svq1: replace struct svq1_frame_size with an array.
[ffmpeg.git] / libavcodec / dca.c
index c1b19e3..0f1eeec 100644 (file)
 /*
- * DCA compatible decoder
- * Copyright (C) 2004 Gildas Bazin
- * Copyright (C) 2004 Benjamin Zores
- * Copyright (C) 2006 Benjamin Larsson
- * Copyright (C) 2007 Konstantin Shishkov
+ * DCA compatible decoder data
  *
- * This file is part of FFmpeg.
+ * This file is part of Libav.
  *
- * FFmpeg is free software; you can redistribute it and/or
+ * 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.
  *
- * FFmpeg is distributed in the hope that it will be useful,
+ * 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 FFmpeg; if not, write to the Free Software
+ * 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 <stdint.h>
 
-#include "libavutil/intmath.h"
-#include "libavutil/intreadwrite.h"
-#include "avcodec.h"
-#include "dsputil.h"
-#include "fft.h"
-#include "get_bits.h"
-#include "put_bits.h"
-#include "dcadata.h"
-#include "dcahuff.h"
 #include "dca.h"
-#include "synth_filter.h"
-#include "dcadsp.h"
 
-//#define TRACE
-
-#define DCA_PRIM_CHANNELS_MAX (5)
-#define DCA_SUBBANDS (32)
-#define DCA_ABITS_MAX (32)      /* Should be 28 */
-#define DCA_SUBSUBFRAMES_MAX (4)
-#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
-};
-
-/* 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)
+const uint32_t avpriv_dca_sample_rates[16] =
 {
-    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_LFE_MAX * (DCA_BLOCKS_MAX + 4)];      ///< 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];
-    DECLARE_ALIGNED(16, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32];
-    int hist_index[DCA_PRIM_CHANNELS_MAX];
-    DECLARE_ALIGNED(16, float, raXin)[32];
-
-    int output;                 ///< type of output
-    float add_bias;             ///< output bias
-    float scale_bias;           ///< output scale
-
-    DECLARE_ALIGNED(16, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
-    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;
-    FFTContext imdct;
-    SynthFilterContext synth;
-    DCADSPContext dcadsp;
-} 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 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;
-    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 int dca_parse_frame_header(DCAContext * s)
-{
-    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;
-
-    return dca_parse_audio_coding_header(s);
-}
-
-
-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, int block_index)
-{
-    /* 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 * (4 + block_index);
-        int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes);
-        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 */
-        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_end_sample; 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 * (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, float bias)
-{
-    const float *prCoeff;
-    int i;
-
-    int sb_act = s->subband_activity[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++) {
-        /* Load in one sample from each subband and clear inactive subbands */
-        for (i = 0; i < sb_act; i++){
-            uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ ((i-1)&2)<<30;
-            AV_WN32A(&s->raXin[i], v);
-        }
-        for (; i < 32; i++)
-            s->raXin[i] = 0.0;
-
-        s->synth.synth_filter_float(&s->imdct,
-                              s->subband_fir_hist[chans], &s->hist_index[chans],
-                              s->subband_fir_noidea[chans], prCoeff,
-                              samples_out, s->raXin, scale, bias);
-        samples_out+= 32;
-
-    }
-}
-
-static void lfe_interpolation_fir(DCAContext *s, 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;
-    const float *prCoeff;
-    int deciindex;
-
-    /* Select decimation filter */
-    if (decimation_select == 1) {
-        decifactor = 64;
-        prCoeff = lfe_fir_128;
-    } else {
-        decifactor = 32;
-        prCoeff = lfe_fir_64;
-    }
-    /* Interpolation */
-    for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
-        s->dcadsp.lfe_fir(samples_out, samples_in, prCoeff, decifactor,
-                          scale, bias);
-        samples_in++;
-        samples_out += 2 * decifactor;
-    }
-}
-
-/* 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++) {
-        int div = FASTDIV(code, levels);
-        values[i] = code - offset - div*levels;
-        code = div;
-    }
-
-    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 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(int, block, [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];
-
-            /*
-             * 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 {
-                /* Deal with transients */
-                int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];
-                float rscale = 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;
-
-                        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]);
-                    }else{
-                        /* no coding */
-                        for (m = 0; m < 8; m++)
-                            block[m] = get_sbits(&s->gb, abits - 3);
-                    }
-                }else{
-                    /* Huffman coded */
-                    for (m = 0; m < 8; m++)
-                        block[m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);
-                }
-
-                s->dsp.int32_to_float_fmul_scalar(subband_samples[k][l],
-                                                  block, rscale, 8);
-            }
-
-            /*
-             * 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]));
-
-    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};*/
-         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->avctx->request_channels == 2 && s->prim_channels > 2) {
-        dca_downmix(s->samples, s->amode, s->downmix_coef);
-    }
-
-    /* Generate LFE samples for this subsubframe FIXME!!! */
-    if (s->output & DCA_LFE) {
-        lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,
-                              s->lfe_data + 2 * s->lfe * (block_index + 4),
-                              &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;
-
-    /*
-     * 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);
-
-    return 0;
-}
-
-/**
- * Decode a dca frame block
- *
- * @param s     pointer to the DCAContext
- */
-
-static int dca_decode_block(DCAContext * s, int block_index)
-{
-
-    /* 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, block_index))
-            return -1;
-    }
-
-    /* Read subsubframe */
-#ifdef TRACE
-    av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");
-#endif
-    if (dca_subsubframe(s, block_index))
-        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,
-                            void *data, int *data_size,
-                            AVPacket *avpkt)
-{
-    const uint8_t *buf = avpkt->data;
-    int buf_size = avpkt->size;
-
-    int lfe_samples;
-    int i;
-    int16_t *samples = data;
-    DCAContext *s = avctx->priv_data;
-    int channels;
-
-
-    s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer, DCA_MAX_FRAME_SIZE);
-    if (s->dca_buffer_size == -1) {
-        av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
-        return -1;
-    }
-
-    init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
-    if (dca_parse_frame_header(s) < 0) {
-        //seems like the frame is corrupt, try with the next one
-        *data_size=0;
-        return buf_size;
-    }
-    //set AVCodec values with parsed data
-    avctx->sample_rate = s->sample_rate;
-    avctx->bit_rate = s->bit_rate;
-
-    for (i = 0; i < (s->sample_blocks / 8); i++) {
-        dca_decode_block(s, i);
-    }
-
-    channels = s->prim_channels + !!s->lfe;
-
-    if (s->amode<16) {
-        avctx->channel_layout = dca_core_channel_layout[s->amode];
-
-        if (s->lfe) {
-            avctx->channel_layout |= 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 (s->prim_channels > 0 &&
-            s->channel_order_tab[s->prim_channels - 1] < 0)
-            return -1;
-
-        if(avctx->request_channels == 2 && s->prim_channels > 2) {
-            channels = 2;
-            s->output = DCA_STEREO;
-            avctx->channel_layout = CH_LAYOUT_STEREO;
-        }
-    } else {
-        av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n",s->amode);
-        return -1;
-    }
-
-
-    /* There is nothing that prevents a dts frame to change channel configuration
-       but FFmpeg doesn't support that so only set the channels if it is previously
-       unset. Ideally during the first probe for channels the crc should be checked
-       and only set avctx->channels when the crc is ok. Right now the decoder could
-       set the channels based on a broken first frame.*/
-    if (!avctx->channels)
-        avctx->channels = channels;
-
-    if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
-        return -1;
-    *data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels;
-
-    /* filter to get final output */
-    for (i = 0; i < (s->sample_blocks / 8); i++) {
-        dca_filter_channels(s, i);
-        s->dsp.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels);
-        samples += 256 * channels;
-    }
-
-    /* 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];
-    }
-
-    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;
-    int i;
-
-    s->avctx = avctx;
-    dca_init_vlcs();
-
-    dsputil_init(&s->dsp, avctx);
-    ff_mdct_init(&s->imdct, 6, 1, 1.0);
-    ff_synth_filter_init(&s->synth);
-    ff_dcadsp_init(&s->dcadsp);
-
-    for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)
-        s->samples_chanptr[i] = s->samples + i * 256;
-    avctx->sample_fmt = SAMPLE_FMT_S16;
-
-    if(s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
-        s->add_bias = 385.0f;
-        s->scale_bias = 1.0 / 32768.0;
-    } else {
-        s->add_bias = 0.0f;
-        s->scale_bias = 1.0;
-
-        /* allow downmixing to stereo */
-        if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
-                avctx->request_channels == 2) {
-            avctx->channels = avctx->request_channels;
-        }
-    }
-
-
-    return 0;
-}
-
-static av_cold int dca_decode_end(AVCodecContext * avctx)
-{
-    DCAContext *s = avctx->priv_data;
-    ff_mdct_end(&s->imdct);
-    return 0;
-}
-
-AVCodec dca_decoder = {
-    .name = "dca",
-    .type = AVMEDIA_TYPE_AUDIO,
-    .id = CODEC_ID_DTS,
-    .priv_data_size = sizeof(DCAContext),
-    .init = dca_decode_init,
-    .decode = dca_decode_frame,
-    .close = dca_decode_end,
-    .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
+    0, 8000, 16000, 32000, 0, 0, 11025, 22050, 44100, 0, 0,
+    12000, 24000, 48000, 96000, 192000
 };