svq1: replace struct svq1_frame_size with an array.
[ffmpeg.git] / libavcodec / dca.c
index 6dbcda0..0f1eeec 100644 (file)
@@ -1,9 +1,5 @@
 /*
- * 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 Libav.
  *
  * 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/common.h"
-#include "libavutil/intmath.h"
-#include "libavutil/intreadwrite.h"
-#include "libavutil/audioconvert.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 "dca_parser.h"
-#include "synth_filter.h"
-#include "dcadsp.h"
-#include "fmtconvert.h"
 
-#if ARCH_ARM
-#   include "arm/dca.h"
-#endif
-
-//#define TRACE
-
-#define DCA_PRIM_CHANNELS_MAX  (7)
-#define DCA_SUBBANDS          (32)
-#define DCA_ABITS_MAX         (32)      /* Should be 28 */
-#define DCA_SUBSUBFRAMES_MAX   (4)
-#define DCA_SUBFRAMES_MAX     (16)
-#define DCA_BLOCKS_MAX        (16)
-#define DCA_LFE_MAX            (3)
-
-enum DCAMode {
-    DCA_MONO = 0,
-    DCA_CHANNEL,
-    DCA_STEREO,
-    DCA_STEREO_SUMDIFF,
-    DCA_STEREO_TOTAL,
-    DCA_3F,
-    DCA_2F1R,
-    DCA_3F1R,
-    DCA_2F2R,
-    DCA_3F2R,
-    DCA_4F2R
-};
-
-/* these are unconfirmed but should be mostly correct */
-enum DCAExSSSpeakerMask {
-    DCA_EXSS_FRONT_CENTER          = 0x0001,
-    DCA_EXSS_FRONT_LEFT_RIGHT      = 0x0002,
-    DCA_EXSS_SIDE_REAR_LEFT_RIGHT  = 0x0004,
-    DCA_EXSS_LFE                   = 0x0008,
-    DCA_EXSS_REAR_CENTER           = 0x0010,
-    DCA_EXSS_FRONT_HIGH_LEFT_RIGHT = 0x0020,
-    DCA_EXSS_REAR_LEFT_RIGHT       = 0x0040,
-    DCA_EXSS_FRONT_HIGH_CENTER     = 0x0080,
-    DCA_EXSS_OVERHEAD              = 0x0100,
-    DCA_EXSS_CENTER_LEFT_RIGHT     = 0x0200,
-    DCA_EXSS_WIDE_LEFT_RIGHT       = 0x0400,
-    DCA_EXSS_SIDE_LEFT_RIGHT       = 0x0800,
-    DCA_EXSS_LFE2                  = 0x1000,
-    DCA_EXSS_SIDE_HIGH_LEFT_RIGHT  = 0x2000,
-    DCA_EXSS_REAR_HIGH_CENTER      = 0x4000,
-    DCA_EXSS_REAR_HIGH_LEFT_RIGHT  = 0x8000,
-};
-
-enum DCAExtensionMask {
-    DCA_EXT_CORE       = 0x001, ///< core in core substream
-    DCA_EXT_XXCH       = 0x002, ///< XXCh channels extension in core substream
-    DCA_EXT_X96        = 0x004, ///< 96/24 extension in core substream
-    DCA_EXT_XCH        = 0x008, ///< XCh channel extension in core substream
-    DCA_EXT_EXSS_CORE  = 0x010, ///< core in ExSS (extension substream)
-    DCA_EXT_EXSS_XBR   = 0x020, ///< extended bitrate extension in ExSS
-    DCA_EXT_EXSS_XXCH  = 0x040, ///< XXCh channels extension in ExSS
-    DCA_EXT_EXSS_X96   = 0x080, ///< 96/24 extension in ExSS
-    DCA_EXT_EXSS_LBR   = 0x100, ///< low bitrate component in ExSS
-    DCA_EXT_EXSS_XLL   = 0x200, ///< lossless extension in ExSS
-};
-
-/* -1 are reserved or unknown */
-static const int dca_ext_audio_descr_mask[] = {
-    DCA_EXT_XCH,
-    -1,
-    DCA_EXT_X96,
-    DCA_EXT_XCH | DCA_EXT_X96,
-    -1,
-    -1,
-    DCA_EXT_XXCH,
-    -1,
-};
-
-/* extensions that reside in core substream */
-#define DCA_CORE_EXTS (DCA_EXT_XCH | DCA_EXT_XXCH | DCA_EXT_X96)
-
-/* Tables for mapping dts channel configurations to libavcodec multichannel api.
- * Some compromises have been made for special configurations. Most configurations
- * are never used so complete accuracy is not needed.
- *
- * L = left, R = right, C = center, S = surround, F = front, R = rear, T = total, OV = overhead.
- * S  -> side, when both rear and back are configured move one of them to the side channel
- * OV -> center back
- * All 2 channel configurations -> AV_CH_LAYOUT_STEREO
- */
-static const uint64_t dca_core_channel_layout[] = {
-    AV_CH_FRONT_CENTER,                                                     ///< 1, A
-    AV_CH_LAYOUT_STEREO,                                                    ///< 2, A + B (dual mono)
-    AV_CH_LAYOUT_STEREO,                                                    ///< 2, L + R (stereo)
-    AV_CH_LAYOUT_STEREO,                                                    ///< 2, (L + R) + (L - R) (sum-difference)
-    AV_CH_LAYOUT_STEREO,                                                    ///< 2, LT + RT (left and right total)
-    AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER,                               ///< 3, C + L + R
-    AV_CH_LAYOUT_STEREO | AV_CH_BACK_CENTER,                                ///< 3, L + R + S
-    AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER,           ///< 4, C + L + R + S
-    AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT,               ///< 4, L + R + SL + SR
-
-    AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_SIDE_LEFT |
-    AV_CH_SIDE_RIGHT,                                                       ///< 5, C + L + R + SL + SR
-
-    AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |
-    AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER,               ///< 6, CL + CR + L + R + SL + SR
-
-    AV_CH_LAYOUT_STEREO | AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT |
-    AV_CH_FRONT_CENTER  | AV_CH_BACK_CENTER,                                ///< 6, C + L + R + LR + RR + OV
-
-    AV_CH_FRONT_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |
-    AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_BACK_CENTER   |
-    AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT,                                     ///< 6, CF + CR + LF + RF + LR + RR
-
-    AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER   |
-    AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |
-    AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT,                                     ///< 7, CL + C + CR + L + R + SL + SR
-
-    AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |
-    AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |
-    AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT,                                     ///< 8, CL + CR + L + R + SL1 + SL2 + SR1 + SR2
-
-    AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER   |
-    AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |
-    AV_CH_SIDE_LEFT | AV_CH_BACK_CENTER | AV_CH_SIDE_RIGHT,                 ///< 8, CL + C + CR + L + R + SL + S + SR
-};
-
-static const int8_t dca_lfe_index[] = {
-    1, 2, 2, 2, 2, 3, 2, 3, 2, 3, 2, 3, 1, 3, 2, 3
-};
-
-static const int8_t dca_channel_reorder_lfe[][9] = {
-    { 0, -1, -1, -1, -1, -1, -1, -1, -1},
-    { 0,  1, -1, -1, -1, -1, -1, -1, -1},
-    { 0,  1, -1, -1, -1, -1, -1, -1, -1},
-    { 0,  1, -1, -1, -1, -1, -1, -1, -1},
-    { 0,  1, -1, -1, -1, -1, -1, -1, -1},
-    { 2,  0,  1, -1, -1, -1, -1, -1, -1},
-    { 0,  1,  3, -1, -1, -1, -1, -1, -1},
-    { 2,  0,  1,  4, -1, -1, -1, -1, -1},
-    { 0,  1,  3,  4, -1, -1, -1, -1, -1},
-    { 2,  0,  1,  4,  5, -1, -1, -1, -1},
-    { 3,  4,  0,  1,  5,  6, -1, -1, -1},
-    { 2,  0,  1,  4,  5,  6, -1, -1, -1},
-    { 0,  6,  4,  5,  2,  3, -1, -1, -1},
-    { 4,  2,  5,  0,  1,  6,  7, -1, -1},
-    { 5,  6,  0,  1,  7,  3,  8,  4, -1},
-    { 4,  2,  5,  0,  1,  6,  8,  7, -1},
-};
-
-static const int8_t dca_channel_reorder_lfe_xch[][9] = {
-    { 0,  2, -1, -1, -1, -1, -1, -1, -1},
-    { 0,  1,  3, -1, -1, -1, -1, -1, -1},
-    { 0,  1,  3, -1, -1, -1, -1, -1, -1},
-    { 0,  1,  3, -1, -1, -1, -1, -1, -1},
-    { 0,  1,  3, -1, -1, -1, -1, -1, -1},
-    { 2,  0,  1,  4, -1, -1, -1, -1, -1},
-    { 0,  1,  3,  4, -1, -1, -1, -1, -1},
-    { 2,  0,  1,  4,  5, -1, -1, -1, -1},
-    { 0,  1,  4,  5,  3, -1, -1, -1, -1},
-    { 2,  0,  1,  5,  6,  4, -1, -1, -1},
-    { 3,  4,  0,  1,  6,  7,  5, -1, -1},
-    { 2,  0,  1,  4,  5,  6,  7, -1, -1},
-    { 0,  6,  4,  5,  2,  3,  7, -1, -1},
-    { 4,  2,  5,  0,  1,  7,  8,  6, -1},
-    { 5,  6,  0,  1,  8,  3,  9,  4,  7},
-    { 4,  2,  5,  0,  1,  6,  9,  8,  7},
-};
-
-static const int8_t dca_channel_reorder_nolfe[][9] = {
-    { 0, -1, -1, -1, -1, -1, -1, -1, -1},
-    { 0,  1, -1, -1, -1, -1, -1, -1, -1},
-    { 0,  1, -1, -1, -1, -1, -1, -1, -1},
-    { 0,  1, -1, -1, -1, -1, -1, -1, -1},
-    { 0,  1, -1, -1, -1, -1, -1, -1, -1},
-    { 2,  0,  1, -1, -1, -1, -1, -1, -1},
-    { 0,  1,  2, -1, -1, -1, -1, -1, -1},
-    { 2,  0,  1,  3, -1, -1, -1, -1, -1},
-    { 0,  1,  2,  3, -1, -1, -1, -1, -1},
-    { 2,  0,  1,  3,  4, -1, -1, -1, -1},
-    { 2,  3,  0,  1,  4,  5, -1, -1, -1},
-    { 2,  0,  1,  3,  4,  5, -1, -1, -1},
-    { 0,  5,  3,  4,  1,  2, -1, -1, -1},
-    { 3,  2,  4,  0,  1,  5,  6, -1, -1},
-    { 4,  5,  0,  1,  6,  2,  7,  3, -1},
-    { 3,  2,  4,  0,  1,  5,  7,  6, -1},
-};
-
-static const int8_t dca_channel_reorder_nolfe_xch[][9] = {
-    { 0,  1, -1, -1, -1, -1, -1, -1, -1},
-    { 0,  1,  2, -1, -1, -1, -1, -1, -1},
-    { 0,  1,  2, -1, -1, -1, -1, -1, -1},
-    { 0,  1,  2, -1, -1, -1, -1, -1, -1},
-    { 0,  1,  2, -1, -1, -1, -1, -1, -1},
-    { 2,  0,  1,  3, -1, -1, -1, -1, -1},
-    { 0,  1,  2,  3, -1, -1, -1, -1, -1},
-    { 2,  0,  1,  3,  4, -1, -1, -1, -1},
-    { 0,  1,  3,  4,  2, -1, -1, -1, -1},
-    { 2,  0,  1,  4,  5,  3, -1, -1, -1},
-    { 2,  3,  0,  1,  5,  6,  4, -1, -1},
-    { 2,  0,  1,  3,  4,  5,  6, -1, -1},
-    { 0,  5,  3,  4,  1,  2,  6, -1, -1},
-    { 3,  2,  4,  0,  1,  6,  7,  5, -1},
-    { 4,  5,  0,  1,  7,  2,  8,  3,  6},
-    { 3,  2,  4,  0,  1,  5,  8,  7,  6},
-};
-
-#define DCA_DOLBY                  101           /* FIXME */
-
-#define DCA_CHANNEL_BITS             6
-#define DCA_CHANNEL_MASK          0x3F
-
-#define DCA_LFE                   0x80
-
-#define HEADER_SIZE                 14
-
-#define DCA_MAX_FRAME_SIZE       16384
-#define DCA_MAX_EXSS_HEADER_SIZE  4096
-
-#define DCA_BUFFER_PADDING_SIZE   1024
-
-/** 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;
-    AVFrame frame;
-    /* 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 is_channels_set;        ///< check for if the channel number is already set
-    int total_channels;         ///< number of channels including extensions
-    int prim_channels;          ///< number of primary audio channels
-    int subband_activity[DCA_PRIM_CHANNELS_MAX];    ///< subband activity count
-    int vq_start_subband[DCA_PRIM_CHANNELS_MAX];    ///< high frequency vq start subband
-    int joint_intensity[DCA_PRIM_CHANNELS_MAX];     ///< joint intensity coding index
-    int transient_huffman[DCA_PRIM_CHANNELS_MAX];   ///< transient mode code book
-    int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book
-    int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX];    ///< bit allocation quantizer select
-    int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select
-    float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX];   ///< scale factor adjustment
-
-    /* Primary audio coding side information */
-    int subsubframes[DCA_SUBFRAMES_MAX];                         ///< number of subsubframes
-    int partial_samples[DCA_SUBFRAMES_MAX];                      ///< partial subsubframe samples count
-    int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];    ///< prediction mode (ADPCM used or not)
-    int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];      ///< prediction VQ coefs
-    int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];           ///< bit allocation index
-    int transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];    ///< transition mode (transients)
-    int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2];    ///< scale factors (2 if transient)
-    int joint_huff[DCA_PRIM_CHANNELS_MAX];                       ///< joint subband scale factors codebook
-    int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors
-    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) */
-    DECLARE_ALIGNED(16, float, subband_samples_hist)[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
-    DECLARE_ALIGNED(32, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512];
-    DECLARE_ALIGNED(32, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32];
-    int hist_index[DCA_PRIM_CHANNELS_MAX];
-    DECLARE_ALIGNED(32, float, raXin)[32];
-
-    int output;                 ///< type of output
-    float scale_bias;           ///< output scale
-
-    DECLARE_ALIGNED(32, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
-    DECLARE_ALIGNED(32, float, samples)[(DCA_PRIM_CHANNELS_MAX + 1) * 256];
-    const float *samples_chanptr[DCA_PRIM_CHANNELS_MAX + 1];
-
-    uint8_t dca_buffer[DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE + DCA_BUFFER_PADDING_SIZE];
-    int dca_buffer_size;        ///< how much data is in the dca_buffer
-
-    const int8_t *channel_order_tab;  ///< channel reordering table, lfe and non lfe
-    GetBitContext gb;
-    /* Current position in DCA frame */
-    int current_subframe;
-    int current_subsubframe;
-
-    int core_ext_mask;          ///< present extensions in the core substream
-
-    /* XCh extension information */
-    int xch_present;            ///< XCh extension present and valid
-    int xch_base_channel;       ///< index of first (only) channel containing XCH data
-
-    /* ExSS header parser */
-    int static_fields;          ///< static fields present
-    int mix_metadata;           ///< mixing metadata present
-    int num_mix_configs;        ///< number of mix out configurations
-    int mix_config_num_ch[4];   ///< number of channels in each mix out configuration
-
-    int profile;
-
-    int debug_flag;             ///< used for suppressing repeated error messages output
-    DSPContext dsp;
-    FFTContext imdct;
-    SynthFilterContext synth;
-    DCADSPContext dcadsp;
-    FmtConvertContext fmt_conv;
-} DCAContext;
-
-static const uint16_t dca_vlc_offs[] = {
-        0,   512,   640,   768,  1282,  1794,  2436,  3080,  3770,  4454,  5364,
-     5372,  5380,  5388,  5392,  5396,  5412,  5420,  5428,  5460,  5492,  5508,
-     5572,  5604,  5668,  5796,  5860,  5892,  6412,  6668,  6796,  7308,  7564,
-     7820,  8076,  8620,  9132,  9388,  9910, 10166, 10680, 11196, 11726, 12240,
-    12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264,
-    18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622,
-};
-
-static av_cold void dca_init_vlcs(void)
-{
-    static int vlcs_initialized = 0;
-    int i, j, c = 14;
-    static VLC_TYPE dca_table[23622][2];
-
-    if (vlcs_initialized)
-        return;
-
-    dca_bitalloc_index.offset = 1;
-    dca_bitalloc_index.wrap = 2;
-    for (i = 0; i < 5; i++) {
-        dca_bitalloc_index.vlc[i].table = &dca_table[dca_vlc_offs[i]];
-        dca_bitalloc_index.vlc[i].table_allocated = dca_vlc_offs[i + 1] - dca_vlc_offs[i];
-        init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,
-                 bitalloc_12_bits[i], 1, 1,
-                 bitalloc_12_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
-    }
-    dca_scalefactor.offset = -64;
-    dca_scalefactor.wrap = 2;
-    for (i = 0; i < 5; i++) {
-        dca_scalefactor.vlc[i].table = &dca_table[dca_vlc_offs[i + 5]];
-        dca_scalefactor.vlc[i].table_allocated = dca_vlc_offs[i + 6] - dca_vlc_offs[i + 5];
-        init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
-                 scales_bits[i], 1, 1,
-                 scales_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
-    }
-    dca_tmode.offset = 0;
-    dca_tmode.wrap = 1;
-    for (i = 0; i < 4; i++) {
-        dca_tmode.vlc[i].table = &dca_table[dca_vlc_offs[i + 10]];
-        dca_tmode.vlc[i].table_allocated = dca_vlc_offs[i + 11] - dca_vlc_offs[i + 10];
-        init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
-                 tmode_bits[i], 1, 1,
-                 tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
-    }
-
-    for (i = 0; i < 10; i++)
-        for (j = 0; j < 7; j++) {
-            if (!bitalloc_codes[i][j])
-                break;
-            dca_smpl_bitalloc[i + 1].offset                 = bitalloc_offsets[i];
-            dca_smpl_bitalloc[i + 1].wrap                   = 1 + (j > 4);
-            dca_smpl_bitalloc[i + 1].vlc[j].table           = &dca_table[dca_vlc_offs[c]];
-            dca_smpl_bitalloc[i + 1].vlc[j].table_allocated = dca_vlc_offs[c + 1] - dca_vlc_offs[c];
-
-            init_vlc(&dca_smpl_bitalloc[i + 1].vlc[j], bitalloc_maxbits[i][j],
-                     bitalloc_sizes[i],
-                     bitalloc_bits[i][j], 1, 1,
-                     bitalloc_codes[i][j], 2, 2, INIT_VLC_USE_NEW_STATIC);
-            c++;
-        }
-    vlcs_initialized = 1;
-}
-
-static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
-{
-    while (len--)
-        *dst++ = get_bits(gb, bits);
-}
-
-static int dca_parse_audio_coding_header(DCAContext *s, int base_channel)
+const uint32_t avpriv_dca_sample_rates[16] =
 {
-    int i, j;
-    static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
-    static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
-    static const int thr[11]    = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
-
-    s->total_channels = get_bits(&s->gb, 3) + 1 + base_channel;
-    s->prim_channels  = s->total_channels;
-
-    if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)
-        s->prim_channels = DCA_PRIM_CHANNELS_MAX;
-
-
-    for (i = base_channel; i < s->prim_channels; i++) {
-        s->subband_activity[i] = get_bits(&s->gb, 5) + 2;
-        if (s->subband_activity[i] > DCA_SUBBANDS)
-            s->subband_activity[i] = DCA_SUBBANDS;
-    }
-    for (i = base_channel; i < s->prim_channels; i++) {
-        s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
-        if (s->vq_start_subband[i] > DCA_SUBBANDS)
-            s->vq_start_subband[i] = DCA_SUBBANDS;
-    }
-    get_array(&s->gb, s->joint_intensity + base_channel,     s->prim_channels - base_channel, 3);
-    get_array(&s->gb, s->transient_huffman + base_channel,   s->prim_channels - base_channel, 2);
-    get_array(&s->gb, s->scalefactor_huffman + base_channel, s->prim_channels - base_channel, 3);
-    get_array(&s->gb, s->bitalloc_huffman + base_channel,    s->prim_channels - base_channel, 3);
-
-    /* Get codebooks quantization indexes */
-    if (!base_channel)
-        memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
-    for (j = 1; j < 11; j++)
-        for (i = base_channel; i < s->prim_channels; i++)
-            s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
-
-    /* Get scale factor adjustment */
-    for (j = 0; j < 11; j++)
-        for (i = base_channel; i < s->prim_channels; i++)
-            s->scalefactor_adj[i][j] = 1;
-
-    for (j = 1; j < 11; j++)
-        for (i = base_channel; i < s->prim_channels; i++)
-            if (s->quant_index_huffman[i][j] < thr[j])
-                s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
-
-    if (s->crc_present) {
-        /* Audio header CRC check */
-        get_bits(&s->gb, 16);
-    }
-
-    s->current_subframe    = 0;
-    s->current_subsubframe = 0;
-
-#ifdef TRACE
-    av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes);
-    av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels);
-    for (i = base_channel; i < s->prim_channels; i++) {
-        av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n",
-               s->subband_activity[i]);
-        av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n",
-               s->vq_start_subband[i]);
-        av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n",
-               s->joint_intensity[i]);
-        av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n",
-               s->transient_huffman[i]);
-        av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n",
-               s->scalefactor_huffman[i]);
-        av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n",
-               s->bitalloc_huffman[i]);
-        av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:");
-        for (j = 0; j < 11; j++)
-            av_log(s->avctx, AV_LOG_DEBUG, " %i", s->quant_index_huffman[i][j]);
-        av_log(s->avctx, AV_LOG_DEBUG, "\n");
-        av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");
-        for (j = 0; j < 11; j++)
-            av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]);
-        av_log(s->avctx, AV_LOG_DEBUG, "\n");
-    }
-#endif
-
-    return 0;
-}
-
-static int dca_parse_frame_header(DCAContext *s)
-{
-    init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
-
-    /* Sync code */
-    skip_bits_long(&s->gb, 32);
-
-    /* Frame header */
-    s->frame_type        = get_bits(&s->gb, 1);
-    s->samples_deficit   = get_bits(&s->gb, 5) + 1;
-    s->crc_present       = get_bits(&s->gb, 1);
-    s->sample_blocks     = get_bits(&s->gb, 7) + 1;
-    s->frame_size        = get_bits(&s->gb, 14) + 1;
-    if (s->frame_size < 95)
-        return AVERROR_INVALIDDATA;
-    s->amode             = get_bits(&s->gb, 6);
-    s->sample_rate       = dca_sample_rates[get_bits(&s->gb, 4)];
-    if (!s->sample_rate)
-        return AVERROR_INVALIDDATA;
-    s->bit_rate_index    = get_bits(&s->gb, 5);
-    s->bit_rate          = dca_bit_rates[s->bit_rate_index];
-    if (!s->bit_rate)
-        return AVERROR_INVALIDDATA;
-
-    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, 0);
-}
-
-
-static inline int get_scale(GetBitContext *gb, int level, int value, int log2range)
-{
-    if (level < 5) {
-        /* huffman encoded */
-        value += get_bitalloc(gb, &dca_scalefactor, level);
-        value = av_clip_uintp2(value, log2range);
-    } else if (level < 8) {
-        if (level + 1 > log2range) {
-            skip_bits(gb, level + 1 - log2range);
-            value = get_bits(gb, log2range);
-        } else {
-            value = get_bits(gb, level + 1);
-        }
-    }
-    return value;
-}
-
-static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
-{
-    /* Primary audio coding side information */
-    int j, k;
-
-    if (get_bits_left(&s->gb) < 0)
-        return AVERROR_INVALIDDATA;
-
-    if (!base_channel) {
-        s->subsubframes[s->current_subframe]    = get_bits(&s->gb, 2) + 1;
-        s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3);
-    }
-
-    for (j = base_channel; j < s->prim_channels; j++) {
-        for (k = 0; k < s->subband_activity[j]; k++)
-            s->prediction_mode[j][k] = get_bits(&s->gb, 1);
-    }
-
-    /* Get prediction codebook */
-    for (j = base_channel; j < s->prim_channels; j++) {
-        for (k = 0; k < s->subband_activity[j]; k++) {
-            if (s->prediction_mode[j][k] > 0) {
-                /* (Prediction coefficient VQ address) */
-                s->prediction_vq[j][k] = get_bits(&s->gb, 12);
-            }
-        }
-    }
-
-    /* Bit allocation index */
-    for (j = base_channel; j < s->prim_channels; j++) {
-        for (k = 0; k < s->vq_start_subband[j]; k++) {
-            if (s->bitalloc_huffman[j] == 6)
-                s->bitalloc[j][k] = get_bits(&s->gb, 5);
-            else if (s->bitalloc_huffman[j] == 5)
-                s->bitalloc[j][k] = get_bits(&s->gb, 4);
-            else if (s->bitalloc_huffman[j] == 7) {
-                av_log(s->avctx, AV_LOG_ERROR,
-                       "Invalid bit allocation index\n");
-                return AVERROR_INVALIDDATA;
-            } else {
-                s->bitalloc[j][k] =
-                    get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);
-            }
-
-            if (s->bitalloc[j][k] > 26) {
-                // av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index [%i][%i] too big (%i)\n",
-                //        j, k, s->bitalloc[j][k]);
-                return AVERROR_INVALIDDATA;
-            }
-        }
-    }
-
-    /* Transition mode */
-    for (j = base_channel; j < s->prim_channels; j++) {
-        for (k = 0; k < s->subband_activity[j]; k++) {
-            s->transition_mode[j][k] = 0;
-            if (s->subsubframes[s->current_subframe] > 1 &&
-                k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) {
-                s->transition_mode[j][k] =
-                    get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]);
-            }
-        }
-    }
-
-    if (get_bits_left(&s->gb) < 0)
-        return AVERROR_INVALIDDATA;
-
-    for (j = base_channel; j < s->prim_channels; j++) {
-        const uint32_t *scale_table;
-        int scale_sum, log_size;
-
-        memset(s->scale_factor[j], 0,
-               s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);
-
-        if (s->scalefactor_huffman[j] == 6) {
-            scale_table = scale_factor_quant7;
-            log_size = 7;
-        } else {
-            scale_table = scale_factor_quant6;
-            log_size = 6;
-        }
-
-        /* When huffman coded, only the difference is encoded */
-        scale_sum = 0;
-
-        for (k = 0; k < s->subband_activity[j]; k++) {
-            if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) {
-                scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size);
-                s->scale_factor[j][k][0] = scale_table[scale_sum];
-            }
-
-            if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) {
-                /* Get second scale factor */
-                scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size);
-                s->scale_factor[j][k][1] = scale_table[scale_sum];
-            }
-        }
-    }
-
-    /* Joint subband scale factor codebook select */
-    for (j = base_channel; j < s->prim_channels; j++) {
-        /* Transmitted only if joint subband coding enabled */
-        if (s->joint_intensity[j] > 0)
-            s->joint_huff[j] = get_bits(&s->gb, 3);
-    }
-
-    if (get_bits_left(&s->gb) < 0)
-        return AVERROR_INVALIDDATA;
-
-    /* Scale factors for joint subband coding */
-    for (j = base_channel; j < s->prim_channels; j++) {
-        int source_channel;
-
-        /* Transmitted only if joint subband coding enabled */
-        if (s->joint_intensity[j] > 0) {
-            int scale = 0;
-            source_channel = s->joint_intensity[j] - 1;
-
-            /* When huffman coded, only the difference is encoded
-             * (is this valid as well for joint scales ???) */
-
-            for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) {
-                scale = get_scale(&s->gb, s->joint_huff[j], 64 /* bias */, 7);
-                s->joint_scale_factor[j][k] = scale;    /*joint_scale_table[scale]; */
-            }
-
-            if (!(s->debug_flag & 0x02)) {
-                av_log(s->avctx, AV_LOG_DEBUG,
-                       "Joint stereo coding not supported\n");
-                s->debug_flag |= 0x02;
-            }
-        }
-    }
-
-    /* Stereo downmix coefficients */
-    if (!base_channel && s->prim_channels > 2) {
-        if (s->downmix) {
-            for (j = base_channel; 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;
-            if (am >= FF_ARRAY_ELEMS(dca_default_coeffs)) {
-                av_log(s->avctx, AV_LOG_ERROR,
-                       "Invalid channel mode %d\n", am);
-                return AVERROR_INVALIDDATA;
-            }
-            for (j = base_channel; 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 (!base_channel && s->dynrange)
-        s->dynrange_coef = get_bits(&s->gb, 8);
-
-    /* Side information CRC check word */
-    if (s->crc_present) {
-        get_bits(&s->gb, 16);
-    }
-
-    /*
-     * Primary audio data arrays
-     */
-
-    /* VQ encoded high frequency subbands */
-    for (j = base_channel; j < s->prim_channels; j++)
-        for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
-            /* 1 vector -> 32 samples */
-            s->high_freq_vq[j][k] = get_bits(&s->gb, 10);
-
-    /* Low frequency effect data */
-    if (!base_channel && s->lfe) {
-        /* LFE samples */
-        int lfe_samples = 2 * s->lfe * (4 + block_index);
-        int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);
-        float lfe_scale;
-
-        for (j = lfe_samples; j < lfe_end_sample; j++) {
-            /* Signed 8 bits int */
-            s->lfe_data[j] = get_sbits(&s->gb, 8);
-        }
-
-        /* Scale factor index */
-        skip_bits(&s->gb, 1);
-        s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 7)];
-
-        /* Quantization step size * scale factor */
-        lfe_scale = 0.035 * s->lfe_scale_factor;
-
-        for (j = lfe_samples; j < lfe_end_sample; j++)
-            s->lfe_data[j] *= lfe_scale;
-    }
-
-#ifdef TRACE
-    av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n",
-           s->subsubframes[s->current_subframe]);
-    av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n",
-           s->partial_samples[s->current_subframe]);
-
-    for (j = base_channel; j < s->prim_channels; j++) {
-        av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:");
-        for (k = 0; k < s->subband_activity[j]; k++)
-            av_log(s->avctx, AV_LOG_DEBUG, " %i", s->prediction_mode[j][k]);
-        av_log(s->avctx, AV_LOG_DEBUG, "\n");
-    }
-    for (j = base_channel; j < s->prim_channels; j++) {
-        for (k = 0; k < s->subband_activity[j]; k++)
-            av_log(s->avctx, AV_LOG_DEBUG,
-                   "prediction coefs: %f, %f, %f, %f\n",
-                   (float) adpcm_vb[s->prediction_vq[j][k]][0] / 8192,
-                   (float) adpcm_vb[s->prediction_vq[j][k]][1] / 8192,
-                   (float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192,
-                   (float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192);
-    }
-    for (j = base_channel; j < s->prim_channels; j++) {
-        av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: ");
-        for (k = 0; k < s->vq_start_subband[j]; k++)
-            av_log(s->avctx, AV_LOG_DEBUG, "%2.2i ", s->bitalloc[j][k]);
-        av_log(s->avctx, AV_LOG_DEBUG, "\n");
-    }
-    for (j = base_channel; j < s->prim_channels; j++) {
-        av_log(s->avctx, AV_LOG_DEBUG, "Transition mode:");
-        for (k = 0; k < s->subband_activity[j]; k++)
-            av_log(s->avctx, AV_LOG_DEBUG, " %i", s->transition_mode[j][k]);
-        av_log(s->avctx, AV_LOG_DEBUG, "\n");
-    }
-    for (j = base_channel; j < s->prim_channels; j++) {
-        av_log(s->avctx, AV_LOG_DEBUG, "Scale factor:");
-        for (k = 0; k < s->subband_activity[j]; k++) {
-            if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0)
-                av_log(s->avctx, AV_LOG_DEBUG, " %i", s->scale_factor[j][k][0]);
-            if (k < s->vq_start_subband[j] && s->transition_mode[j][k])
-                av_log(s->avctx, AV_LOG_DEBUG, " %i(t)", s->scale_factor[j][k][1]);
-        }
-        av_log(s->avctx, AV_LOG_DEBUG, "\n");
-    }
-    for (j = base_channel; j < s->prim_channels; j++) {
-        if (s->joint_intensity[j] > 0) {
-            int source_channel = s->joint_intensity[j] - 1;
-            av_log(s->avctx, AV_LOG_DEBUG, "Joint scale factor index:\n");
-            for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++)
-                av_log(s->avctx, AV_LOG_DEBUG, " %i", s->joint_scale_factor[j][k]);
-            av_log(s->avctx, AV_LOG_DEBUG, "\n");
-        }
-    }
-    if (!base_channel && 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 = base_channel; j < s->prim_channels; j++)
-        for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
-            av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]);
-    if (!base_channel && s->lfe) {
-        int lfe_samples = 2 * s->lfe * (4 + block_index);
-        int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);
-
-        av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n");
-        for (j = lfe_samples; j < lfe_end_sample; j++)
-            av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]);
-        av_log(s->avctx, AV_LOG_DEBUG, "\n");
-    }
-#endif
-
-    return 0;
-}
-
-static void qmf_32_subbands(DCAContext *s, int chans,
-                            float samples_in[32][8], float *samples_out,
-                            float scale)
-{
-    const float *prCoeff;
-    int 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;
-
-    for (i = sb_act; i < 32; i++)
-        s->raXin[i] = 0.0;
-
-    /* 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++) {
-            unsigned sign = (i - 1) & 2;
-            uint32_t v    = AV_RN32A(&samples_in[i][subindex]) ^ sign << 30;
-            AV_WN32A(&s->raXin[i], v);
-        }
-
-        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);
-        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)
-{
-    /* 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);
-        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 + c];                                                \
-    u = samples[i + l];                                                \
-    v = samples[i + r];                                                \
-    samples[i]     = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \
-    samples[i+256] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];
-
-#define DOWNMIX_TO_STEREO(op1, op2)             \
-    for (i = 0; i < 256; i++) {                 \
-        op1                                     \
-        op2                                     \
-    }
-
-static void dca_downmix(float *samples, int srcfmt,
-                        int downmix_coef[DCA_PRIM_CHANNELS_MAX][2],
-                        const int8_t *channel_mapping)
-{
-    int c, l, r, sl, sr, s;
-    int i;
-    float t, u, v;
-    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:
-        c = channel_mapping[0] * 256;
-        l = channel_mapping[1] * 256;
-        r = channel_mapping[2] * 256;
-        DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );
-        break;
-    case DCA_2F1R:
-        s = channel_mapping[2] * 256;
-        DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + s, 2, coef), );
-        break;
-    case DCA_3F1R:
-        c = channel_mapping[0] * 256;
-        l = channel_mapping[1] * 256;
-        r = channel_mapping[2] * 256;
-        s = channel_mapping[3] * 256;
-        DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
-                          MIX_REAR1(samples, i + s, 3, coef));
-        break;
-    case DCA_2F2R:
-        sl = channel_mapping[2] * 256;
-        sr = channel_mapping[3] * 256;
-        DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + sl, i + sr, 2, coef), );
-        break;
-    case DCA_3F2R:
-        c  = channel_mapping[0] * 256;
-        l  = channel_mapping[1] * 256;
-        r  = channel_mapping[2] * 256;
-        sl = channel_mapping[3] * 256;
-        sr = channel_mapping[4] * 256;
-        DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
-                          MIX_REAR2(samples, i + sl, i + sr, 3, coef));
-        break;
-    }
-}
-
-
-#ifndef decode_blockcodes
-/* Very compact version of the block code decoder that does not use table
- * look-up but is slightly slower */
-static int decode_blockcode(int code, int levels, 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;
-    }
-
-    return code;
-}
-
-static int decode_blockcodes(int code1, int code2, int levels, int *values)
-{
-    return decode_blockcode(code1, levels, values) |
-           decode_blockcode(code2, levels, values + 4);
-}
-#endif
-
-static const uint8_t abits_sizes[7]  = { 7, 10, 12, 13, 15, 17, 19 };
-static const uint8_t abits_levels[7] = { 3,  5,  7,  9, 13, 17, 25 };
-
-#ifndef int8x8_fmul_int32
-static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)
-{
-    float fscale = scale / 16.0;
-    int i;
-    for (i = 0; i < 8; i++)
-        dst[i] = src[i] * fscale;
-}
-#endif
-
-static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
-{
-    int k, l;
-    int subsubframe = s->current_subsubframe;
-
-    const float *quant_step_table;
-
-    /* FIXME */
-    float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
-    LOCAL_ALIGNED_16(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 = base_channel; k < s->prim_channels; k++) {
-        if (get_bits_left(&s->gb) < 0)
-            return AVERROR_INVALIDDATA;
-
-        for (l = 0; l < s->vq_start_subband[k]; l++) {
-            int m;
-
-            /* Select the mid-tread linear quantizer */
-            int abits = s->bitalloc[k][l];
-
-            float quant_step_size = quant_step_table[abits];
-
-            /*
-             * Determine quantization index code book and its type
-             */
-
-            /* Select quantization index code book */
-            int sel = s->quant_index_huffman[k][abits];
-
-            /*
-             * Extract bits from the bit stream
-             */
-            if (!abits) {
-                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, err;
-
-                        size   = abits_sizes[abits - 1];
-                        levels = abits_levels[abits - 1];
-
-                        block_code1 = get_bits(&s->gb, size);
-                        block_code2 = get_bits(&s->gb, size);
-                        err = decode_blockcodes(block_code1, block_code2,
-                                                levels, block);
-                        if (err) {
-                            av_log(s->avctx, AV_LOG_ERROR,
-                                   "ERROR: block code look-up failed\n");
-                            return AVERROR_INVALIDDATA;
-                        }
-                    } else {
-                        /* no coding */
-                        for (m = 0; m < 8; m++)
-                            block[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->fmt_conv.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 hfvq = s->high_freq_vq[k][l];
-
-            if (!s->debug_flag & 0x01) {
-                av_log(s->avctx, AV_LOG_DEBUG,
-                       "Stream with high frequencies VQ coding\n");
-                s->debug_flag |= 0x01;
-            }
-
-            int8x8_fmul_int32(subband_samples[k][l],
-                              &high_freq_vq[hfvq][subsubframe * 8],
-                              s->scale_factor[k][l][0]);
-        }
-    }
-
-    /* Check for DSYNC after subsubframe */
-    if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {
-        if (0xFFFF == get_bits(&s->gb, 16)) {   /* 0xFFFF */
-#ifdef TRACE
-            av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");
-#endif
-        } else {
-            av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
-        }
-    }
-
-    /* Backup predictor history for adpcm */
-    for (k = base_channel; 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] */);
-    }
-
-    /* Down mixing */
-    if (s->avctx->request_channels == 2 && s->prim_channels > 2) {
-        dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab);
-    }
-
-    /* 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);
-        /* Outputs 20bits pcm samples */
-    }
-
-    return 0;
-}
-
-
-static int dca_subframe_footer(DCAContext *s, int base_channel)
-{
-    int aux_data_count = 0, i;
-
-    /*
-     * Unpack optional information
-     */
-
-    /* presumably optional information only appears in the core? */
-    if (!base_channel) {
-        if (s->timestamp)
-            skip_bits_long(&s->gb, 32);
-
-        if (s->aux_data)
-            aux_data_count = get_bits(&s->gb, 6);
-
-        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 base_channel, int block_index)
-{
-    int ret;
-
-    /* Sanity check */
-    if (s->current_subframe >= s->subframes) {
-        av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
-               s->current_subframe, s->subframes);
-        return AVERROR_INVALIDDATA;
-    }
-
-    if (!s->current_subsubframe) {
-#ifdef TRACE
-        av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
-#endif
-        /* Read subframe header */
-        if ((ret = dca_subframe_header(s, base_channel, block_index)))
-            return ret;
-    }
-
-    /* Read subsubframe */
-#ifdef TRACE
-    av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");
-#endif
-    if ((ret = dca_subsubframe(s, base_channel, block_index)))
-        return ret;
-
-    /* Update state */
-    s->current_subsubframe++;
-    if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {
-        s->current_subsubframe = 0;
-        s->current_subframe++;
-    }
-    if (s->current_subframe >= s->subframes) {
-#ifdef TRACE
-        av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");
-#endif
-        /* Read subframe footer */
-        if ((ret = dca_subframe_footer(s, base_channel)))
-            return ret;
-    }
-
-    return 0;
-}
-
-/**
- * Return the number of channels in an ExSS speaker mask (HD)
- */
-static int dca_exss_mask2count(int mask)
-{
-    /* count bits that mean speaker pairs twice */
-    return av_popcount(mask) +
-           av_popcount(mask & (DCA_EXSS_CENTER_LEFT_RIGHT      |
-                               DCA_EXSS_FRONT_LEFT_RIGHT       |
-                               DCA_EXSS_FRONT_HIGH_LEFT_RIGHT  |
-                               DCA_EXSS_WIDE_LEFT_RIGHT        |
-                               DCA_EXSS_SIDE_LEFT_RIGHT        |
-                               DCA_EXSS_SIDE_HIGH_LEFT_RIGHT   |
-                               DCA_EXSS_SIDE_REAR_LEFT_RIGHT   |
-                               DCA_EXSS_REAR_LEFT_RIGHT        |
-                               DCA_EXSS_REAR_HIGH_LEFT_RIGHT));
-}
-
-/**
- * Skip mixing coefficients of a single mix out configuration (HD)
- */
-static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch)
-{
-    int i;
-
-    for (i = 0; i < channels; i++) {
-        int mix_map_mask = get_bits(gb, out_ch);
-        int num_coeffs = av_popcount(mix_map_mask);
-        skip_bits_long(gb, num_coeffs * 6);
-    }
-}
-
-/**
- * Parse extension substream asset header (HD)
- */
-static int dca_exss_parse_asset_header(DCAContext *s)
-{
-    int header_pos = get_bits_count(&s->gb);
-    int header_size;
-    int channels;
-    int embedded_stereo = 0;
-    int embedded_6ch    = 0;
-    int drc_code_present;
-    int extensions_mask;
-    int i, j;
-
-    if (get_bits_left(&s->gb) < 16)
-        return -1;
-
-    /* We will parse just enough to get to the extensions bitmask with which
-     * we can set the profile value. */
-
-    header_size = get_bits(&s->gb, 9) + 1;
-    skip_bits(&s->gb, 3); // asset index
-
-    if (s->static_fields) {
-        if (get_bits1(&s->gb))
-            skip_bits(&s->gb, 4); // asset type descriptor
-        if (get_bits1(&s->gb))
-            skip_bits_long(&s->gb, 24); // language descriptor
-
-        if (get_bits1(&s->gb)) {
-            /* How can one fit 1024 bytes of text here if the maximum value
-             * for the asset header size field above was 512 bytes? */
-            int text_length = get_bits(&s->gb, 10) + 1;
-            if (get_bits_left(&s->gb) < text_length * 8)
-                return -1;
-            skip_bits_long(&s->gb, text_length * 8); // info text
-        }
-
-        skip_bits(&s->gb, 5); // bit resolution - 1
-        skip_bits(&s->gb, 4); // max sample rate code
-        channels = get_bits(&s->gb, 8) + 1;
-
-        if (get_bits1(&s->gb)) { // 1-to-1 channels to speakers
-            int spkr_remap_sets;
-            int spkr_mask_size = 16;
-            int num_spkrs[7];
-
-            if (channels > 2)
-                embedded_stereo = get_bits1(&s->gb);
-            if (channels > 6)
-                embedded_6ch = get_bits1(&s->gb);
-
-            if (get_bits1(&s->gb)) {
-                spkr_mask_size = (get_bits(&s->gb, 2) + 1) << 2;
-                skip_bits(&s->gb, spkr_mask_size); // spkr activity mask
-            }
-
-            spkr_remap_sets = get_bits(&s->gb, 3);
-
-            for (i = 0; i < spkr_remap_sets; i++) {
-                /* std layout mask for each remap set */
-                num_spkrs[i] = dca_exss_mask2count(get_bits(&s->gb, spkr_mask_size));
-            }
-
-            for (i = 0; i < spkr_remap_sets; i++) {
-                int num_dec_ch_remaps = get_bits(&s->gb, 5) + 1;
-                if (get_bits_left(&s->gb) < 0)
-                    return -1;
-
-                for (j = 0; j < num_spkrs[i]; j++) {
-                    int remap_dec_ch_mask = get_bits_long(&s->gb, num_dec_ch_remaps);
-                    int num_dec_ch = av_popcount(remap_dec_ch_mask);
-                    skip_bits_long(&s->gb, num_dec_ch * 5); // remap codes
-                }
-            }
-
-        } else {
-            skip_bits(&s->gb, 3); // representation type
-        }
-    }
-
-    drc_code_present = get_bits1(&s->gb);
-    if (drc_code_present)
-        get_bits(&s->gb, 8); // drc code
-
-    if (get_bits1(&s->gb))
-        skip_bits(&s->gb, 5); // dialog normalization code
-
-    if (drc_code_present && embedded_stereo)
-        get_bits(&s->gb, 8); // drc stereo code
-
-    if (s->mix_metadata && get_bits1(&s->gb)) {
-        skip_bits(&s->gb, 1); // external mix
-        skip_bits(&s->gb, 6); // post mix gain code
-
-        if (get_bits(&s->gb, 2) != 3) // mixer drc code
-            skip_bits(&s->gb, 3); // drc limit
-        else
-            skip_bits(&s->gb, 8); // custom drc code
-
-        if (get_bits1(&s->gb)) // channel specific scaling
-            for (i = 0; i < s->num_mix_configs; i++)
-                skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6); // scale codes
-        else
-            skip_bits_long(&s->gb, s->num_mix_configs * 6); // scale codes
-
-        for (i = 0; i < s->num_mix_configs; i++) {
-            if (get_bits_left(&s->gb) < 0)
-                return -1;
-            dca_exss_skip_mix_coeffs(&s->gb, channels, s->mix_config_num_ch[i]);
-            if (embedded_6ch)
-                dca_exss_skip_mix_coeffs(&s->gb, 6, s->mix_config_num_ch[i]);
-            if (embedded_stereo)
-                dca_exss_skip_mix_coeffs(&s->gb, 2, s->mix_config_num_ch[i]);
-        }
-    }
-
-    switch (get_bits(&s->gb, 2)) {
-    case 0: extensions_mask = get_bits(&s->gb, 12); break;
-    case 1: extensions_mask = DCA_EXT_EXSS_XLL;     break;
-    case 2: extensions_mask = DCA_EXT_EXSS_LBR;     break;
-    case 3: extensions_mask = 0; /* aux coding */   break;
-    }
-
-    /* not parsed further, we were only interested in the extensions mask */
-
-    if (get_bits_left(&s->gb) < 0)
-        return -1;
-
-    if (get_bits_count(&s->gb) - header_pos > header_size * 8) {
-        av_log(s->avctx, AV_LOG_WARNING, "Asset header size mismatch.\n");
-        return -1;
-    }
-    skip_bits_long(&s->gb, header_pos + header_size * 8 - get_bits_count(&s->gb));
-
-    if (extensions_mask & DCA_EXT_EXSS_XLL)
-        s->profile = FF_PROFILE_DTS_HD_MA;
-    else if (extensions_mask & (DCA_EXT_EXSS_XBR | DCA_EXT_EXSS_X96 |
-                                DCA_EXT_EXSS_XXCH))
-        s->profile = FF_PROFILE_DTS_HD_HRA;
-
-    if (!(extensions_mask & DCA_EXT_CORE))
-        av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n");
-    if ((extensions_mask & DCA_CORE_EXTS) != s->core_ext_mask)
-        av_log(s->avctx, AV_LOG_WARNING,
-               "DTS extensions detection mismatch (%d, %d)\n",
-               extensions_mask & DCA_CORE_EXTS, s->core_ext_mask);
-
-    return 0;
-}
-
-/**
- * Parse extension substream header (HD)
- */
-static void dca_exss_parse_header(DCAContext *s)
-{
-    int ss_index;
-    int blownup;
-    int num_audiop = 1;
-    int num_assets = 1;
-    int active_ss_mask[8];
-    int i, j;
-
-    if (get_bits_left(&s->gb) < 52)
-        return;
-
-    skip_bits(&s->gb, 8); // user data
-    ss_index = get_bits(&s->gb, 2);
-
-    blownup = get_bits1(&s->gb);
-    skip_bits(&s->gb,  8 + 4 * blownup); // header_size
-    skip_bits(&s->gb, 16 + 4 * blownup); // hd_size
-
-    s->static_fields = get_bits1(&s->gb);
-    if (s->static_fields) {
-        skip_bits(&s->gb, 2); // reference clock code
-        skip_bits(&s->gb, 3); // frame duration code
-
-        if (get_bits1(&s->gb))
-            skip_bits_long(&s->gb, 36); // timestamp
-
-        /* a single stream can contain multiple audio assets that can be
-         * combined to form multiple audio presentations */
-
-        num_audiop = get_bits(&s->gb, 3) + 1;
-        if (num_audiop > 1) {
-            av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio presentations.");
-            /* ignore such streams for now */
-            return;
-        }
-
-        num_assets = get_bits(&s->gb, 3) + 1;
-        if (num_assets > 1) {
-            av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio assets.");
-            /* ignore such streams for now */
-            return;
-        }
-
-        for (i = 0; i < num_audiop; i++)
-            active_ss_mask[i] = get_bits(&s->gb, ss_index + 1);
-
-        for (i = 0; i < num_audiop; i++)
-            for (j = 0; j <= ss_index; j++)
-                if (active_ss_mask[i] & (1 << j))
-                    skip_bits(&s->gb, 8); // active asset mask
-
-        s->mix_metadata = get_bits1(&s->gb);
-        if (s->mix_metadata) {
-            int mix_out_mask_size;
-
-            skip_bits(&s->gb, 2); // adjustment level
-            mix_out_mask_size  = (get_bits(&s->gb, 2) + 1) << 2;
-            s->num_mix_configs =  get_bits(&s->gb, 2) + 1;
-
-            for (i = 0; i < s->num_mix_configs; i++) {
-                int mix_out_mask        = get_bits(&s->gb, mix_out_mask_size);
-                s->mix_config_num_ch[i] = dca_exss_mask2count(mix_out_mask);
-            }
-        }
-    }
-
-    for (i = 0; i < num_assets; i++)
-        skip_bits_long(&s->gb, 16 + 4 * blownup);  // asset size
-
-    for (i = 0; i < num_assets; i++) {
-        if (dca_exss_parse_asset_header(s))
-            return;
-    }
-
-    /* not parsed further, we were only interested in the extensions mask
-     * from the asset header */
-}
-
-/**
- * Main frame decoding function
- * FIXME add arguments
- */
-static int dca_decode_frame(AVCodecContext *avctx, void *data,
-                            int *got_frame_ptr, AVPacket *avpkt)
-{
-    const uint8_t *buf = avpkt->data;
-    int buf_size = avpkt->size;
-
-    int lfe_samples;
-    int num_core_channels = 0;
-    int i, ret;
-    float   *samples_flt;
-    int16_t *samples_s16;
-    DCAContext *s = avctx->priv_data;
-    int channels;
-    int core_ss_end;
-
-
-    s->xch_present = 0;
-
-    s->dca_buffer_size = ff_dca_convert_bitstream(buf, buf_size, s->dca_buffer,
-                                                  DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);
-    if (s->dca_buffer_size == AVERROR_INVALIDDATA) {
-        av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
-        return AVERROR_INVALIDDATA;
-    }
-
-    init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
-    if ((ret = dca_parse_frame_header(s)) < 0) {
-        //seems like the frame is corrupt, try with the next one
-        return ret;
-    }
-    //set AVCodec values with parsed data
-    avctx->sample_rate = s->sample_rate;
-    avctx->bit_rate    = s->bit_rate;
-
-    s->profile = FF_PROFILE_DTS;
-
-    for (i = 0; i < (s->sample_blocks / 8); i++) {
-        if ((ret = dca_decode_block(s, 0, i))) {
-            av_log(avctx, AV_LOG_ERROR, "error decoding block\n");
-            return ret;
-        }
-    }
-
-    /* record number of core channels incase less than max channels are requested */
-    num_core_channels = s->prim_channels;
-
-    if (s->ext_coding)
-        s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];
-    else
-        s->core_ext_mask = 0;
-
-    core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;
-
-    /* only scan for extensions if ext_descr was unknown or indicated a
-     * supported XCh extension */
-    if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {
-
-        /* if ext_descr was unknown, clear s->core_ext_mask so that the
-         * extensions scan can fill it up */
-        s->core_ext_mask = FFMAX(s->core_ext_mask, 0);
-
-        /* extensions start at 32-bit boundaries into bitstream */
-        skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
-
-        while (core_ss_end - get_bits_count(&s->gb) >= 32) {
-            uint32_t bits = get_bits_long(&s->gb, 32);
-
-            switch (bits) {
-            case 0x5a5a5a5a: {
-                int ext_amode, xch_fsize;
-
-                s->xch_base_channel = s->prim_channels;
-
-                /* validate sync word using XCHFSIZE field */
-                xch_fsize = show_bits(&s->gb, 10);
-                if ((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&
-                    (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))
-                    continue;
-
-                /* skip length-to-end-of-frame field for the moment */
-                skip_bits(&s->gb, 10);
-
-                s->core_ext_mask |= DCA_EXT_XCH;
-
-                /* extension amode(number of channels in extension) should be 1 */
-                /* AFAIK XCh is not used for more channels */
-                if ((ext_amode = get_bits(&s->gb, 4)) != 1) {
-                    av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"
-                           " supported!\n", ext_amode);
-                    continue;
-                }
-
-                /* much like core primary audio coding header */
-                dca_parse_audio_coding_header(s, s->xch_base_channel);
-
-                for (i = 0; i < (s->sample_blocks / 8); i++)
-                    if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {
-                        av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");
-                        continue;
-                    }
-
-                s->xch_present = 1;
-                break;
-            }
-            case 0x47004a03:
-                /* XXCh: extended channels */
-                /* usually found either in core or HD part in DTS-HD HRA streams,
-                 * but not in DTS-ES which contains XCh extensions instead */
-                s->core_ext_mask |= DCA_EXT_XXCH;
-                break;
-
-            case 0x1d95f262: {
-                int fsize96 = show_bits(&s->gb, 12) + 1;
-                if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)
-                    continue;
-
-                av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n",
-                       get_bits_count(&s->gb));
-                skip_bits(&s->gb, 12);
-                av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);
-                av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));
-
-                s->core_ext_mask |= DCA_EXT_X96;
-                break;
-            }
-            }
-
-            skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
-        }
-    } else {
-        /* no supported extensions, skip the rest of the core substream */
-        skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));
-    }
-
-    if (s->core_ext_mask & DCA_EXT_X96)
-        s->profile = FF_PROFILE_DTS_96_24;
-    else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))
-        s->profile = FF_PROFILE_DTS_ES;
-
-    /* check for ExSS (HD part) */
-    if (s->dca_buffer_size - s->frame_size > 32 &&
-        get_bits_long(&s->gb, 32) == DCA_HD_MARKER)
-        dca_exss_parse_header(s);
-
-    avctx->profile = s->profile;
-
-    channels = s->prim_channels + !!s->lfe;
-
-    if (s->amode < 16) {
-        avctx->channel_layout = dca_core_channel_layout[s->amode];
-
-        if (s->xch_present && (!avctx->request_channels ||
-                               avctx->request_channels > num_core_channels + !!s->lfe)) {
-            avctx->channel_layout |= AV_CH_BACK_CENTER;
-            if (s->lfe) {
-                avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
-                s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode];
-            } else {
-                s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];
-            }
-        } else {
-            channels = num_core_channels + !!s->lfe;
-            s->xch_present = 0; /* disable further xch processing */
-            if (s->lfe) {
-                avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
-                s->channel_order_tab = dca_channel_reorder_lfe[s->amode];
-            } else
-                s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];
-        }
-
-        if (channels > !!s->lfe &&
-            s->channel_order_tab[channels - 1 - !!s->lfe] < 0)
-            return AVERROR_INVALIDDATA;
-
-        if (avctx->request_channels == 2 && s->prim_channels > 2) {
-            channels = 2;
-            s->output = DCA_STEREO;
-            avctx->channel_layout = AV_CH_LAYOUT_STEREO;
-        }
-    } else {
-        av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);
-        return AVERROR_INVALIDDATA;
-    }
-
-
-    /* There is nothing that prevents a dts frame to change channel configuration
-       but Libav 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 (s->is_channels_set == 0) {
-        s->is_channels_set = 1;
-        avctx->channels = channels;
-    }
-    if (avctx->channels != channels) {
-        av_log(avctx, AV_LOG_ERROR, "DCA decoder does not support number of "
-               "channels changing in stream. Skipping frame.\n");
-        return AVERROR_PATCHWELCOME;
-    }
-
-    /* get output buffer */
-    s->frame.nb_samples = 256 * (s->sample_blocks / 8);
-    if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
-        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
-        return ret;
-    }
-    samples_flt = (float *)   s->frame.data[0];
-    samples_s16 = (int16_t *) s->frame.data[0];
-
-    /* filter to get final output */
-    for (i = 0; i < (s->sample_blocks / 8); i++) {
-        dca_filter_channels(s, i);
-
-        /* If this was marked as a DTS-ES stream we need to subtract back- */
-        /* channel from SL & SR to remove matrixed back-channel signal */
-        if ((s->source_pcm_res & 1) && s->xch_present) {
-            float *back_chan = s->samples + s->channel_order_tab[s->xch_base_channel]     * 256;
-            float *lt_chan   = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256;
-            float *rt_chan   = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;
-            s->dsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);
-            s->dsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);
-        }
-
-        if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
-            s->fmt_conv.float_interleave(samples_flt, s->samples_chanptr, 256,
-                                         channels);
-            samples_flt += 256 * channels;
-        } else {
-            s->fmt_conv.float_to_int16_interleave(samples_s16,
-                                                  s->samples_chanptr, 256,
-                                                  channels);
-            samples_s16 += 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];
-
-    *got_frame_ptr    = 1;
-    *(AVFrame *) data = s->frame;
-
-    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();
-
-    ff_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);
-    ff_fmt_convert_init(&s->fmt_conv, avctx);
-
-    for (i = 0; i < DCA_PRIM_CHANNELS_MAX + 1; i++)
-        s->samples_chanptr[i] = s->samples + i * 256;
-
-    if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
-        avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
-        s->scale_bias     = 1.0 / 32768.0;
-    } else {
-        avctx->sample_fmt = AV_SAMPLE_FMT_S16;
-        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;
-    }
-
-    avcodec_get_frame_defaults(&s->frame);
-    avctx->coded_frame = &s->frame;
-
-    return 0;
-}
-
-static av_cold int dca_decode_end(AVCodecContext *avctx)
-{
-    DCAContext *s = avctx->priv_data;
-    ff_mdct_end(&s->imdct);
-    return 0;
-}
-
-static const AVProfile profiles[] = {
-    { FF_PROFILE_DTS,        "DTS"        },
-    { FF_PROFILE_DTS_ES,     "DTS-ES"     },
-    { FF_PROFILE_DTS_96_24,  "DTS 96/24"  },
-    { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },
-    { FF_PROFILE_DTS_HD_MA,  "DTS-HD MA"  },
-    { FF_PROFILE_UNKNOWN },
-};
-
-AVCodec ff_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)"),
-    .capabilities    = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
-    .sample_fmts     = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
-                                                       AV_SAMPLE_FMT_S16,
-                                                       AV_SAMPLE_FMT_NONE },
-    .profiles        = NULL_IF_CONFIG_SMALL(profiles),
+    0, 8000, 16000, 32000, 0, 0, 11025, 22050, 44100, 0, 0,
+    12000, 24000, 48000, 96000, 192000
 };