dca: Rename dca.c ---> dcadec.c

This will allow adding dca.c with tables used from other files.

1 files changed, 1971 insertions, 0 deletions

diff --git a/libavcodec/dcadec.c b/libavcodec/dcadec.c
new file mode 100644
index 0000000000..b37dc49d3f
--- /dev/null
+++ b/libavcodec/dcadec.c
@@ -0,0 +1,1971 @@
+/*
+ * DCA compatible decoder
+ * Copyright (C) 2004 Gildas Bazin
+ * Copyright (C) 2004 Benjamin Zores
+ * Copyright (C) 2006 Benjamin Larsson
+ * Copyright (C) 2007 Konstantin Shishkov
+ *
+ * This file is part of Libav.
+ *
+ * Libav is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * Libav is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <math.h>
+#include <stddef.h>
+#include <stdio.h>
+
+#include "libavutil/common.h"
+#include "libavutil/float_dsp.h"
+#include "libavutil/intmath.h"
+#include "libavutil/intreadwrite.h"
+#include "libavutil/mathematics.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
+    AVFloatDSPContext fdsp;
+    FFTContext imdct;
+    SynthFilterContext synth;
+    DCADSPContext dcadsp;
+    FmtConvertContext fmt_conv;
+} DCAContext;
+
+static const uint16_t dca_vlc_offs[] = {
+        0,   512,   640,   768,  1282,  1794,  2436,  3080,  3770,  4454,  5364,
+     5372,  5380,  5388,  5392,  5396,  5412,  5420,  5428,  5460,  5492,  5508,
+     5572,  5604,  5668,  5796,  5860,  5892,  6412,  6668,  6796,  7308,  7564,
+     7820,  8076,  8620,  9132,  9388,  9910, 10166, 10680, 11196, 11726, 12240,
+    12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264,
+    18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622,
+};
+
+static av_cold void dca_init_vlcs(void)
+{
+    static int vlcs_initialized = 0;
+    int i, j, c = 14;
+    static VLC_TYPE dca_table[23622][2];
+
+    if (vlcs_initialized)
+        return;
+
+    dca_bitalloc_index.offset = 1;
+    dca_bitalloc_index.wrap = 2;
+    for (i = 0; i < 5; i++) {
+        dca_bitalloc_index.vlc[i].table = &dca_table[dca_vlc_offs[i]];
+        dca_bitalloc_index.vlc[i].table_allocated = dca_vlc_offs[i + 1] - dca_vlc_offs[i];
+        init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,
+                 bitalloc_12_bits[i], 1, 1,
+                 bitalloc_12_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
+    }
+    dca_scalefactor.offset = -64;
+    dca_scalefactor.wrap = 2;
+    for (i = 0; i < 5; i++) {
+        dca_scalefactor.vlc[i].table = &dca_table[dca_vlc_offs[i + 5]];
+        dca_scalefactor.vlc[i].table_allocated = dca_vlc_offs[i + 6] - dca_vlc_offs[i + 5];
+        init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
+                 scales_bits[i], 1, 1,
+                 scales_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
+    }
+    dca_tmode.offset = 0;
+    dca_tmode.wrap = 1;
+    for (i = 0; i < 4; i++) {
+        dca_tmode.vlc[i].table = &dca_table[dca_vlc_offs[i + 10]];
+        dca_tmode.vlc[i].table_allocated = dca_vlc_offs[i + 11] - dca_vlc_offs[i + 10];
+        init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
+                 tmode_bits[i], 1, 1,
+                 tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
+    }
+
+    for (i = 0; i < 10; i++)
+        for (j = 0; j < 7; j++) {
+            if (!bitalloc_codes[i][j])
+                break;
+            dca_smpl_bitalloc[i + 1].offset                 = bitalloc_offsets[i];
+            dca_smpl_bitalloc[i + 1].wrap                   = 1 + (j > 4);
+            dca_smpl_bitalloc[i + 1].vlc[j].table           = &dca_table[dca_vlc_offs[c]];
+            dca_smpl_bitalloc[i + 1].vlc[j].table_allocated = dca_vlc_offs[c + 1] - dca_vlc_offs[c];
+
+            init_vlc(&dca_smpl_bitalloc[i + 1].vlc[j], bitalloc_maxbits[i][j],
+                     bitalloc_sizes[i],
+                     bitalloc_bits[i][j], 1, 1,
+                     bitalloc_codes[i][j], 2, 2, INIT_VLC_USE_NEW_STATIC);
+            c++;
+        }
+    vlcs_initialized = 1;
+}
+
+static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
+{
+    while (len--)
+        *dst++ = get_bits(gb, bits);
+}
+
+static int dca_parse_audio_coding_header(DCAContext *s, int base_channel)
+{
+    int i, j;
+    static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
+    static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
+    static const int thr[11]    = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
+
+    s->total_channels = get_bits(&s->gb, 3) + 1 + base_channel;
+    s->prim_channels  = s->total_channels;
+
+    if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)
+        s->prim_channels = DCA_PRIM_CHANNELS_MAX;
+
+
+    for (i = base_channel; i < s->prim_channels; i++) {
+        s->subband_activity[i] = get_bits(&s->gb, 5) + 2;
+        if (s->subband_activity[i] > DCA_SUBBANDS)
+            s->subband_activity[i] = DCA_SUBBANDS;
+    }
+    for (i = base_channel; i < s->prim_channels; i++) {
+        s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
+        if (s->vq_start_subband[i] > DCA_SUBBANDS)
+            s->vq_start_subband[i] = DCA_SUBBANDS;
+    }
+    get_array(&s->gb, s->joint_intensity + base_channel,     s->prim_channels - base_channel, 3);
+    get_array(&s->gb, s->transient_huffman + base_channel,   s->prim_channels - base_channel, 2);
+    get_array(&s->gb, s->scalefactor_huffman + base_channel, s->prim_channels - base_channel, 3);
+    get_array(&s->gb, s->bitalloc_huffman + base_channel,    s->prim_channels - base_channel, 3);
+
+    /* Get codebooks quantization indexes */
+    if (!base_channel)
+        memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
+    for (j = 1; j < 11; j++)
+        for (i = base_channel; i < s->prim_channels; i++)
+            s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
+
+    /* Get scale factor adjustment */
+    for (j = 0; j < 11; j++)
+        for (i = base_channel; i < s->prim_channels; i++)
+            s->scalefactor_adj[i][j] = 1;
+
+    for (j = 1; j < 11; j++)
+        for (i = base_channel; i < s->prim_channels; i++)
+            if (s->quant_index_huffman[i][j] < thr[j])
+                s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
+
+    if (s->crc_present) {
+        /* Audio header CRC check */
+        get_bits(&s->gb, 16);
+    }
+
+    s->current_subframe    = 0;
+    s->current_subsubframe = 0;
+
+#ifdef TRACE
+    av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes);
+    av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels);
+    for (i = base_channel; i < s->prim_channels; i++) {
+        av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n",
+               s->subband_activity[i]);
+        av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n",
+               s->vq_start_subband[i]);
+        av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n",
+               s->joint_intensity[i]);
+        av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n",
+               s->transient_huffman[i]);
+        av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n",
+               s->scalefactor_huffman[i]);
+        av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n",
+               s->bitalloc_huffman[i]);
+        av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:");
+        for (j = 0; j < 11; j++)
+            av_log(s->avctx, AV_LOG_DEBUG, " %i", s->quant_index_huffman[i][j]);
+        av_log(s->avctx, AV_LOG_DEBUG, "\n");
+        av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");
+        for (j = 0; j < 11; j++)
+            av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]);
+        av_log(s->avctx, AV_LOG_DEBUG, "\n");
+    }
+#endif
+
+    return 0;
+}
+
+static int dca_parse_frame_header(DCAContext *s)
+{
+    init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
+
+    /* Sync code */
+    skip_bits_long(&s->gb, 32);
+
+    /* Frame header */
+    s->frame_type        = get_bits(&s->gb, 1);
+    s->samples_deficit   = get_bits(&s->gb, 5) + 1;
+    s->crc_present       = get_bits(&s->gb, 1);
+    s->sample_blocks     = get_bits(&s->gb, 7) + 1;
+    s->frame_size        = get_bits(&s->gb, 14) + 1;
+    if (s->frame_size < 95)
+        return AVERROR_INVALIDDATA;
+    s->amode             = get_bits(&s->gb, 6);
+    s->sample_rate       = 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(value, 0, (1 << log2range) - 1);
+    } else if (level < 8) {
+        if (level + 1 > log2range) {
+            skip_bits(gb, level + 1 - log2range);
+            value = get_bits(gb, log2range);
+        } else {
+            value = get_bits(gb, level + 1);
+        }
+    }
+    return value;
+}
+
+static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
+{
+    /* Primary audio coding side information */
+    int j, k;
+
+    if (get_bits_left(&s->gb) < 0)
+        return AVERROR_INVALIDDATA;
+
+    if (!base_channel) {
+        s->subsubframes[s->current_subframe]    = get_bits(&s->gb, 2) + 1;
+        s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3);
+    }
+
+    for (j = base_channel; j < s->prim_channels; j++) {
+        for (k = 0; k < s->subband_activity[j]; k++)
+            s->prediction_mode[j][k] = get_bits(&s->gb, 1);
+    }
+
+    /* Get prediction codebook */
+    for (j = base_channel; j < s->prim_channels; j++) {
+        for (k = 0; k < s->subband_activity[j]; k++) {
+            if (s->prediction_mode[j][k] > 0) {
+                /* (Prediction coefficient VQ address) */
+                s->prediction_vq[j][k] = get_bits(&s->gb, 12);
+            }
+        }
+    }
+
+    /* Bit allocation index */
+    for (j = base_channel; j < s->prim_channels; j++) {
+        for (k = 0; k < s->vq_start_subband[j]; k++) {
+            if (s->bitalloc_huffman[j] == 6)
+                s->bitalloc[j][k] = get_bits(&s->gb, 5);
+            else if (s->bitalloc_huffman[j] == 5)
+                s->bitalloc[j][k] = get_bits(&s->gb, 4);
+            else if (s->bitalloc_huffman[j] == 7) {
+                av_log(s->avctx, AV_LOG_ERROR,
+                       "Invalid bit allocation index\n");
+                return AVERROR_INVALIDDATA;
+            } else {
+                s->bitalloc[j][k] =
+                    get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);
+            }
+
+            if (s->bitalloc[j][k] > 26) {
+                // av_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->fdsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);
+            s->fdsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);
+        }
+
+        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();
+
+    avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
+    ff_mdct_init(&s->imdct, 6, 1, 1.0);
+    ff_synth_filter_init(&s->synth);
+    ff_dcadsp_init(&s->dcadsp);
+    ff_fmt_convert_init(&s->fmt_conv, avctx);
+
+    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),
+};