DCA decoder

Originally committed as revision 8141 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 1322 insertions, 0 deletions

diff --git a/libavcodec/dca.c b/libavcodec/dca.c
new file mode 100644
index 0000000000..a57dcdc442
--- /dev/null
+++ b/libavcodec/dca.c
@@ -0,0 +1,1322 @@
+/*
+ * DCA compatible decoder
+ * Copyright (C) 2004 Gildas Bazin
+ * Copyright (C) 2004 Benjamin Zores
+ * Copyright (C) 2006 Benjamin Larsson
+ * Copyright (C) 2007 Konstantin Shishkov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file dca.c
+ */
+
+#include <math.h>
+#include <stddef.h>
+#include <stdio.h>
+
+#include "avcodec.h"
+#include "dsputil.h"
+#include "bitstream.h"
+#include "dcadata.h"
+#include "dcahuff.h"
+#include "parser.h"
+
+/** DCA syncwords, also used for bitstream type detection */
+//@{
+#define DCA_MARKER_RAW_BE 0x7FFE8001
+#define DCA_MARKER_RAW_LE 0xFE7F0180
+#define DCA_MARKER_14B_BE 0x1FFFE800
+#define DCA_MARKER_14B_LE 0xFF1F00E8
+//@}
+
+//#define TRACE
+
+#define DCA_PRIM_CHANNELS_MAX (5)
+#define DCA_SUBBANDS (32)
+#define DCA_ABITS_MAX (32)      /* Should be 28 */
+#define DCA_SUBSUBFAMES_MAX (4)
+#define DCA_LFE_MAX (3)
+
+enum DCAMode {
+    DCA_MONO = 0,
+    DCA_CHANNEL,
+    DCA_STEREO,
+    DCA_STEREO_SUMDIFF,
+    DCA_STEREO_TOTAL,
+    DCA_3F,
+    DCA_2F1R,
+    DCA_3F1R,
+    DCA_2F2R,
+    DCA_3F2R,
+    DCA_4F2R
+};
+
+#define DCA_DOLBY 101           /* FIXME */
+
+#define DCA_CHANNEL_BITS 6
+#define DCA_CHANNEL_MASK 0x3F
+
+#define DCA_LFE 0x80
+
+#define HEADER_SIZE 14
+#define CONVERT_BIAS 384
+
+#define DCA_MAX_FRAME_SIZE 16383
+
+/** 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
+
+/** Pre-calculated cosine modulation coefs for the QMF */
+static float cos_mod[544];
+
+static int av_always_inline get_bitalloc(GetBitContext *gb, BitAlloc *ba, int idx)
+{
+    return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + ba->offset;
+}
+
+typedef struct {
+    AVCodecContext *avctx;
+    /* Frame header */
+    int frame_type;             ///< type of the current frame
+    int samples_deficit;        ///< deficit sample count
+    int crc_present;            ///< crc is present in the bitstream
+    int sample_blocks;          ///< number of PCM sample blocks
+    int frame_size;             ///< primary frame byte size
+    int amode;                  ///< audio channels arrangement
+    int sample_rate;            ///< audio sampling rate
+    int bit_rate;               ///< transmission bit rate
+
+    int 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 prim_channels;          ///< number of primary audio channels
+    int subband_activity[DCA_PRIM_CHANNELS_MAX];    ///< subband activity count
+    int vq_start_subband[DCA_PRIM_CHANNELS_MAX];    ///< high frequency vq start subband
+    int joint_intensity[DCA_PRIM_CHANNELS_MAX];     ///< joint intensity coding index
+    int transient_huffman[DCA_PRIM_CHANNELS_MAX];   ///< transient mode code book
+    int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book
+    int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX];    ///< bit allocation quantizer select
+    int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select
+    float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX];   ///< scale factor adjustment
+
+    /* Primary audio coding side information */
+    int subsubframes;           ///< number of subsubframes
+    int partial_samples;        ///< partial subsubframe samples count
+    int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];    ///< prediction mode (ADPCM used or not)
+    int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];      ///< prediction VQ coefs
+    int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];           ///< bit allocation index
+    int transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];    ///< transition mode (transients)
+    int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2];    ///< scale factors (2 if transient)
+    int joint_huff[DCA_PRIM_CHANNELS_MAX];                       ///< joint subband scale factors codebook
+    int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors
+    int downmix_coef[DCA_PRIM_CHANNELS_MAX][2];                  ///< stereo downmix coefficients
+    int dynrange_coef;                                           ///< dynamic range coefficient
+
+    int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];       ///< VQ encoded high frequency subbands
+
+    float lfe_data[2 * DCA_SUBSUBFAMES_MAX * DCA_LFE_MAX *
+                   2 /*history */ ];    ///< Low frequency effect data
+    int lfe_scale_factor;
+
+    /* Subband samples history (for ADPCM) */
+    float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
+    float subband_fir_hist[DCA_PRIM_CHANNELS_MAX][512];
+    float subband_fir_noidea[DCA_PRIM_CHANNELS_MAX][64];
+
+    int output;                 ///< type of output
+    int bias;                   ///< output bias
+
+    DECLARE_ALIGNED_16(float, samples[1536]);  /* 6 * 256 = 1536, might only need 5 */
+    DECLARE_ALIGNED_16(int16_t, tsamples[1536]);
+
+    uint8_t dca_buffer[DCA_MAX_FRAME_SIZE];
+    int dca_buffer_size;        ///< how much data is in the dca_buffer
+
+    GetBitContext gb;
+    /* Current position in DCA frame */
+    int current_subframe;
+    int current_subsubframe;
+
+    int debug_flag;             ///< used for suppressing repeated error messages output
+    DSPContext dsp;
+} DCAContext;
+
+static void dca_init_vlcs()
+{
+    static int vlcs_inited = 0;
+    int i, j;
+
+    if (vlcs_inited)
+        return;
+
+    dca_bitalloc_index.offset = 1;
+    dca_bitalloc_index.wrap = 1;
+    for (i = 0; i < 5; i++)
+        init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,
+                 bitalloc_12_bits[i], 1, 1,
+                 bitalloc_12_codes[i], 2, 2, 1);
+    dca_scalefactor.offset = -64;
+    dca_scalefactor.wrap = 2;
+    for (i = 0; i < 5; i++)
+        init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
+                 scales_bits[i], 1, 1,
+                 scales_codes[i], 2, 2, 1);
+    dca_tmode.offset = 0;
+    dca_tmode.wrap = 1;
+    for (i = 0; i < 4; i++)
+        init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
+                 tmode_bits[i], 1, 1,
+                 tmode_codes[i], 2, 2, 1);
+
+    for(i = 0; i < 10; i++)
+        for(j = 0; j < 7; j++){
+            if(!bitalloc_codes[i][j]) break;
+            dca_smpl_bitalloc[i+1].offset = bitalloc_offsets[i];
+            dca_smpl_bitalloc[i+1].wrap = 1 + (j > 4);
+            init_vlc(&dca_smpl_bitalloc[i+1].vlc[j], bitalloc_maxbits[i][j],
+                     bitalloc_sizes[i],
+                     bitalloc_bits[i][j], 1, 1,
+                     bitalloc_codes[i][j], 2, 2, 1);
+        }
+    vlcs_inited = 1;
+}
+
+static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
+{
+    while(len--)
+        *dst++ = get_bits(gb, bits);
+}
+
+static int dca_parse_frame_header(DCAContext * s)
+{
+    int i, j;
+    static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
+    static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
+    static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
+
+    s->bias = CONVERT_BIAS;
+
+    init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
+
+    /* Sync code */
+    get_bits(&s->gb, 32);
+
+    /* Frame header */
+    s->frame_type        = get_bits(&s->gb, 1);
+    s->samples_deficit   = get_bits(&s->gb, 5) + 1;
+    s->crc_present       = get_bits(&s->gb, 1);
+    s->sample_blocks     = get_bits(&s->gb, 7) + 1;
+    s->frame_size        = get_bits(&s->gb, 14) + 1;
+    if (s->frame_size < 95)
+        return -1;
+    s->amode             = get_bits(&s->gb, 6);
+    s->sample_rate       = dca_sample_rates[get_bits(&s->gb, 4)];
+    if (!s->sample_rate)
+        return -1;
+    s->bit_rate          = dca_bit_rates[get_bits(&s->gb, 5)];
+    if (!s->bit_rate)
+        return -1;
+
+    s->downmix           = get_bits(&s->gb, 1);
+    s->dynrange          = get_bits(&s->gb, 1);
+    s->timestamp         = get_bits(&s->gb, 1);
+    s->aux_data          = get_bits(&s->gb, 1);
+    s->hdcd              = get_bits(&s->gb, 1);
+    s->ext_descr         = get_bits(&s->gb, 3);
+    s->ext_coding        = get_bits(&s->gb, 1);
+    s->aspf              = get_bits(&s->gb, 1);
+    s->lfe               = get_bits(&s->gb, 2);
+    s->predictor_history = get_bits(&s->gb, 1);
+
+    /* TODO: check CRC */
+    if (s->crc_present)
+        s->header_crc    = get_bits(&s->gb, 16);
+
+    s->multirate_inter   = get_bits(&s->gb, 1);
+    s->version           = get_bits(&s->gb, 4);
+    s->copy_history      = get_bits(&s->gb, 2);
+    s->source_pcm_res    = get_bits(&s->gb, 3);
+    s->front_sum         = get_bits(&s->gb, 1);
+    s->surround_sum      = get_bits(&s->gb, 1);
+    s->dialog_norm       = get_bits(&s->gb, 4);
+
+    /* FIXME: channels mixing levels */
+    s->output = DCA_STEREO;
+
+#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 (%i Hz)\n",
+           s->sample_rate, dca_sample_rates[s->sample_rate]);
+    av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i (%i bits/s)\n",
+           s->bit_rate, dca_bit_rates[s->bit_rate]);
+    av_log(s->avctx, AV_LOG_DEBUG, "downmix: %i\n", s->downmix);
+    av_log(s->avctx, AV_LOG_DEBUG, "dynrange: %i\n", s->dynrange);
+    av_log(s->avctx, AV_LOG_DEBUG, "timestamp: %i\n", s->timestamp);
+    av_log(s->avctx, AV_LOG_DEBUG, "aux_data: %i\n", s->aux_data);
+    av_log(s->avctx, AV_LOG_DEBUG, "hdcd: %i\n", s->hdcd);
+    av_log(s->avctx, AV_LOG_DEBUG, "ext descr: %i\n", s->ext_descr);
+    av_log(s->avctx, AV_LOG_DEBUG, "ext coding: %i\n", s->ext_coding);
+    av_log(s->avctx, AV_LOG_DEBUG, "aspf: %i\n", s->aspf);
+    av_log(s->avctx, AV_LOG_DEBUG, "lfe: %i\n", s->lfe);
+    av_log(s->avctx, AV_LOG_DEBUG, "predictor history: %i\n",
+           s->predictor_history);
+    av_log(s->avctx, AV_LOG_DEBUG, "header crc: %i\n", s->header_crc);
+    av_log(s->avctx, AV_LOG_DEBUG, "multirate inter: %i\n",
+           s->multirate_inter);
+    av_log(s->avctx, AV_LOG_DEBUG, "version number: %i\n", s->version);
+    av_log(s->avctx, AV_LOG_DEBUG, "copy history: %i\n", s->copy_history);
+    av_log(s->avctx, AV_LOG_DEBUG,
+           "source pcm resolution: %i (%i bits/sample)\n",
+           s->source_pcm_res, dca_bits_per_sample[s->source_pcm_res]);
+    av_log(s->avctx, AV_LOG_DEBUG, "front sum: %i\n", s->front_sum);
+    av_log(s->avctx, AV_LOG_DEBUG, "surround sum: %i\n", s->surround_sum);
+    av_log(s->avctx, AV_LOG_DEBUG, "dialog norm: %i\n", s->dialog_norm);
+    av_log(s->avctx, AV_LOG_DEBUG, "\n");
+#endif
+
+    /* Primary audio coding header */
+    s->subframes         = get_bits(&s->gb, 4) + 1;
+    s->prim_channels     = get_bits(&s->gb, 3) + 1;
+
+
+    for (i = 0; i < s->prim_channels; i++) {
+        s->subband_activity[i] = get_bits(&s->gb, 5) + 2;
+        if (s->subband_activity[i] > DCA_SUBBANDS)
+            s->subband_activity[i] = DCA_SUBBANDS;
+    }
+    for (i = 0; i < s->prim_channels; i++) {
+        s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
+        if (s->vq_start_subband[i] > DCA_SUBBANDS)
+            s->vq_start_subband[i] = DCA_SUBBANDS;
+    }
+    get_array(&s->gb, s->joint_intensity,     s->prim_channels, 3);
+    get_array(&s->gb, s->transient_huffman,   s->prim_channels, 2);
+    get_array(&s->gb, s->scalefactor_huffman, s->prim_channels, 3);
+    get_array(&s->gb, s->bitalloc_huffman,    s->prim_channels, 3);
+
+    /* Get codebooks quantization indexes */
+    memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
+    for (j = 1; j < 11; j++)
+        for (i = 0; i < s->prim_channels; i++)
+            s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
+
+    /* Get scale factor adjustment */
+    for (j = 0; j < 11; j++)
+        for (i = 0; i < s->prim_channels; i++)
+            s->scalefactor_adj[i][j] = 1;
+
+    for (j = 1; j < 11; j++)
+        for (i = 0; i < s->prim_channels; i++)
+            if (s->quant_index_huffman[i][j] < thr[j])
+                s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
+
+    if (s->crc_present) {
+        /* Audio header CRC check */
+        get_bits(&s->gb, 16);
+    }
+
+    s->current_subframe = 0;
+    s->current_subsubframe = 0;
+
+#ifdef TRACE
+    av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes);
+    av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels);
+    for(i = 0; i < s->prim_channels; i++){
+        av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", s->subband_activity[i]);
+        av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", s->vq_start_subband[i]);
+        av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", s->joint_intensity[i]);
+        av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", s->transient_huffman[i]);
+        av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", s->scalefactor_huffman[i]);
+        av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", s->bitalloc_huffman[i]);
+        av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:");
+        for (j = 0; j < 11; j++)
+            av_log(s->avctx, AV_LOG_DEBUG, " %i",
+                   s->quant_index_huffman[i][j]);
+        av_log(s->avctx, AV_LOG_DEBUG, "\n");
+        av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");
+        for (j = 0; j < 11; j++)
+            av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]);
+        av_log(s->avctx, AV_LOG_DEBUG, "\n");
+    }
+#endif
+
+    return 0;
+}
+
+
+static inline int get_scale(GetBitContext *gb, int level, int index, int value)
+{
+   if (level < 5) {
+       /* huffman encoded */
+       value += get_bitalloc(gb, &dca_scalefactor, index);
+   } else if(level < 8)
+       value = get_bits(gb, level + 1);
+   return value;
+}
+
+static int dca_subframe_header(DCAContext * s)
+{
+    /* Primary audio coding side information */
+    int j, k;
+
+    s->subsubframes = get_bits(&s->gb, 2) + 1;
+    s->partial_samples = get_bits(&s->gb, 3);
+    for (j = 0; j < s->prim_channels; j++) {
+        for (k = 0; k < s->subband_activity[j]; k++)
+            s->prediction_mode[j][k] = get_bits(&s->gb, 1);
+    }
+
+    /* Get prediction codebook */
+    for (j = 0; j < s->prim_channels; j++) {
+        for (k = 0; k < s->subband_activity[j]; k++) {
+            if (s->prediction_mode[j][k] > 0) {
+                /* (Prediction coefficient VQ address) */
+                s->prediction_vq[j][k] = get_bits(&s->gb, 12);
+            }
+        }
+    }
+
+    /* Bit allocation index */
+    for (j = 0; j < s->prim_channels; j++) {
+        for (k = 0; k < s->vq_start_subband[j]; k++) {
+            if (s->bitalloc_huffman[j] == 6)
+                s->bitalloc[j][k] = get_bits(&s->gb, 5);
+            else if (s->bitalloc_huffman[j] == 5)
+                s->bitalloc[j][k] = get_bits(&s->gb, 4);
+            else {
+                s->bitalloc[j][k] =
+                    get_bitalloc(&s->gb, &dca_bitalloc_index, j);
+            }
+
+            if (s->bitalloc[j][k] > 26) {
+//                 av_log(s->avctx,AV_LOG_DEBUG,"bitalloc index [%i][%i] too big (%i)\n",
+//                          j, k, s->bitalloc[j][k]);
+                return -1;
+            }
+        }
+    }
+
+    /* Transition mode */
+    for (j = 0; j < s->prim_channels; j++) {
+        for (k = 0; k < s->subband_activity[j]; k++) {
+            s->transition_mode[j][k] = 0;
+            if (s->subsubframes > 1 &&
+                k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) {
+                s->transition_mode[j][k] =
+                    get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]);
+            }
+        }
+    }
+
+    for (j = 0; j < s->prim_channels; j++) {
+        uint32_t *scale_table;
+        int scale_sum;
+
+        memset(s->scale_factor[j], 0, s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);
+
+        if (s->scalefactor_huffman[j] == 6)
+            scale_table = (uint32_t *) scale_factor_quant7;
+        else
+            scale_table = (uint32_t *) scale_factor_quant6;
+
+        /* When huffman coded, only the difference is encoded */
+        scale_sum = 0;
+
+        for (k = 0; k < s->subband_activity[j]; k++) {
+            if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) {
+                scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], j, scale_sum);
+                s->scale_factor[j][k][0] = scale_table[scale_sum];
+            }
+
+            if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) {
+                /* Get second scale factor */
+                scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], j, scale_sum);
+                s->scale_factor[j][k][1] = scale_table[scale_sum];
+            }
+        }
+    }
+
+    /* Joint subband scale factor codebook select */
+    for (j = 0; j < s->prim_channels; j++) {
+        /* Transmitted only if joint subband coding enabled */
+        if (s->joint_intensity[j] > 0)
+            s->joint_huff[j] = get_bits(&s->gb, 3);
+    }
+
+    /* Scale factors for joint subband coding */
+    for (j = 0; j < s->prim_channels; j++) {
+        int source_channel;
+
+        /* Transmitted only if joint subband coding enabled */
+        if (s->joint_intensity[j] > 0) {
+            int scale = 0;
+            source_channel = s->joint_intensity[j] - 1;
+
+            /* When huffman coded, only the difference is encoded
+             * (is this valid as well for joint scales ???) */
+
+            for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) {
+                scale = get_scale(&s->gb, s->joint_huff[j], j, 0);
+                scale += 64;    /* bias */
+                s->joint_scale_factor[j][k] = scale;    /*joint_scale_table[scale]; */
+            }
+
+            if (!s->debug_flag & 0x02) {
+                av_log(s->avctx, AV_LOG_DEBUG,
+                       "Joint stereo coding not supported\n");
+                s->debug_flag |= 0x02;
+            }
+        }
+    }
+
+    /* Stereo downmix coefficients */
+    if (s->prim_channels > 2 && s->downmix) {
+        for (j = 0; j < s->prim_channels; j++) {
+            s->downmix_coef[j][0] = get_bits(&s->gb, 7);
+            s->downmix_coef[j][1] = get_bits(&s->gb, 7);
+        }
+    }
+
+    /* Dynamic range coefficient */
+    if (s->dynrange)
+        s->dynrange_coef = get_bits(&s->gb, 8);
+
+    /* Side information CRC check word */
+    if (s->crc_present) {
+        get_bits(&s->gb, 16);
+    }
+
+    /*
+     * Primary audio data arrays
+     */
+
+    /* VQ encoded high frequency subbands */
+    for (j = 0; j < s->prim_channels; j++)
+        for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
+            /* 1 vector -> 32 samples */
+            s->high_freq_vq[j][k] = get_bits(&s->gb, 10);
+
+    /* Low frequency effect data */
+    if (s->lfe) {
+        /* LFE samples */
+        int lfe_samples = 2 * s->lfe * s->subsubframes;
+        float lfe_scale;
+
+        for (j = lfe_samples; j < lfe_samples * 2; j++) {
+            /* Signed 8 bits int */
+            s->lfe_data[j] = get_sbits(&s->gb, 8);
+        }
+
+        /* Scale factor index */
+        s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 8)];
+
+        /* Quantization step size * scale factor */
+        lfe_scale = 0.035 * s->lfe_scale_factor;
+
+        for (j = lfe_samples; j < lfe_samples * 2; j++)
+            s->lfe_data[j] *= lfe_scale;
+    }
+
+#ifdef TRACE
+    av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n", s->subsubframes);
+    av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n",
+           s->partial_samples);
+    for (j = 0; j < s->prim_channels; j++) {
+        av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:");
+        for (k = 0; k < s->subband_activity[j]; k++)
+            av_log(s->avctx, AV_LOG_DEBUG, " %i", s->prediction_mode[j][k]);
+        av_log(s->avctx, AV_LOG_DEBUG, "\n");
+    }
+    for (j = 0; j < s->prim_channels; j++) {
+        for (k = 0; k < s->subband_activity[j]; k++)
+                av_log(s->avctx, AV_LOG_DEBUG,
+                       "prediction coefs: %f, %f, %f, %f\n",
+                       (float) adpcm_vb[s->prediction_vq[j][k]][0] / 8192,
+                       (float) adpcm_vb[s->prediction_vq[j][k]][1] / 8192,
+                       (float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192,
+                       (float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192);
+    }
+    for (j = 0; j < s->prim_channels; j++) {
+        av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: ");
+        for (k = 0; k < s->vq_start_subband[j]; k++)
+            av_log(s->avctx, AV_LOG_DEBUG, "%2.2i ", s->bitalloc[j][k]);
+        av_log(s->avctx, AV_LOG_DEBUG, "\n");
+    }
+    for (j = 0; j < s->prim_channels; j++) {
+        av_log(s->avctx, AV_LOG_DEBUG, "Transition mode:");
+        for (k = 0; k < s->subband_activity[j]; k++)
+            av_log(s->avctx, AV_LOG_DEBUG, " %i", s->transition_mode[j][k]);
+        av_log(s->avctx, AV_LOG_DEBUG, "\n");
+    }
+    for (j = 0; j < s->prim_channels; j++) {
+        av_log(s->avctx, AV_LOG_DEBUG, "Scale factor:");
+        for (k = 0; k < s->subband_activity[j]; k++) {
+            if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0)
+                av_log(s->avctx, AV_LOG_DEBUG, " %i", s->scale_factor[j][k][0]);
+            if (k < s->vq_start_subband[j] && s->transition_mode[j][k])
+                av_log(s->avctx, AV_LOG_DEBUG, " %i(t)", s->scale_factor[j][k][1]);
+        }
+        av_log(s->avctx, AV_LOG_DEBUG, "\n");
+    }
+    for (j = 0; j < s->prim_channels; j++) {
+        if (s->joint_intensity[j] > 0) {
+            av_log(s->avctx, AV_LOG_DEBUG, "Joint scale factor index:\n");
+            for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++)
+                av_log(s->avctx, AV_LOG_DEBUG, " %i", s->joint_scale_factor[j][k]);
+            av_log(s->avctx, AV_LOG_DEBUG, "\n");
+        }
+    }
+    if (s->prim_channels > 2 && s->downmix) {
+        av_log(s->avctx, AV_LOG_DEBUG, "Downmix coeffs:\n");
+        for (j = 0; j < s->prim_channels; j++) {
+            av_log(s->avctx, AV_LOG_DEBUG, "Channel 0,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][0]]);
+            av_log(s->avctx, AV_LOG_DEBUG, "Channel 1,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][1]]);
+        }
+        av_log(s->avctx, AV_LOG_DEBUG, "\n");
+    }
+    for (j = 0; j < s->prim_channels; j++)
+        for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
+            av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]);
+    if(s->lfe){
+        av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n");
+        for (j = lfe_samples; j < lfe_samples * 2; j++)
+            av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]);
+        av_log(s->avctx, AV_LOG_DEBUG, "\n");
+    }
+#endif
+
+    return 0;
+}
+
+static void qmf_32_subbands(DCAContext * s, int chans,
+                            float samples_in[32][8], float *samples_out,
+                            float scale, float bias)
+{
+    float *prCoeff;
+    int i, j, k;
+    float praXin[33], *raXin = &praXin[1];
+
+    float *subband_fir_hist = s->subband_fir_hist[chans];
+    float *subband_fir_hist2 = s->subband_fir_noidea[chans];
+
+    int chindex = 0, subindex;
+
+    praXin[0] = 0.0;
+
+    /* Select filter */
+    if (!s->multirate_inter)    /* Non-perfect reconstruction */
+        prCoeff = (float *) fir_32bands_nonperfect;
+    else                        /* Perfect reconstruction */
+        prCoeff = (float *) fir_32bands_perfect;
+
+    /* Reconstructed channel sample index */
+    for (subindex = 0; subindex < 8; subindex++) {
+        float t1, t2, sum[16], diff[16];
+
+        /* Load in one sample from each subband and clear inactive subbands */
+        for (i = 0; i < s->subband_activity[chans]; i++)
+            raXin[i] = samples_in[i][subindex];
+        for (; i < 32; i++)
+            raXin[i] = 0.0;
+
+        /* Multiply by cosine modulation coefficients and
+         * create temporary arrays SUM and DIFF */
+        for (j = 0, k = 0; k < 16; k++) {
+            t1 = 0.0;
+            t2 = 0.0;
+            for (i = 0; i < 16; i++, j++){
+                t1 += (raXin[2 * i] + raXin[2 * i + 1]) * cos_mod[j];
+                t2 += (raXin[2 * i] + raXin[2 * i - 1]) * cos_mod[j + 256];
+            }
+            sum[k] = t1 + t2;
+            diff[k] = t1 - t2;
+        }
+
+        j = 512;
+        /* Store history */
+        for (k = 0; k < 16; k++)
+            subband_fir_hist[k] = cos_mod[j++] * sum[k];
+        for (k = 0; k < 16; k++)
+            subband_fir_hist[32-k-1] = cos_mod[j++] * diff[k];
+
+        /* Multiply by filter coefficients */
+        for (k = 31, i = 0; i < 32; i++, k--)
+            for (j = 0; j < 512; j += 64){
+                subband_fir_hist2[i]    += prCoeff[i+j]  * ( subband_fir_hist[i+j] - subband_fir_hist[j+k]);
+                subband_fir_hist2[i+32] += prCoeff[i+j+32]*(-subband_fir_hist[i+j] - subband_fir_hist[j+k]);
+            }
+
+        /* Create 32 PCM output samples */
+        for (i = 0; i < 32; i++)
+            samples_out[chindex++] = subband_fir_hist2[i] * scale + bias;
+
+        /* Update working arrays */
+        memmove(&subband_fir_hist[32], &subband_fir_hist[0], (512 - 32) * sizeof(float));
+        memmove(&subband_fir_hist2[0], &subband_fir_hist2[32], 32 * sizeof(float));
+        memset(&subband_fir_hist2[32], 0, 32 * sizeof(float));
+    }
+}
+
+static void lfe_interpolation_fir(int decimation_select,
+                                  int num_deci_sample, float *samples_in,
+                                  float *samples_out, float scale,
+                                  float bias)
+{
+    /* samples_in: An array holding decimated samples.
+     *   Samples in current subframe starts from samples_in[0],
+     *   while samples_in[-1], samples_in[-2], ..., stores samples
+     *   from last subframe as history.
+     *
+     * samples_out: An array holding interpolated samples
+     */
+
+    int decifactor, k, j;
+    const float *prCoeff;
+
+    int interp_index = 0;       /* Index to the interpolated samples */
+    int deciindex;
+
+    /* Select decimation filter */
+    if (decimation_select == 1) {
+        decifactor = 128;
+        prCoeff = lfe_fir_128;
+    } else {
+        decifactor = 64;
+        prCoeff = lfe_fir_64;
+    }
+    /* Interpolation */
+    for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
+        /* One decimated sample generates decifactor interpolated ones */
+        for (k = 0; k < decifactor; k++) {
+            float rTmp = 0.0;
+            //FIXME the coeffs are symetric, fix that
+            for (j = 0; j < 512 / decifactor; j++)
+                rTmp += samples_in[deciindex - j] * prCoeff[k + j * decifactor];
+            samples_out[interp_index++] = rTmp / scale + bias;
+        }
+    }
+}
+
+/* downmixing routines */
+#define MIX_REAR1(samples, si1) \
+     samples[i] += samples[si1]; \
+     samples[i+256] += samples[si1];
+
+#define MIX_REAR2(samples, si1, si2) \
+     samples[i] += samples[si1]; \
+     samples[i+256] += samples[si2];
+
+#define MIX_FRONT3(samples) \
+    t = samples[i]; \
+    samples[i] += samples[i+256]; \
+    samples[i+256] = samples[i+512] + t;
+
+#define DOWNMIX_TO_STEREO(op1, op2) \
+    for(i = 0; i < 256; i++){ \
+        op1 \
+        op2 \
+    }
+
+static void dca_downmix(float *samples, int srcfmt)
+{
+    int i;
+    float t;
+
+    switch (srcfmt) {
+    case DCA_MONO:
+    case DCA_CHANNEL:
+    case DCA_STEREO_TOTAL:
+    case DCA_STEREO_SUMDIFF:
+    case DCA_4F2R:
+        av_log(NULL, 0, "Not implemented!\n");
+        break;
+    case DCA_STEREO:
+        break;
+    case DCA_3F:
+        DOWNMIX_TO_STEREO(MIX_FRONT3(samples),);
+        break;
+    case DCA_2F1R:
+        DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + 512),);
+        break;
+    case DCA_3F1R:
+        DOWNMIX_TO_STEREO(MIX_FRONT3(samples),
+                          MIX_REAR1(samples, i + 768));
+        break;
+    case DCA_2F2R:
+        DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + 512, i + 768),);
+        break;
+    case DCA_3F2R:
+        DOWNMIX_TO_STEREO(MIX_FRONT3(samples),
+                          MIX_REAR2(samples, i + 768, i + 1024));
+        break;
+    }
+}
+
+
+/* Very compact version of the block code decoder that does not use table
+ * look-up but is slightly slower */
+static int decode_blockcode(int code, int levels, int *values)
+{
+    int i;
+    int offset = (levels - 1) >> 1;
+
+    for (i = 0; i < 4; i++) {
+        values[i] = (code % levels) - offset;
+        code /= levels;
+    }
+
+    if (code == 0)
+        return 0;
+    else {
+        av_log(NULL, AV_LOG_ERROR, "ERROR: block code look-up failed\n");
+        return -1;
+    }
+}
+
+static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
+static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
+
+static int dca_subsubframe(DCAContext * s)
+{
+    int k, l;
+    int subsubframe = s->current_subsubframe;
+
+    float *quant_step_table;
+
+    /* FIXME */
+    float subband_samples[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
+
+    /*
+     * Audio data
+     */
+
+    /* Select quantization step size table */
+    if (s->bit_rate == 0x1f)
+        quant_step_table = (float *) lossless_quant_d;
+    else
+        quant_step_table = (float *) lossy_quant_d;
+
+    for (k = 0; k < s->prim_channels; k++) {
+        for (l = 0; l < s->vq_start_subband[k]; l++) {
+            int m;
+
+            /* Select the mid-tread linear quantizer */
+            int abits = s->bitalloc[k][l];
+
+            float quant_step_size = quant_step_table[abits];
+            float rscale;
+
+            /*
+             * Determine quantization index code book and its type
+             */
+
+            /* Select quantization index code book */
+            int sel = s->quant_index_huffman[k][abits];
+
+            /*
+             * Extract bits from the bit stream
+             */
+            if(!abits){
+                memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));
+            }else if(abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){
+                if(abits <= 7){
+                    /* Block code */
+                    int block_code1, block_code2, size, levels;
+                    int block[8];
+
+                    size = abits_sizes[abits-1];
+                    levels = abits_levels[abits-1];
+
+                    block_code1 = get_bits(&s->gb, size);
+                    /* FIXME Should test return value */
+                    decode_blockcode(block_code1, levels, block);
+                    block_code2 = get_bits(&s->gb, size);
+                    decode_blockcode(block_code2, levels, &block[4]);
+                    for (m = 0; m < 8; m++)
+                        subband_samples[k][l][m] = block[m];
+                }else{
+                    /* no coding */
+                    for (m = 0; m < 8; m++)
+                        subband_samples[k][l][m] = get_sbits(&s->gb, abits - 3);
+                }
+            }else{
+                /* Huffman coded */
+                for (m = 0; m < 8; m++)
+                    subband_samples[k][l][m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);
+            }
+
+            /* Deal with transients */
+            if (s->transition_mode[k][l] &&
+                subsubframe >= s->transition_mode[k][l])
+                rscale = quant_step_size * s->scale_factor[k][l][1];
+            else
+                rscale = quant_step_size * s->scale_factor[k][l][0];
+
+            rscale *= s->scalefactor_adj[k][sel];
+
+            for (m = 0; m < 8; m++)
+                subband_samples[k][l][m] *= rscale;
+
+            /*
+             * Inverse ADPCM if in prediction mode
+             */
+            if (s->prediction_mode[k][l]) {
+                int n;
+                for (m = 0; m < 8; m++) {
+                    for (n = 1; n <= 4; n++)
+                        if (m >= n)
+                            subband_samples[k][l][m] +=
+                                (adpcm_vb[s->prediction_vq[k][l]][n - 1] *
+                                 subband_samples[k][l][m - n] / 8192);
+                        else if (s->predictor_history)
+                            subband_samples[k][l][m] +=
+                                (adpcm_vb[s->prediction_vq[k][l]][n - 1] *
+                                 s->subband_samples_hist[k][l][m - n +
+                                                               4] / 8192);
+                }
+            }
+        }
+
+        /*
+         * Decode VQ encoded high frequencies
+         */
+        for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {
+            /* 1 vector -> 32 samples but we only need the 8 samples
+             * for this subsubframe. */
+            int m;
+
+            if (!s->debug_flag & 0x01) {
+                av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n");
+                s->debug_flag |= 0x01;
+            }
+
+            for (m = 0; m < 8; m++) {
+                subband_samples[k][l][m] =
+                    high_freq_vq[s->high_freq_vq[k][l]][subsubframe * 8 +
+                                                        m]
+                    * (float) s->scale_factor[k][l][0] / 16.0;
+            }
+        }
+    }
+
+    /* Check for DSYNC after subsubframe */
+    if (s->aspf || subsubframe == s->subsubframes - 1) {
+        if (0xFFFF == get_bits(&s->gb, 16)) {   /* 0xFFFF */
+#ifdef TRACE
+            av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");
+#endif
+        } else {
+            av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
+        }
+    }
+
+    /* Backup predictor history for adpcm */
+    for (k = 0; k < s->prim_channels; k++)
+        for (l = 0; l < s->vq_start_subband[k]; l++)
+            memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4],
+                        4 * sizeof(subband_samples[0][0][0]));
+
+    /* 32 subbands QMF */
+    for (k = 0; k < s->prim_channels; k++) {
+/*        static float pcm_to_double[8] =
+            {32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/
+         qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * k],
+                            2.0 / 3 /*pcm_to_double[s->source_pcm_res] */ ,
+                            0 /*s->bias */ );
+    }
+
+    /* Down mixing */
+
+    if (s->prim_channels > dca_channels[s->output & DCA_CHANNEL_MASK]) {
+        dca_downmix(s->samples, s->amode);
+    }
+
+    /* Generate LFE samples for this subsubframe FIXME!!! */
+    if (s->output & DCA_LFE) {
+        int lfe_samples = 2 * s->lfe * s->subsubframes;
+        int i_channels = dca_channels[s->output & DCA_CHANNEL_MASK];
+
+        lfe_interpolation_fir(s->lfe, 2 * s->lfe,
+                              s->lfe_data + lfe_samples +
+                              2 * s->lfe * subsubframe,
+                              &s->samples[256 * i_channels],
+                              8388608.0, s->bias);
+        /* Outputs 20bits pcm samples */
+    }
+
+    return 0;
+}
+
+
+static int dca_subframe_footer(DCAContext * s)
+{
+    int aux_data_count = 0, i;
+    int lfe_samples;
+
+    /*
+     * Unpack optional information
+     */
+
+    if (s->timestamp)
+        get_bits(&s->gb, 32);
+
+    if (s->aux_data)
+        aux_data_count = get_bits(&s->gb, 6);
+
+    for (i = 0; i < aux_data_count; i++)
+        get_bits(&s->gb, 8);
+
+    if (s->crc_present && (s->downmix || s->dynrange))
+        get_bits(&s->gb, 16);
+
+    lfe_samples = 2 * s->lfe * s->subsubframes;
+    for (i = 0; i < lfe_samples; i++) {
+        s->lfe_data[i] = s->lfe_data[i + lfe_samples];
+    }
+
+    return 0;
+}
+
+/**
+ * Decode a dca frame block
+ *
+ * @param s     pointer to the DCAContext
+ */
+
+static int dca_decode_block(DCAContext * s)
+{
+
+    /* Sanity check */
+    if (s->current_subframe >= s->subframes) {
+        av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
+               s->current_subframe, s->subframes);
+        return -1;
+    }
+
+    if (!s->current_subsubframe) {
+#ifdef TRACE
+        av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
+#endif
+        /* Read subframe header */
+        if (dca_subframe_header(s))
+            return -1;
+    }
+
+    /* Read subsubframe */
+#ifdef TRACE
+    av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");
+#endif
+    if (dca_subsubframe(s))
+        return -1;
+
+    /* Update state */
+    s->current_subsubframe++;
+    if (s->current_subsubframe >= s->subsubframes) {
+        s->current_subsubframe = 0;
+        s->current_subframe++;
+    }
+    if (s->current_subframe >= s->subframes) {
+#ifdef TRACE
+        av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");
+#endif
+        /* Read subframe footer */
+        if (dca_subframe_footer(s))
+            return -1;
+    }
+
+    return 0;
+}
+
+/**
+ * Convert bitstream to one representation based on sync marker
+ */
+static int dca_convert_bitstream(uint8_t * src, int src_size, uint8_t * dst,
+                          int max_size)
+{
+    uint32_t mrk;
+    int i, tmp;
+    uint16_t *ssrc = (uint16_t *) src, *sdst = (uint16_t *) dst;
+    PutBitContext pb;
+
+    mrk = AV_RB32(src);
+    switch (mrk) {
+    case DCA_MARKER_RAW_BE:
+        memcpy(dst, src, FFMIN(src_size, max_size));
+        return FFMIN(src_size, max_size);
+    case DCA_MARKER_RAW_LE:
+        for (i = 0; i < (FFMIN(src_size, max_size) + 1) >> 1; i++)
+            *sdst++ = bswap_16(*ssrc++);
+        return FFMIN(src_size, max_size);
+    case DCA_MARKER_14B_BE:
+    case DCA_MARKER_14B_LE:
+        init_put_bits(&pb, dst, max_size);
+        for (i = 0; i < (src_size + 1) >> 1; i++, src += 2) {
+            tmp = ((mrk == DCA_MARKER_14B_BE) ? AV_RB16(src) : AV_RL16(src)) & 0x3FFF;
+            put_bits(&pb, 14, tmp);
+        }
+        flush_put_bits(&pb);
+        return (put_bits_count(&pb) + 7) >> 3;
+    default:
+        return -1;
+    }
+}
+
+/**
+ * Main frame decoding function
+ * FIXME add arguments
+ */
+static int dca_decode_frame(AVCodecContext * avctx,
+                            void *data, int *data_size,
+                            uint8_t * buf, int buf_size)
+{
+
+    int i, j, k;
+    int16_t *samples = data;
+    DCAContext *s = avctx->priv_data;
+    int channels;
+
+
+    s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer, DCA_MAX_FRAME_SIZE);
+    if (s->dca_buffer_size == -1) {
+        av_log(avctx, AV_LOG_ERROR, "Not a DCA frame\n");
+        return -1;
+    }
+
+    init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
+    if (dca_parse_frame_header(s) < 0) {
+        //seems like the frame is corrupt, try with the next one
+        return buf_size;
+    }
+    //set AVCodec values with parsed data
+    avctx->sample_rate = s->sample_rate;
+    avctx->channels = 2; //FIXME
+    avctx->bit_rate = s->bit_rate;
+
+    channels = dca_channels[s->output];
+    if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
+        return -1;
+    *data_size = 0;
+    for (i = 0; i < (s->sample_blocks / 8); i++) {
+        dca_decode_block(s);
+        s->dsp.float_to_int16(s->tsamples, s->samples, 256 * channels);
+        /* interleave samples */
+        for (j = 0; j < 256; j++) {
+            for (k = 0; k < channels; k++)
+                samples[k] = s->tsamples[j + k * 256];
+            samples += channels;
+        }
+        *data_size += 256 * sizeof(int16_t) * channels;
+    }
+
+    return buf_size;
+}
+
+
+
+/**
+ * Build the cosine modulation tables for the QMF
+ *
+ * @param s     pointer to the DCAContext
+ */
+
+static void pre_calc_cosmod(DCAContext * s)
+{
+    int i, j, k;
+    static int cosmod_inited = 0;
+
+    if(cosmod_inited) return;
+    for (j = 0, k = 0; k < 16; k++)
+        for (i = 0; i < 16; i++)
+            cos_mod[j++] = cos((2 * i + 1) * (2 * k + 1) * M_PI / 64);
+
+    for (k = 0; k < 16; k++)
+        for (i = 0; i < 16; i++)
+            cos_mod[j++] = cos((i) * (2 * k + 1) * M_PI / 32);
+
+    for (k = 0; k < 16; k++)
+        cos_mod[j++] = 0.25 / (2 * cos((2 * k + 1) * M_PI / 128));
+
+    for (k = 0; k < 16; k++)
+        cos_mod[j++] = -0.25 / (2.0 * sin((2 * k + 1) * M_PI / 128));
+
+    cosmod_inited = 1;
+}
+
+
+/**
+ * DCA initialization
+ *
+ * @param avctx     pointer to the AVCodecContext
+ */
+
+static int dca_decode_init(AVCodecContext * avctx)
+{
+    DCAContext *s = avctx->priv_data;
+
+    s->avctx = avctx;
+    dca_init_vlcs();
+    pre_calc_cosmod(s);
+
+    dsputil_init(&s->dsp, avctx);
+    return 0;
+}
+
+
+AVCodec dca_decoder = {
+    .name = "dca",
+    .type = CODEC_TYPE_AUDIO,
+    .id = CODEC_ID_DTS,
+    .priv_data_size = sizeof(DCAContext),
+    .init = dca_decode_init,
+    .decode = dca_decode_frame,
+};
+
+#ifdef CONFIG_DCA_PARSER
+
+typedef struct DCAParseContext {
+    ParseContext pc;
+    uint32_t lastmarker;
+} DCAParseContext;
+
+#define IS_MARKER(state, i, buf, buf_size) \
+ ((state == DCA_MARKER_14B_LE && (i < buf_size-2) && (buf[i+1] & 0xF0) == 0xF0 && buf[i+2] == 0x07) \
+ || (state == DCA_MARKER_14B_BE && (i < buf_size-2) && buf[i+1] == 0x07 && (buf[i+2] & 0xF0) == 0xF0) \
+ || state == DCA_MARKER_RAW_LE || state == DCA_MARKER_RAW_BE)
+
+/**
+ * finds the end of the current frame in the bitstream.
+ * @return the position of the first byte of the next frame, or -1
+ */
+static int dca_find_frame_end(DCAParseContext * pc1, const uint8_t * buf,
+                              int buf_size)
+{
+    int start_found, i;
+    uint32_t state;
+    ParseContext *pc = &pc1->pc;
+
+    start_found = pc->frame_start_found;
+    state = pc->state;
+
+    i = 0;
+    if (!start_found) {
+        for (i = 0; i < buf_size; i++) {
+            state = (state << 8) | buf[i];
+            if (IS_MARKER(state, i, buf, buf_size)) {
+                if (pc1->lastmarker && state == pc1->lastmarker) {
+                    start_found = 1;
+                    break;
+                } else if (!pc1->lastmarker) {
+                    start_found = 1;
+                    pc1->lastmarker = state;
+                    break;
+                }
+            }
+        }
+    }
+    if (start_found) {
+        for (; i < buf_size; i++) {
+            state = (state << 8) | buf[i];
+            if (state == pc1->lastmarker && IS_MARKER(state, i, buf, buf_size)) {
+                pc->frame_start_found = 0;
+                pc->state = -1;
+                return i - 3;
+            }
+        }
+    }
+    pc->frame_start_found = start_found;
+    pc->state = state;
+    return END_NOT_FOUND;
+}
+
+static int dca_parse_init(AVCodecParserContext * s)
+{
+    DCAParseContext *pc1 = s->priv_data;
+
+    pc1->lastmarker = 0;
+    return 0;
+}
+
+static int dca_parse(AVCodecParserContext * s,
+                     AVCodecContext * avctx,
+                     uint8_t ** poutbuf, int *poutbuf_size,
+                     const uint8_t * buf, int buf_size)
+{
+    DCAParseContext *pc1 = s->priv_data;
+    ParseContext *pc = &pc1->pc;
+    int next;
+
+    if (s->flags & PARSER_FLAG_COMPLETE_FRAMES) {
+        next = buf_size;
+    } else {
+        next = dca_find_frame_end(pc1, buf, buf_size);
+
+        if (ff_combine_frame(pc, next, (uint8_t **) & buf, &buf_size) < 0) {
+            *poutbuf = NULL;
+            *poutbuf_size = 0;
+            return buf_size;
+        }
+    }
+    *poutbuf = (uint8_t *) buf;
+    *poutbuf_size = buf_size;
+    return next;
+}
+
+AVCodecParser dca_parser = {
+    {CODEC_ID_DTS},
+    sizeof(DCAParseContext),
+    dca_parse_init,
+    dca_parse,
+    ff_parse_close,
+};
+#endif /* CONFIG_DCA_PARSER */