PeerStreamer

/*

 * 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 FFmpeg.

 *

 * FFmpeg 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.

 *

 * FFmpeg 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 FFmpeg; if not, write to the Free Software

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

 */

/**

 * @file libavcodec/dca.c

 */

#include <math.h>

#include <stddef.h>

#include <stdio.h>

#include "avcodec.h"

#include "dsputil.h"

#include "bitstream.h"

#include "put_bits.h"

#include "dcadata.h"

#include "dcahuff.h"

#include "dca.h"

//#define TRACE

#define DCA_PRIM_CHANNELS_MAX (5)

#define DCA_SUBBANDS (32)

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

#define DCA_SUBSUBFAMES_MAX (4)

#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

};

/* 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 -> CH_LAYOUT_STEREO

 */

static const int64_t dca_core_channel_layout[] = {

    CH_FRONT_CENTER,                                               ///< 1, A

    CH_LAYOUT_STEREO,                                              ///< 2, A + B (dual mono)

    CH_LAYOUT_STEREO,                                              ///< 2, L + R (stereo)

    CH_LAYOUT_STEREO,                                              ///< 2, (L+R) + (L-R) (sum-difference)

    CH_LAYOUT_STEREO,                                              ///< 2, LT +RT (left and right total)

    CH_LAYOUT_STEREO|CH_FRONT_CENTER,                              ///< 3, C+L+R

    CH_LAYOUT_STEREO|CH_BACK_CENTER,                               ///< 3, L+R+S

    CH_LAYOUT_STEREO|CH_FRONT_CENTER|CH_BACK_CENTER,               ///< 4, C + L + R+ S

    CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_SIDE_RIGHT,                   ///< 4, L + R +SL+ SR

    CH_LAYOUT_STEREO|CH_FRONT_CENTER|CH_SIDE_LEFT|CH_SIDE_RIGHT,   ///< 5, C + L + R+ SL+SR

    CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_SIDE_RIGHT|CH_FRONT_LEFT_OF_CENTER|CH_FRONT_RIGHT_OF_CENTER,                 ///< 6, CL + CR + L + R + SL + SR

    CH_LAYOUT_STEREO|CH_BACK_LEFT|CH_BACK_RIGHT|CH_FRONT_CENTER|CH_BACK_CENTER,                                   ///< 6, C + L + R+ LR + RR + OV

    CH_FRONT_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_FRONT_LEFT_OF_CENTER|CH_BACK_CENTER|CH_BACK_LEFT|CH_BACK_RIGHT,   ///< 6, CF+ CR+LF+ RF+LR + RR

    CH_FRONT_LEFT_OF_CENTER|CH_FRONT_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_SIDE_RIGHT, ///< 7, CL + C + CR + L + R + SL + SR

    CH_FRONT_LEFT_OF_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_SIDE_RIGHT|CH_BACK_LEFT|CH_BACK_RIGHT, ///< 8, CL + CR + L + R + SL1 + SL2+ SR1 + SR2

    CH_FRONT_LEFT_OF_CENTER|CH_FRONT_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_BACK_CENTER|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[][8] = {

    { 0, -1, -1, -1, -1, -1, -1, -1},

    { 0,  1, -1, -1, -1, -1, -1, -1},

    { 0,  1, -1, -1, -1, -1, -1, -1},

    { 0,  1, -1, -1, -1, -1, -1, -1},

    { 0,  1, -1, -1, -1, -1, -1, -1},

    { 2,  0,  1, -1, -1, -1, -1, -1},

    { 0,  1,  3, -1, -1, -1, -1, -1},

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

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

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

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

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

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

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

    { 5,  6,  0,  1,  7,  3,  8,  4},

    { 4,  2,  5,  0,  1,  6,  8,  7},

};

static const int8_t dca_channel_reorder_nolfe[][8] = {

    { 0, -1, -1, -1, -1, -1, -1, -1},

    { 0,  1, -1, -1, -1, -1, -1, -1},

    { 0,  1, -1, -1, -1, -1, -1, -1},

    { 0,  1, -1, -1, -1, -1, -1, -1},

    { 0,  1, -1, -1, -1, -1, -1, -1},

    { 2,  0,  1, -1, -1, -1, -1, -1},

    { 0,  1,  2, -1, -1, -1, -1, -1},

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

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

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

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

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

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

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

    { 4,  5,  0,  1,  6,  2,  7,  3},

    { 3,  2,  4,  0,  1,  5,  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

/** 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 {

    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 downmix;                ///< embedded downmix enabled

    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;           ///< number of subsubframes

    int partial_samples;        ///< 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

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

    int dynrange_coef;                                           ///< dynamic range coefficient

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

    float lfe_data[2 * DCA_SUBSUBFAMES_MAX * DCA_LFE_MAX *

                   2 /*history */ ];    ///< Low frequency effect data

    int lfe_scale_factor;

    /* Subband samples history (for ADPCM) */

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

    DECLARE_ALIGNED_16(float, subband_fir_hist[DCA_PRIM_CHANNELS_MAX][512]);

    float subband_fir_noidea[DCA_PRIM_CHANNELS_MAX][32];

    int hist_index[DCA_PRIM_CHANNELS_MAX];

    int output;                 ///< type of output

    float add_bias;             ///< output bias

    float scale_bias;           ///< output scale

    DECLARE_ALIGNED_16(float, samples[1536]);  /* 6 * 256 = 1536, might only need 5 */

    const float *samples_chanptr[6];

    uint8_t dca_buffer[DCA_MAX_FRAME_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 debug_flag;             ///< used for suppressing repeated error messages output

    DSPContext dsp;

    MDCTContext imdct;

} DCAContext;

static av_cold void dca_init_vlcs(void)

{

    static int vlcs_initialized = 0;

    int i, j;

    if (vlcs_initialized)

        return;

    dca_bitalloc_index.offset = 1;

    dca_bitalloc_index.wrap = 2;

    for (i = 0; i < 5; 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_STATIC);

    dca_scalefactor.offset = -64;

    dca_scalefactor.wrap = 2;

    for (i = 0; i < 5; i++)

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

                 scales_bits[i], 1, 1,

                 scales_codes[i], 2, 2, INIT_VLC_USE_STATIC);

    dca_tmode.offset = 0;

    dca_tmode.wrap = 1;

    for (i = 0; i < 4; i++)

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

                 tmode_bits[i], 1, 1,

                 tmode_codes[i], 2, 2, INIT_VLC_USE_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);

            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_STATIC);

        }

    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_frame_header(DCAContext * s)

{

    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 };

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

    /* Sync code */

    get_bits(&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 -1;

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

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

    if (!s->sample_rate)

        return -1;

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

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

    if (!s->bit_rate)

        return -1;

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

    /* 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, "downmix: %i\n", s->downmix);

    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;

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

    s->prim_channels     = s->total_channels;

    if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)

        s->prim_channels = DCA_PRIM_CHANNELS_MAX;   /* We only support DTS core */

    for (i = 0; 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 = 0; 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,     s->prim_channels, 3);

    get_array(&s->gb, s->transient_huffman,   s->prim_channels, 2);

    get_array(&s->gb, s->scalefactor_huffman, s->prim_channels, 3);

    get_array(&s->gb, s->bitalloc_huffman,    s->prim_channels, 3);

    /* Get codebooks quantization indexes */

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

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

        for (i = 0; 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 = 0; i < s->prim_channels; i++)

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

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

        for (i = 0; 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 = 0; 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 inline int get_scale(GetBitContext *gb, int level, int value)

{

   if (level < 5) {

       /* huffman encoded */

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

   } else if(level < 8)

       value = get_bits(gb, level + 1);

   return value;

}

static int dca_subframe_header(DCAContext * s)

{

    /* Primary audio coding side information */

    int j, k;

    s->subsubframes = get_bits(&s->gb, 2) + 1;

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

    for (j = 0; 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 = 0; 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 = 0; 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 -1;

            } else {

                s->bitalloc[j][k] =

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

            }

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

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

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

                return -1;

            }

        }

    }

    /* Transition mode */

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

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

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

            if (s->subsubframes > 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]);

            }

        }

    }

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

        const uint32_t *scale_table;

        int scale_sum;

        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;

        else

            scale_table = scale_factor_quant6;

        /* 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);

                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);

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

            }

        }

    }

    /* Joint subband scale factor codebook select */

    for (j = 0; 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);

    }

    /* Scale factors for joint subband coding */

    for (j = 0; 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], 0);

                scale += 64;    /* bias */

                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;

            }

        }

    }

    /* Stereo downmix coefficients */

    if (s->prim_channels > 2) {

        if(s->downmix) {

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

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

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

            }

        } else {

            int am = s->amode & DCA_CHANNEL_MASK;

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

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

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

            }

        }

    }

    /* Dynamic range coefficient */

    if (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 = 0; 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 (s->lfe) {

        /* LFE samples */

        int lfe_samples = 2 * s->lfe * s->subsubframes;

        float lfe_scale;

        for (j = lfe_samples; j < lfe_samples * 2; j++) {

            /* Signed 8 bits int */

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

        }

        /* Scale factor index */

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

        /* Quantization step size * scale factor */

        lfe_scale = 0.035 * s->lfe_scale_factor;

        for (j = lfe_samples; j < lfe_samples * 2; j++)

            s->lfe_data[j] *= lfe_scale;

    }

#ifdef TRACE

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

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

           s->partial_samples);

    for (j = 0; 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 = 0; 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 = 0; 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 = 0; 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 = 0; 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 = 0; 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");

        }

    }

    if (s->prim_channels > 2 && s->downmix) {

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

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

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

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

        }

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

    }

    for (j = 0; 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(s->lfe){

        int lfe_samples = 2 * s->lfe * s->subsubframes;

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

        for (j = lfe_samples; j < lfe_samples * 2; 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, float bias)

{

    const float *prCoeff;

    int i, j;

    DECLARE_ALIGNED_16(float, raXin[32]);

    int hist_index= s->hist_index[chans];

    float *subband_fir_hist2 = s->subband_fir_noidea[chans];

    int subindex;

    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;

    /* Reconstructed channel sample index */

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

        float *subband_fir_hist = s->subband_fir_hist[chans] + hist_index;

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

        for (i = 0; i < s->subband_activity[chans]; i++){

            if((i-1)&2) raXin[i] = -samples_in[i][subindex];

            else        raXin[i] =  samples_in[i][subindex];

        }

        for (; i < 32; i++)

            raXin[i] = 0.0;

        ff_imdct_half(&s->imdct, subband_fir_hist, raXin);

        /* Multiply by filter coefficients */

        for (i = 0; i < 16; i++){

            float a= subband_fir_hist2[i   ];

            float b= subband_fir_hist2[i+16];

            float c= 0;

            float d= 0;

            for (j = 0; j < 512-hist_index; j += 64){

                a += prCoeff[i+j   ]*(-subband_fir_hist[15-i+j]);

                b += prCoeff[i+j+16]*( subband_fir_hist[   i+j]);

                c += prCoeff[i+j+32]*( subband_fir_hist[16+i+j]);

                d += prCoeff[i+j+48]*( subband_fir_hist[31-i+j]);

            }

            for (     ; j < 512; j += 64){

                a += prCoeff[i+j   ]*(-subband_fir_hist[15-i+j-512]);

                b += prCoeff[i+j+16]*( subband_fir_hist[   i+j-512]);

                c += prCoeff[i+j+32]*( subband_fir_hist[16+i+j-512]);

                d += prCoeff[i+j+48]*( subband_fir_hist[31-i+j-512]);

            }

            samples_out[i   ] = a * scale + bias;

            samples_out[i+16] = b * scale + bias;

            subband_fir_hist2[i   ] = c;

            subband_fir_hist2[i+16] = d;

        }

        samples_out+= 32;

        hist_index = (hist_index-32)&511;

    }

    s->hist_index[chans]= hist_index;

}

static void lfe_interpolation_fir(int decimation_select,

                                  int num_deci_sample, float *samples_in,

                                  float *samples_out, float scale,

                                  float bias)

{

    /* 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 decifactor, k, j;

    const float *prCoeff;

    int interp_index = 0;       /* Index to the interpolated samples */

    int deciindex;

    /* Select decimation filter */

    if (decimation_select == 1) {

        decifactor = 128;

        prCoeff = lfe_fir_128;

    } else {

        decifactor = 64;

        prCoeff = lfe_fir_64;

    }

    /* Interpolation */

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

        /* One decimated sample generates decifactor interpolated ones */

        for (k = 0; k < decifactor; k++) {

            float rTmp = 0.0;

            //FIXME the coeffs are symetric, fix that

            for (j = 0; j < 512 / decifactor; j++)

                rTmp += samples_in[deciindex - j] * prCoeff[k + j * decifactor];

            samples_out[interp_index++] = (rTmp * scale) + bias;

        }

    }

}

/* downmixing routines */

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

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

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

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

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

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

#define MIX_FRONT3(samples, coef) \

    t = samples[i]; \

    samples[i]     = t * coef[0][0] + samples[i+256] * coef[1][0] + samples[i+512] * coef[2][0]; \

    samples[i+256] = t * coef[0][1] + samples[i+256] * coef[1][1] + samples[i+512] * 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 downmix_coef[DCA_PRIM_CHANNELS_MAX][2])

{

    int i;

    float t;

    float coef[DCA_PRIM_CHANNELS_MAX][2];

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

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

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

    }

    switch (srcfmt) {

    case DCA_MONO:

    case DCA_CHANNEL:

    case DCA_STEREO_TOTAL:

    case DCA_STEREO_SUMDIFF:

    case DCA_4F2R:

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

        break;

    case DCA_STEREO:

        break;

    case DCA_3F:

        DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),);

        break;

    case DCA_2F1R:

        DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + 512, 2, coef),);

        break;

    case DCA_3F1R:

        DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

                          MIX_REAR1(samples, i + 768, 3, coef));

        break;

    case DCA_2F2R:

        DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + 512, i + 768, 2, coef),);

        break;

    case DCA_3F2R:

        DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

                          MIX_REAR2(samples, i + 768, i + 1024, 3, coef));

        break;

    }

}

/* 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, int *values)

{

    int i;

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

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

        values[i] = (code % levels) - offset;

        code /= levels;

    }

    if (code == 0)

        return 0;

    else {

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

        return -1;

    }

}

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 };

static int dca_subsubframe(DCAContext * s)

{

    int k, l;

    int subsubframe = s->current_subsubframe;

    const float *quant_step_table;

    /* FIXME */

    float subband_samples[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];

    /*

     * 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 = 0; k < s->prim_channels; k++) {

        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];

            float rscale;

            /*

             * 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){

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

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

                if(abits <= 7){

                    /* Block code */

                    int block_code1, block_code2, size, levels;

                    int block[8];

                    size = abits_sizes[abits-1];

                    levels = abits_levels[abits-1];

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

                    /* FIXME Should test return value */

                    decode_blockcode(block_code1, levels, block);

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

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

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

                        subband_samples[k][l][m] = block[m];

                }else{

                    /* no coding */

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

                        subband_samples[k][l][m] = get_sbits(&s->gb, abits - 3);

                }

            }else{

                /* Huffman coded */

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

                    subband_samples[k][l][m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);

            }

            /* Deal with transients */

            if (s->transition_mode[k][l] &&

                subsubframe >= s->transition_mode[k][l])

                rscale = quant_step_size * s->scale_factor[k][l][1];

            else

                rscale = quant_step_size * s->scale_factor[k][l][0];

            rscale *= s->scalefactor_adj[k][sel];

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

                subband_samples[k][l][m] *= rscale;

            /*

             * 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 m;

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

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

                s->debug_flag |= 0x01;

            }

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

                subband_samples[k][l][m] =

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

                                                        m]

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

            }

        }

    }

    /* Check for DSYNC after subsubframe */

    if (s->aspf || subsubframe == s->subsubframes - 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");

        }

    }

    /* Backup predictor history for adpcm */

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

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

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

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

    /* 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};*/

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

                            M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ ,

                            s->add_bias );

    }

    /* Down mixing */

    if (s->prim_channels > dca_channels[s->output & DCA_CHANNEL_MASK]) {

        dca_downmix(s->samples, s->amode, s->downmix_coef);

    }

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

    if (s->output & DCA_LFE) {

        int lfe_samples = 2 * s->lfe * s->subsubframes;

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

                              s->lfe_data + lfe_samples +

                              2 * s->lfe * subsubframe,

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

                              (1.0/256.0)*s->scale_bias,  s->add_bias);

        /* Outputs 20bits pcm samples */

    }

    return 0;

}

static int dca_subframe_footer(DCAContext * s)

{

    int aux_data_count = 0, i;

    int lfe_samples;

    /*

     * Unpack optional information

     */

    if (s->timestamp)

        get_bits(&s->gb, 32);

    if (s->aux_data)

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

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

        get_bits(&s->gb, 8);

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

        get_bits(&s->gb, 16);

    lfe_samples = 2 * s->lfe * s->subsubframes;

    for (i = 0; i < lfe_samples; i++) {

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

    }

    return 0;

}

/**

 * Decode a dca frame block

 *

 * @param s     pointer to the DCAContext

 */

static int dca_decode_block(DCAContext * s)

{

    /* 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 -1;

    }

    if (!s->current_subsubframe) {

#ifdef TRACE

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

#endif

        /* Read subframe header */

        if (dca_subframe_header(s))

            return -1;

    }

    /* Read subsubframe */

#ifdef TRACE

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

#endif

    if (dca_subsubframe(s))

        return -1;

    /* Update state */

    s->current_subsubframe++;

    if (s->current_subsubframe >= s->subsubframes) {

        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 (dca_subframe_footer(s))

            return -1;

    }

    return 0;

}

/**

 * Convert bitstream to one representation based on sync marker

 */

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

                          int max_size)

{

    uint32_t mrk;

    int i, tmp;

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

    uint16_t *sdst = (uint16_t *) dst;

    PutBitContext pb;

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

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

//        return -1;

        src_size = max_size;

    }

    mrk = AV_RB32(src);

    switch (mrk) {

    case DCA_MARKER_RAW_BE:

        memcpy(dst, src, src_size);

        return src_size;

    case DCA_MARKER_RAW_LE:

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

            *sdst++ = bswap_16(*ssrc++);

        return src_size;

    case DCA_MARKER_14B_BE:

    case DCA_MARKER_14B_LE:

        init_put_bits(&pb, dst, max_size);

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

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

            put_bits(&pb, 14, tmp);

        }

        flush_put_bits(&pb);

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

    default:

        return -1;

    }

}

/**

 * Main frame decoding function

 * FIXME add arguments

 */

static int dca_decode_frame(AVCodecContext * avctx,