for (k = 0; k < s->subband_activity[j]; k++) {
            if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0)
                av_log(s->avctx, AV_LOG_DEBUG, " %i", s->scale_factor[j][k][0]);
            if (k < s->vq_start_subband[j] && s->transition_mode[j][k])
                av_log(s->avctx, AV_LOG_DEBUG, " %i(t)", s->scale_factor[j][k][1]);
        }
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
    }
    for (j = base_channel; j < s->prim_channels; j++) {
        if (s->joint_intensity[j] > 0) {
            int source_channel = s->joint_intensity[j] - 1;
            av_log(s->avctx, AV_LOG_DEBUG, "Joint scale factor index:\n");
            for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++)
                av_log(s->avctx, AV_LOG_DEBUG, " %i", s->joint_scale_factor[j][k]);
            av_log(s->avctx, AV_LOG_DEBUG, "\n");
        }
    }
    for (j = base_channel; j < s->prim_channels; j++)
        for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
            av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]);
    if (!base_channel && s->lfe) {
        int lfe_samples = 2 * s->lfe * (4 + block_index);
        int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);

        av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n");
        for (j = lfe_samples; j < lfe_end_sample; j++)
            av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]);
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
    }
#endif

    return 0;
}

static void qmf_32_subbands(DCAContext *s, int chans,
                            float samples_in[32][8], float *samples_out,
                            float scale)
{
    const float *prCoeff;

    int sb_act = s->subband_activity[chans];

    scale *= sqrt(1 / 8.0);

    /* Select filter */
    if (!s->multirate_inter)    /* Non-perfect reconstruction */
        prCoeff = fir_32bands_nonperfect;
    else                        /* Perfect reconstruction */
        prCoeff = fir_32bands_perfect;

    s->dcadsp.qmf_32_subbands(samples_in, sb_act, &s->synth, &s->imdct,
                              s->subband_fir_hist[chans],
                              &s->hist_index[chans],
                              s->subband_fir_noidea[chans], prCoeff,
                              samples_out, s->raXin, scale);
}

static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
                                  int num_deci_sample, float *samples_in,
                                  float *samples_out, float scale)
{
    /* samples_in: An array holding decimated samples.
     *   Samples in current subframe starts from samples_in[0],
     *   while samples_in[-1], samples_in[-2], ..., stores samples
     *   from last subframe as history.
     *
     * samples_out: An array holding interpolated samples
     */

    int idx;
    const float *prCoeff;
    int deciindex;

    /* Select decimation filter */
    if (decimation_select == 1) {
        idx = 1;
        prCoeff = lfe_fir_128;
    } else {
        idx = 0;
        prCoeff = lfe_fir_64;
    }
    /* Interpolation */
    for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
        s->dcadsp.lfe_fir[idx](samples_out, samples_in, prCoeff, scale);
        samples_in++;
        samples_out += 2 * 32 * (1 + idx);
    }
}

/* downmixing routines */
#define MIX_REAR1(samples, s1, rs, coef)            \
    samples[0][i] += samples[s1][i] * coef[rs][0];  \
    samples[1][i] += samples[s1][i] * coef[rs][1];

#define MIX_REAR2(samples, s1, s2, rs, coef)                                          \
    samples[0][i] += samples[s1][i] * coef[rs][0] + samples[s2][i] * coef[rs + 1][0]; \
    samples[1][i] += samples[s1][i] * coef[rs][1] + samples[s2][i] * coef[rs + 1][1];

#define MIX_FRONT3(samples, coef)                                      \
    t = samples[c][i];                                                 \
    u = samples[l][i];                                                 \
    v = samples[r][i];                                                 \
    samples[0][i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0];  \
    samples[1][i] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];

#define DOWNMIX_TO_STEREO(op1, op2)             \
    for (i = 0; i < 256; i++) {                 \
        op1                                     \
        op2                                     \
    }

static void dca_downmix(float **samples, int srcfmt, int lfe_present,
                        float coef[DCA_PRIM_CHANNELS_MAX + 1][2],
                        const int8_t *channel_mapping)
{
    int c, l, r, sl, sr, s;
    int i;
    float t, u, v;

    switch (srcfmt) {
    case DCA_MONO:
    case DCA_4F2R:
        av_log(NULL, 0, "Not implemented!\n");
        break;
    case DCA_CHANNEL:
    case DCA_STEREO:
    case DCA_STEREO_TOTAL:
    case DCA_STEREO_SUMDIFF:
        break;
    case DCA_3F:
        c = channel_mapping[0];
        l = channel_mapping[1];
        r = channel_mapping[2];
        DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );
        break;
    case DCA_2F1R:
        s = channel_mapping[2];
        DOWNMIX_TO_STEREO(MIX_REAR1(samples, s, 2, coef), );
        break;
    case DCA_3F1R:
        c = channel_mapping[0];
        l = channel_mapping[1];
        r = channel_mapping[2];
        s = channel_mapping[3];
        DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
                          MIX_REAR1(samples, s, 3, coef));
        break;
    case DCA_2F2R:
        sl = channel_mapping[2];
        sr = channel_mapping[3];
        DOWNMIX_TO_STEREO(MIX_REAR2(samples, sl, sr, 2, coef), );
        break;
    case DCA_3F2R:
        c  = channel_mapping[0];
        l  = channel_mapping[1];
        r  = channel_mapping[2];
        sl = channel_mapping[3];
        sr = channel_mapping[4];
        DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
                          MIX_REAR2(samples, sl, sr, 3, coef));
        break;
    }
    if (lfe_present) {
        int lf_buf = dca_lfe_index[srcfmt];
        int lf_idx = dca_channels [srcfmt];
        for (i = 0; i < 256; i++) {
            samples[0][i] += samples[lf_buf][i] * coef[lf_idx][0];
            samples[1][i] += samples[lf_buf][i] * coef[lf_idx][1];
        }
    }
}


#ifndef decode_blockcodes
/* Very compact version of the block code decoder that does not use table
 * look-up but is slightly slower */
static int decode_blockcode(int code, int levels, int32_t *values)
{
    int i;
    int offset = (levels - 1) >> 1;

    for (i = 0; i < 4; i++) {
        int div = FASTDIV(code, levels);
        values[i] = code - offset - div * levels;
        code = div;
    }

    return code;
}

static int decode_blockcodes(int code1, int code2, int levels, int32_t *values)
{
    return decode_blockcode(code1, levels, values) |
           decode_blockcode(code2, levels, values + 4);
}
#endif

static const uint8_t abits_sizes[7]  = { 7, 10, 12, 13, 15, 17, 19 };
static const uint8_t abits_levels[7] = { 3,  5,  7,  9, 13, 17, 25 };

#ifndef int8x8_fmul_int32
static inline void int8x8_fmul_int32(DCADSPContext *dsp, float *dst,
                                     const int8_t *src, int scale)
{
    dsp->int8x8_fmul_int32(dst, src, scale);
}
#endif

static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
{
    int k, l;
    int subsubframe = s->current_subsubframe;

    const float *quant_step_table;

    /* FIXME */
    float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
    LOCAL_ALIGNED_16(int32_t, block, [8 * DCA_SUBBANDS]);

    /*
     * Audio data
     */

    /* Select quantization step size table */
    if (s->bit_rate_index == 0x1f)
        quant_step_table = lossless_quant_d;
    else
        quant_step_table = lossy_quant_d;

    for (k = base_channel; k < s->prim_channels; k++) {
        float rscale[DCA_SUBBANDS];

        if (get_bits_left(&s->gb) < 0)
            return AVERROR_INVALIDDATA;

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

            /* Select the mid-tread linear quantizer */
            int abits = s->bitalloc[k][l];

            float quant_step_size = quant_step_table[abits];

            /*
             * Determine quantization index code book and its type
             */

            /* Select quantization index code book */
            int sel = s->quant_index_huffman[k][abits];

            /*
             * Extract bits from the bit stream
             */
            if (!abits) {
                rscale[l] = 0;
                memset(block + 8 * l, 0, 8 * sizeof(block[0]));
            } else {
                /* Deal with transients */
                int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];
                rscale[l] = quant_step_size * s->scale_factor[k][l][sfi] *
                               s->scalefactor_adj[k][sel];

                if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {
                    if (abits <= 7) {
                        /* Block code */
                        int block_code1, block_code2, size, levels, err;

                        size   = abits_sizes[abits - 1];
                        levels = abits_levels[abits - 1];

                        block_code1 = get_bits(&s->gb, size);
                        block_code2 = get_bits(&s->gb, size);
                        err = decode_blockcodes(block_code1, block_code2,
                                                levels, block + 8 * l);
                        if (err) {
                            av_log(s->avctx, AV_LOG_ERROR,
                                   "ERROR: block code look-up failed\n");
                            return AVERROR_INVALIDDATA;
                        }
                    } else {
                        /* no coding */
                        for (m = 0; m < 8; m++)
                            block[8 * l + m] = get_sbits(&s->gb, abits - 3);
                    }
                } else {
                    /* Huffman coded */
                    for (m = 0; m < 8; m++)
                        block[8 * l + m] = get_bitalloc(&s->gb,
                                                &dca_smpl_bitalloc[abits], sel);
                }

            }
        }

        s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[k][0],
                                               block, rscale, 8 * s->vq_start_subband[k]);

        for (l = 0; l < s->vq_start_subband[k]; l++) {
            int m;
            /*
             * Inverse ADPCM if in prediction mode
             */
            if (s->prediction_mode[k][l]) {
                int n;
                for (m = 0; m < 8; m++) {
                    for (n = 1; n <= 4; n++)
                        if (m >= n)
                            subband_samples[k][l][m] +=
                                (adpcm_vb[s->prediction_vq[k][l]][n - 1] *
                                 subband_samples[k][l][m - n] / 8192);
                        else if (s->predictor_history)
                            subband_samples[k][l][m] +=
                                (adpcm_vb[s->prediction_vq[k][l]][n - 1] *
                                 s->subband_samples_hist[k][l][m - n + 4] / 8192);
                }
            }
        }

        /*
         * Decode VQ encoded high frequencies
         */
        for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {
            /* 1 vector -> 32 samples but we only need the 8 samples
             * for this subsubframe. */
            int hfvq = s->high_freq_vq[k][l];

            if (!s->debug_flag & 0x01) {
                av_log(s->avctx, AV_LOG_DEBUG,
                       "Stream with high frequencies VQ coding\n");
                s->debug_flag |= 0x01;
            }

            int8x8_fmul_int32(&s->dcadsp, subband_samples[k][l],
                              &high_freq_vq[hfvq][subsubframe * 8],
                              s->scale_factor[k][l][0]);
        }
    }

    /* Check for DSYNC after subsubframe */
    if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {
        if (0xFFFF == get_bits(&s->gb, 16)) {   /* 0xFFFF */
#ifdef TRACE
            av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");
#endif
        } else {
            av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
            return AVERROR_INVALIDDATA;
        }
    }

    /* Backup predictor history for adpcm */
    for (k = base_channel; k < s->prim_channels; k++)
        for (l = 0; l < s->vq_start_subband[k]; l++)
            AV_COPY128(s->subband_samples_hist[k][l], &subband_samples[k][l][4]);

    return 0;
}

static int dca_filter_channels(DCAContext *s, int block_index)
{
    float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
    int k;

    /* 32 subbands QMF */
    for (k = 0; k < s->prim_channels; k++) {
/*        static float pcm_to_double[8] = { 32768.0, 32768.0, 524288.0, 524288.0,
                                            0, 8388608.0, 8388608.0 };*/
        if (s->channel_order_tab[k] >= 0)
            qmf_32_subbands(s, k, subband_samples[k],
                            s->samples_chanptr[s->channel_order_tab[k]],
                            M_SQRT1_2 / 32768.0 /* pcm_to_double[s->source_pcm_res] */);
    }

    /* Generate LFE samples for this subsubframe FIXME!!! */
    if (s->lfe) {
        lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,
                              s->lfe_data + 2 * s->lfe * (block_index + 4),
                              s->samples_chanptr[dca_lfe_index[s->amode]],
                              1.0 / (256.0 * 32768.0));
        /* Outputs 20bits pcm samples */
    }

    /* Downmixing to Stereo */
    if (s->prim_channels + !!s->lfe > 2 &&
        s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
        dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef,
                    s->channel_order_tab);
    }

    return 0;
}


static int dca_subframe_footer(DCAContext *s, int base_channel)
{
    int in, out, aux_data_count, aux_data_end, reserved;
    uint32_t nsyncaux;

    /*
     * Unpack optional information
     */

    /* presumably optional information only appears in the core? */
    if (!base_channel) {
        if (s->timestamp)
            skip_bits_long(&s->gb, 32);

        if (s->aux_data) {
            aux_data_count = get_bits(&s->gb, 6);

            // align (32-bit)
            skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

            aux_data_end = 8 * aux_data_count + get_bits_count(&s->gb);

            if ((nsyncaux = get_bits_long(&s->gb, 32)) != DCA_NSYNCAUX) {
                av_log(s->avctx, AV_LOG_ERROR, "nSYNCAUX mismatch %#"PRIx32"\n",
                       nsyncaux);
                return AVERROR_INVALIDDATA;
            }

            if (get_bits1(&s->gb)) { // bAUXTimeStampFlag
                avpriv_request_sample(s->avctx,
                                      "Auxiliary Decode Time Stamp Flag");
                // align (4-bit)
                skip_bits(&s->gb, (-get_bits_count(&s->gb)) & 4);
                // 44 bits: nMSByte (8), nMarker (4), nLSByte (28), nMarker (4)
                skip_bits_long(&s->gb, 44);
            }

            if ((s->core_downmix = get_bits1(&s->gb))) {
                int am = get_bits(&s->gb, 3);
                switch (am) {
                case 0:
                    s->core_downmix_amode = DCA_MONO;
                    break;
                case 1:
                    s->core_downmix_amode = DCA_STEREO;
                    break;
                case 2:
                    s->core_downmix_amode = DCA_STEREO_TOTAL;
                    break;
                case 3:
                    s->core_downmix_amode = DCA_3F;
                    break;
                case 4:
                    s->core_downmix_amode = DCA_2F1R;
                    break;
                case 5:
                    s->core_downmix_amode = DCA_2F2R;
                    break;
                case 6:
                    s->core_downmix_amode = DCA_3F1R;
                    break;
                default:
                    av_log(s->avctx, AV_LOG_ERROR,
                           "Invalid mode %d for embedded downmix coefficients\n",
                           am);
                    return AVERROR_INVALIDDATA;
                }
                for (out = 0; out < dca_channels[s->core_downmix_amode]; out++) {
                    for (in = 0; in < s->prim_channels + !!s->lfe; in++) {
                        uint16_t tmp = get_bits(&s->gb, 9);
                        if ((tmp & 0xFF) > 241) {
                            av_log(s->avctx, AV_LOG_ERROR,
                                   "Invalid downmix coefficient code %"PRIu16"\n",
                                   tmp);
                            return AVERROR_INVALIDDATA;
                        }
                        s->core_downmix_codes[in][out] = tmp;
                    }
                }
            }

            align_get_bits(&s->gb); // byte align
            skip_bits(&s->gb, 16);  // nAUXCRC16

            // additional data (reserved, cf. ETSI TS 102 114 V1.4.1)
            if ((reserved = (aux_data_end - get_bits_count(&s->gb))) < 0) {
                 av_log(s->avctx, AV_LOG_ERROR,
                        "Overread auxiliary data by %d bits\n", -reserved);
                return AVERROR_INVALIDDATA;
            } else if (reserved) {
                avpriv_request_sample(s->avctx,
                                      "Core auxiliary data reserved content");
                skip_bits_long(&s->gb, reserved);
            }
        }

        if (s->crc_present && s->dynrange)
            get_bits(&s->gb, 16);
    }

    return 0;
}

/**
 * Decode a dca frame block
 *
 * @param s     pointer to the DCAContext
 */

static int dca_decode_block(DCAContext *s, int base_channel, int block_index)
{
    int ret;

    /* Sanity check */
    if (s->current_subframe >= s->subframes) {
        av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
               s->current_subframe, s->subframes);
        return AVERROR_INVALIDDATA;
    }

    if (!s->current_subsubframe) {
#ifdef TRACE
        av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
#endif
        /* Read subframe header */
        if ((ret = dca_subframe_header(s, base_channel, block_index)))
            return ret;
    }

    /* Read subsubframe */
#ifdef TRACE
    av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");
#endif
    if ((ret = dca_subsubframe(s, base_channel, block_index)))
        return ret;

    /* Update state */
    s->current_subsubframe++;
    if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {
        s->current_subsubframe = 0;
        s->current_subframe++;
    }
    if (s->current_subframe >= s->subframes) {
#ifdef TRACE
        av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");
#endif
        /* Read subframe footer */
        if ((ret = dca_subframe_footer(s, base_channel)))
            return ret;
    }

    return 0;
}

/**
 * Return the number of channels in an ExSS speaker mask (HD)
 */
static int dca_exss_mask2count(int mask)
{
    /* count bits that mean speaker pairs twice */
    return av_popcount(mask) +
           av_popcount(mask & (DCA_EXSS_CENTER_LEFT_RIGHT      |
                               DCA_EXSS_FRONT_LEFT_RIGHT       |
                               DCA_EXSS_FRONT_HIGH_LEFT_RIGHT  |
                               DCA_EXSS_WIDE_LEFT_RIGHT        |
                               DCA_EXSS_SIDE_LEFT_RIGHT        |
                               DCA_EXSS_SIDE_HIGH_LEFT_RIGHT   |
                               DCA_EXSS_SIDE_REAR_LEFT_RIGHT   |
                               DCA_EXSS_REAR_LEFT_RIGHT        |
                               DCA_EXSS_REAR_HIGH_LEFT_RIGHT));
}

/**
 * Skip mixing coefficients of a single mix out configuration (HD)
 */
static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch)
{
    int i;

    for (i = 0; i < channels; i++) {
        int mix_map_mask = get_bits(gb, out_ch);
        int num_coeffs = av_popcount(mix_map_mask);
        skip_bits_long(gb, num_coeffs * 6);
    }
}

/**
 * Parse extension substream asset header (HD)
 */
static int dca_exss_parse_asset_header(DCAContext *s)
{
    int header_pos = get_bits_count(&s->gb);
    int header_size;
    int channels;
    int embedded_stereo = 0;
    int embedded_6ch    = 0;
    int drc_code_present;
    int extensions_mask;
    int i, j;

    if (get_bits_left(&s->gb) < 16)
        return -1;

    /* We will parse just enough to get to the extensions bitmask with which
     * we can set the profile value. */

    header_size = get_bits(&s->gb, 9) + 1;
    skip_bits(&s->gb, 3); // asset index

    if (s->static_fields) {
        if (get_bits1(&s->gb))
            skip_bits(&s->gb, 4); // asset type descriptor
        if (get_bits1(&s->gb))
            skip_bits_long(&s->gb, 24); // language descriptor

        if (get_bits1(&s->gb)) {
            /* How can one fit 1024 bytes of text here if the maximum value
             * for the asset header size field above was 512 bytes? */
            int text_length = get_bits(&s->gb, 10) + 1;
            if (get_bits_left(&s->gb) < text_length * 8)
                return -1;
            skip_bits_long(&s->gb, text_length * 8); // info text
        }

        skip_bits(&s->gb, 5); // bit resolution - 1
        skip_bits(&s->gb, 4); // max sample rate code
        channels = get_bits(&s->gb, 8) + 1;

        if (get_bits1(&s->gb)) { // 1-to-1 channels to speakers
            int spkr_remap_sets;
            int spkr_mask_size = 16;
            int num_spkrs[7];

            if (channels > 2)
                embedded_stereo = get_bits1(&s->gb);
            if (channels > 6)
                embedded_6ch = get_bits1(&s->gb);

            if (get_bits1(&s->gb)) {
                spkr_mask_size = (get_bits(&s->gb, 2) + 1) << 2;
                skip_bits(&s->gb, spkr_mask_size); // spkr activity mask
            }

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

            for (i = 0; i < spkr_remap_sets; i++) {
                /* std layout mask for each remap set */
                num_spkrs[i] = dca_exss_mask2count(get_bits(&s->gb, spkr_mask_size));
            }

            for (i = 0; i < spkr_remap_sets; i++) {
                int num_dec_ch_remaps = get_bits(&s->gb, 5) + 1;
                if (get_bits_left(&s->gb) < 0)
                    return -1;

                for (j = 0; j < num_spkrs[i]; j++) {
                    int remap_dec_ch_mask = get_bits_long(&s->gb, num_dec_ch_remaps);
                    int num_dec_ch = av_popcount(remap_dec_ch_mask);
                    skip_bits_long(&s->gb, num_dec_ch * 5); // remap codes
                }
            }

        } else {
            skip_bits(&s->gb, 3); // representation type
        }
    }

    drc_code_present = get_bits1(&s->gb);
    if (drc_code_present)
        get_bits(&s->gb, 8); // drc code

    if (get_bits1(&s->gb))
        skip_bits(&s->gb, 5); // dialog normalization code

    if (drc_code_present && embedded_stereo)
        get_bits(&s->gb, 8); // drc stereo code

    if (s->mix_metadata && get_bits1(&s->gb)) {
        skip_bits(&s->gb, 1); // external mix
        skip_bits(&s->gb, 6); // post mix gain code

        if (get_bits(&s->gb, 2) != 3) // mixer drc code
            skip_bits(&s->gb, 3); // drc limit
        else
            skip_bits(&s->gb, 8); // custom drc code

        if (get_bits1(&s->gb)) // channel specific scaling
            for (i = 0; i < s->num_mix_configs; i++)
                skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6); // scale codes
        else
            skip_bits_long(&s->gb, s->num_mix_configs * 6); // scale codes

        for (i = 0; i < s->num_mix_configs; i++) {
            if (get_bits_left(&s->gb) < 0)
                return -1;
            dca_exss_skip_mix_coeffs(&s->gb, channels, s->mix_config_num_ch[i]);
            if (embedded_6ch)
                dca_exss_skip_mix_coeffs(&s->gb, 6, s->mix_config_num_ch[i]);
            if (embedded_stereo)
                dca_exss_skip_mix_coeffs(&s->gb, 2, s->mix_config_num_ch[i]);
        }
    }

    switch (get_bits(&s->gb, 2)) {
    case 0: extensions_mask = get_bits(&s->gb, 12); break;
    case 1: extensions_mask = DCA_EXT_EXSS_XLL;     break;
    case 2: extensions_mask = DCA_EXT_EXSS_LBR;     break;
    case 3: extensions_mask = 0; /* aux coding */   break;
    }

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

    if (get_bits_left(&s->gb) < 0)
        return -1;

    if (get_bits_count(&s->gb) - header_pos > header_size * 8) {
        av_log(s->avctx, AV_LOG_WARNING, "Asset header size mismatch.\n");
        return -1;
    }
    skip_bits_long(&s->gb, header_pos + header_size * 8 - get_bits_count(&s->gb));

    if (extensions_mask & DCA_EXT_EXSS_XLL)
        s->profile = FF_PROFILE_DTS_HD_MA;
    else if (extensions_mask & (DCA_EXT_EXSS_XBR | DCA_EXT_EXSS_X96 |
                                DCA_EXT_EXSS_XXCH))
        s->profile = FF_PROFILE_DTS_HD_HRA;

    if (!(extensions_mask & DCA_EXT_CORE))
        av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n");
    if ((extensions_mask & DCA_CORE_EXTS) != s->core_ext_mask)
        av_log(s->avctx, AV_LOG_WARNING,
               "DTS extensions detection mismatch (%d, %d)\n",
               extensions_mask & DCA_CORE_EXTS, s->core_ext_mask);

    return 0;
}

/**
 * Parse extension substream header (HD)
 */
static void dca_exss_parse_header(DCAContext *s)
{
    int ss_index;
    int blownup;
    int num_audiop = 1;
    int num_assets = 1;
    int active_ss_mask[8];
    int i, j;

    if (get_bits_left(&s->gb) < 52)
        return;

    skip_bits(&s->gb, 8); // user data
    ss_index = get_bits(&s->gb, 2);

    blownup = get_bits1(&s->gb);
    skip_bits(&s->gb,  8 + 4 * blownup); // header_size
    skip_bits(&s->gb, 16 + 4 * blownup); // hd_size

    s->static_fields = get_bits1(&s->gb);
    if (s->static_fields) {
        skip_bits(&s->gb, 2); // reference clock code
        skip_bits(&s->gb, 3); // frame duration code

        if (get_bits1(&s->gb))
            skip_bits_long(&s->gb, 36); // timestamp

        /* a single stream can contain multiple audio assets that can be
         * combined to form multiple audio presentations */

        num_audiop = get_bits(&s->gb, 3) + 1;
        if (num_audiop > 1) {
            avpriv_request_sample(s->avctx,
                                  "Multiple DTS-HD audio presentations");
            /* ignore such streams for now */
            return;
        }

        num_assets = get_bits(&s->gb, 3) + 1;
        if (num_assets > 1) {
            avpriv_request_sample(s->avctx, "Multiple DTS-HD audio assets");
            /* ignore such streams for now */
            return;
        }

        for (i = 0; i < num_audiop; i++)
            active_ss_mask[i] = get_bits(&s->gb, ss_index + 1);

        for (i = 0; i < num_audiop; i++)
            for (j = 0; j <= ss_index; j++)
                if (active_ss_mask[i] & (1 << j))
                    skip_bits(&s->gb, 8); // active asset mask

        s->mix_metadata = get_bits1(&s->gb);
        if (s->mix_metadata) {
            int mix_out_mask_size;

            skip_bits(&s->gb, 2); // adjustment level
            mix_out_mask_size  = (get_bits(&s->gb, 2) + 1) << 2;
            s->num_mix_configs =  get_bits(&s->gb, 2) + 1;

            for (i = 0; i < s->num_mix_configs; i++) {
                int mix_out_mask        = get_bits(&s->gb, mix_out_mask_size);
                s->mix_config_num_ch[i] = dca_exss_mask2count(mix_out_mask);
            }
        }
    }

    for (i = 0; i < num_assets; i++)
        skip_bits_long(&s->gb, 16 + 4 * blownup);  // asset size

    for (i = 0; i < num_assets; i++) {
        if (dca_exss_parse_asset_header(s))
            return;
    }

    /* not parsed further, we were only interested in the extensions mask
     * from the asset header */
}

/**
 * Main frame decoding function
 * FIXME add arguments
 */
static int dca_decode_frame(AVCodecContext *avctx, void *data,
                            int *got_frame_ptr, AVPacket *avpkt)
{
    AVFrame *frame     = data;
    const uint8_t *buf = avpkt->data;
    int buf_size = avpkt->size;

    int lfe_samples;
    int num_core_channels = 0;
    int i, ret;
    float  **samples_flt;
    DCAContext *s = avctx->priv_data;
    int channels, full_channels;
    int core_ss_end;


    s->xch_present = 0;

    s->dca_buffer_size = ff_dca_convert_bitstream(buf, buf_size, s->dca_buffer,
                                                  DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);
    if (s->dca_buffer_size == AVERROR_INVALIDDATA) {
        av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
        return AVERROR_INVALIDDATA;
    }

    init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
    if ((ret = dca_parse_frame_header(s)) < 0) {
        //seems like the frame is corrupt, try with the next one
        return ret;
    }
    //set AVCodec values with parsed data
    avctx->sample_rate = s->sample_rate;
    avctx->bit_rate    = s->bit_rate;

    s->profile = FF_PROFILE_DTS;

    for (i = 0; i < (s->sample_blocks / 8); i++) {
        if ((ret = dca_decode_block(s, 0, i))) {
            av_log(avctx, AV_LOG_ERROR, "error decoding block\n");
            return ret;
        }
    }

    /* record number of core channels incase less than max channels are requested */
    num_core_channels = s->prim_channels;

    if (s->ext_coding)
        s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];
    else
        s->core_ext_mask = 0;

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

    /* only scan for extensions if ext_descr was unknown or indicated a
     * supported XCh extension */
    if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {

        /* if ext_descr was unknown, clear s->core_ext_mask so that the
         * extensions scan can fill it up */
        s->core_ext_mask = FFMAX(s->core_ext_mask, 0);

        /* extensions start at 32-bit boundaries into bitstream */
        skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);

        while (core_ss_end - get_bits_count(&s->gb) >= 32) {
            uint32_t bits = get_bits_long(&s->gb, 32);

            switch (bits) {
            case 0x5a5a5a5a: {
                int ext_amode, xch_fsize;

                s->xch_base_channel = s->prim_channels;

                /* validate sync word using XCHFSIZE field */
                xch_fsize = show_bits(&s->gb, 10);
                if ((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&
                    (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))
                    continue;

                /* skip length-to-end-of-frame field for the moment */
                skip_bits(&s->gb, 10);

                s->core_ext_mask |= DCA_EXT_XCH;

                /* extension amode(number of channels in extension) should be 1 */
                /* AFAIK XCh is not used for more channels */
                if ((ext_amode = get_bits(&s->gb, 4)) != 1) {
                    av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"
                           " supported!\n", ext_amode);
                    continue;
                }

                /* much like core primary audio coding header */
                dca_parse_audio_coding_header(s, s->xch_base_channel);

                for (i = 0; i < (s->sample_blocks / 8); i++)
                    if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {
                        av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");
                        continue;
                    }

                s->xch_present = 1;
                break;
            }
            case 0x47004a03:
                /* XXCh: extended channels */
                /* usually found either in core or HD part in DTS-HD HRA streams,
                 * but not in DTS-ES which contains XCh extensions instead */
                s->core_ext_mask |= DCA_EXT_XXCH;
                break;

            case 0x1d95f262: {
                int fsize96 = show_bits(&s->gb, 12) + 1;
                if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)
                    continue;

                av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n",
                       get_bits_count(&s->gb));
                skip_bits(&s->gb, 12);
                av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);
                av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));

                s->core_ext_mask |= DCA_EXT_X96;
                break;
            }
            }

            skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
        }
    } else {
        /* no supported extensions, skip the rest of the core substream */
        skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));
    }

    if (s->core_ext_mask & DCA_EXT_X96)
        s->profile = FF_PROFILE_DTS_96_24;
    else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))
        s->profile = FF_PROFILE_DTS_ES;

    /* check for ExSS (HD part) */
    if (s->dca_buffer_size - s->frame_size > 32 &&
        get_bits_long(&s->gb, 32) == DCA_HD_MARKER)
        dca_exss_parse_header(s);

    avctx->profile = s->profile;

    full_channels = channels = s->prim_channels + !!s->lfe;

    if (s->amode < 16) {
        avctx->channel_layout = dca_core_channel_layout[s->amode];

        if (s->prim_channels + !!s->lfe > 2 &&
            avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
            /*
             * Neither the core's auxiliary data nor our default tables contain
             * downmix coefficients for the additional channel coded in the XCh
             * extension, so when we're doing a Stereo downmix, don't decode it.
             */
            s->xch_disable = 1;
        }

#if FF_API_REQUEST_CHANNELS
FF_DISABLE_DEPRECATION_WARNINGS
        if (s->xch_present && !s->xch_disable &&
            (!avctx->request_channels ||
             avctx->request_channels > num_core_channels + !!s->lfe)) {
FF_ENABLE_DEPRECATION_WARNINGS
#else
        if (s->xch_present && !s->xch_disable) {
#endif
            avctx->channel_layout |= AV_CH_BACK_CENTER;
            if (s->lfe) {
                avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
                s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode];
            } else {
                s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];
            }
        } else {
            channels = num_core_channels + !!s->lfe;
            s->xch_present = 0; /* disable further xch processing */
            if (s->lfe) {
                avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
                s->channel_order_tab = dca_channel_reorder_lfe[s->amode];
            } else
                s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];
        }

        if (channels > !!s->lfe &&
            s->channel_order_tab[channels - 1 - !!s->lfe] < 0)
            return AVERROR_INVALIDDATA;

        if (num_core_channels + !!s->lfe > 2 &&
            avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
            channels = 2;
            s->output = s->prim_channels == 2 ? s->amode : DCA_STEREO;
            avctx->channel_layout = AV_CH_LAYOUT_STEREO;

            /* Stereo downmix coefficients
             *
             * The decoder can only downmix to 2-channel, so we need to ensure
             * embedded downmix coefficients are actually targeting 2-channel.
             */
            if (s->core_downmix && (s->core_downmix_amode == DCA_STEREO ||
                                    s->core_downmix_amode == DCA_STEREO_TOTAL)) {
                int sign, code;
                for (i = 0; i < num_core_channels + !!s->lfe; i++) {
                    sign = s->core_downmix_codes[i][0] & 0x100 ? 1 : -1;
                    code = s->core_downmix_codes[i][0] & 0x0FF;
                    s->downmix_coef[i][0] = (!code ? 0.0f :
                                             sign * dca_dmixtable[code - 1]);
                    sign = s->core_downmix_codes[i][1] & 0x100 ? 1 : -1;
                    code = s->core_downmix_codes[i][1] & 0x0FF;
                    s->downmix_coef[i][1] = (!code ? 0.0f :
                                             sign * dca_dmixtable[code - 1]);
                }
                s->output = s->core_downmix_amode;
            } else {
                int am = s->amode & DCA_CHANNEL_MASK;
                if (am >= FF_ARRAY_ELEMS(dca_default_coeffs)) {
                    av_log(s->avctx, AV_LOG_ERROR,
                           "Invalid channel mode %d\n", am);
                    return AVERROR_INVALIDDATA;
                }
                if (num_core_channels + !!s->lfe >
                    FF_ARRAY_ELEMS(dca_default_coeffs[0])) {
                    avpriv_request_sample(s->avctx, "Downmixing %d channels",
                                          s->prim_channels + !!s->lfe);
                    return AVERROR_PATCHWELCOME;
                }
                for (i = 0; i < num_core_channels + !!s->lfe; i++) {
                    s->downmix_coef[i][0] = dca_default_coeffs[am][i][0];
                    s->downmix_coef[i][1] = dca_default_coeffs[am][i][1];
                }
            }
            av_dlog(s->avctx, "Stereo downmix coeffs:\n");
            for (i = 0; i < num_core_channels + !!s->lfe; i++) {
                av_dlog(s->avctx, "L, input channel %d = %f\n", i,
                        s->downmix_coef[i][0]);
                av_dlog(s->avctx, "R, input channel %d = %f\n", i,
                        s->downmix_coef[i][1]);
            }
            av_dlog(s->avctx, "\n");
        }
    } else {
        av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);
        return AVERROR_INVALIDDATA;
    }
    avctx->channels = channels;

    /* get output buffer */
    frame->nb_samples = 256 * (s->sample_blocks / 8);
    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
        return ret;
    }
    samples_flt = (float **)frame->extended_data;

    /* allocate buffer for extra channels if downmixing */
    if (avctx->channels < full_channels) {
        ret = av_samples_get_buffer_size(NULL, full_channels - channels,
                                         frame->nb_samples,
                                         avctx->sample_fmt, 0);
        if (ret < 0)
            return ret;

        av_fast_malloc(&s->extra_channels_buffer,
                       &s->extra_channels_buffer_size, ret);
        if (!s->extra_channels_buffer)
            return AVERROR(ENOMEM);

        ret = av_samples_fill_arrays((uint8_t **)s->extra_channels, NULL,
                                     s->extra_channels_buffer,
                                     full_channels - channels,
                                     frame->nb_samples, avctx->sample_fmt, 0);
        if (ret < 0)
            return ret;
    }

    /* filter to get final output */
    for (i = 0; i < (s->sample_blocks / 8); i++) {
        int ch;

        for (ch = 0; ch < channels; ch++)
            s->samples_chanptr[ch] = samples_flt[ch] + i * 256;
        for (; ch < full_channels; ch++)
            s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * 256;

        dca_filter_channels(s, i);

        /* If this was marked as a DTS-ES stream we need to subtract back- */
        /* channel from SL & SR to remove matrixed back-channel signal */
        if ((s->source_pcm_res & 1) && s->xch_present) {
            float *back_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel]];
            float *lt_chan   = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 2]];
            float *rt_chan   = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 1]];
            s->fdsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);
            s->fdsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);
        }
    }

    /* update lfe history */
    lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);
    for (i = 0; i < 2 * s->lfe * 4; i++)
        s->lfe_data[i] = s->lfe_data[i + lfe_samples];

    /* AVMatrixEncoding
     *
     * DCA_STEREO_TOTAL (Lt/Rt) is equivalent to Dolby Surround */
    ret = ff_side_data_update_matrix_encoding(frame,
                                              (s->output & ~DCA_LFE) == DCA_STEREO_TOTAL ?
                                              AV_MATRIX_ENCODING_DOLBY : AV_MATRIX_ENCODING_NONE);
    if (ret < 0)
        return ret;

    *got_frame_ptr = 1;

    return buf_size;
}



/**
 * DCA initialization
 *
 * @param avctx     pointer to the AVCodecContext
 */

static av_cold int dca_decode_init(AVCodecContext *avctx)
{
    DCAContext *s = avctx->priv_data;

    s->avctx = avctx;
    dca_init_vlcs();

    avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
    ff_mdct_init(&s->imdct, 6, 1, 1.0);
    ff_synth_filter_init(&s->synth);
    ff_dcadsp_init(&s->dcadsp);
    ff_fmt_convert_init(&s->fmt_conv, avctx);

    avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

    /* allow downmixing to stereo */
#if FF_API_REQUEST_CHANNELS
FF_DISABLE_DEPRECATION_WARNINGS
    if (avctx->request_channels == 2)
        avctx->request_channel_layout = AV_CH_LAYOUT_STEREO;
FF_ENABLE_DEPRECATION_WARNINGS
#endif
    if (avctx->channels > 2 &&
        avctx->request_channel_layout == AV_CH_LAYOUT_STEREO)
        avctx->channels = 2;

    return 0;
}

static av_cold int dca_decode_end(AVCodecContext *avctx)
{
    DCAContext *s = avctx->priv_data;
    ff_mdct_end(&s->imdct);
    av_freep(&s->extra_channels_buffer);
    return 0;
}

static const AVProfile profiles[] = {
    { FF_PROFILE_DTS,        "DTS"        },
    { FF_PROFILE_DTS_ES,     "DTS-ES"     },
    { FF_PROFILE_DTS_96_24,  "DTS 96/24"  },
    { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },
    { FF_PROFILE_DTS_HD_MA,  "DTS-HD MA"  },
    { FF_PROFILE_UNKNOWN },
};

static const AVOption options[] = {
    { "disable_xch", "disable decoding of the XCh extension", offsetof(DCAContext, xch_disable), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM|AV_OPT_FLAG_AUDIO_PARAM },
    { NULL },
};

static const AVClass dca_decoder_class = {
    .class_name = "DCA decoder",
    .item_name  = av_default_item_name,
    .option     = options,
    .version    = LIBAVUTIL_VERSION_INT,
};

AVCodec ff_dca_decoder = {
    .name            = "dca",
    .long_name       = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
    .type            = AVMEDIA_TYPE_AUDIO,
    .id              = AV_CODEC_ID_DTS,
    .priv_data_size  = sizeof(DCAContext),
    .init            = dca_decode_init,
    .decode          = dca_decode_frame,
    .close           = dca_decode_end,
    .capabilities    = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
    .sample_fmts     = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
                                                       AV_SAMPLE_FMT_NONE },
    .profiles        = NULL_IF_CONFIG_SMALL(profiles),
    .priv_class      = &dca_decoder_class,
};
