/*
  * DCA compatible decoder
  * Copyright (C) 2004 Gildas Bazin
  * Copyright (C) 2004 Benjamin Zores
  * Copyright (C) 2006 Benjamin Larsson
  * Copyright (C) 2007 Konstantin Shishkov

  * Copyright (C) 2012 Paul B Mahol
  * Copyright (C) 2014 Niels Möller

  *

  * 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
  */
 
 #include <math.h>
 #include <stddef.h>
 #include <stdio.h>
 

 #include "libavutil/attributes.h"

 #include "libavutil/channel_layout.h"

 #include "libavutil/common.h"

 #include "libavutil/float_dsp.h"

 #include "libavutil/internal.h"

 #include "libavutil/intreadwrite.h"

 #include "libavutil/mathematics.h"

 #include "libavutil/opt.h"

 #include "libavutil/samplefmt.h"

 #include "avcodec.h"

 #include "dca.h"

 #include "dca_syncwords.h"

 #include "dcadata.h"

 #include "dcadsp.h"

 #include "dcahuff.h"
 #include "fft.h"

 #include "fmtconvert.h"

 #include "get_bits.h"

 #include "internal.h"

 #include "mathops.h"
 #include "synth_filter.h"

 #if ARCH_ARM
 #   include "arm/dca.h"
 #endif
 

 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
 };
 

 enum DCAXxchSpeakerMask {
     DCA_XXCH_FRONT_CENTER          = 0x0000001,
     DCA_XXCH_FRONT_LEFT            = 0x0000002,
     DCA_XXCH_FRONT_RIGHT           = 0x0000004,
     DCA_XXCH_SIDE_REAR_LEFT        = 0x0000008,
     DCA_XXCH_SIDE_REAR_RIGHT       = 0x0000010,
     DCA_XXCH_LFE1                  = 0x0000020,
     DCA_XXCH_REAR_CENTER           = 0x0000040,
     DCA_XXCH_SURROUND_REAR_LEFT    = 0x0000080,
     DCA_XXCH_SURROUND_REAR_RIGHT   = 0x0000100,
     DCA_XXCH_SIDE_SURROUND_LEFT    = 0x0000200,
     DCA_XXCH_SIDE_SURROUND_RIGHT   = 0x0000400,
     DCA_XXCH_FRONT_CENTER_LEFT     = 0x0000800,
     DCA_XXCH_FRONT_CENTER_RIGHT    = 0x0001000,
     DCA_XXCH_FRONT_HIGH_LEFT       = 0x0002000,
     DCA_XXCH_FRONT_HIGH_CENTER     = 0x0004000,
     DCA_XXCH_FRONT_HIGH_RIGHT      = 0x0008000,
     DCA_XXCH_LFE2                  = 0x0010000,
     DCA_XXCH_SIDE_FRONT_LEFT       = 0x0020000,
     DCA_XXCH_SIDE_FRONT_RIGHT      = 0x0040000,
     DCA_XXCH_OVERHEAD              = 0x0080000,
     DCA_XXCH_SIDE_HIGH_LEFT        = 0x0100000,
     DCA_XXCH_SIDE_HIGH_RIGHT       = 0x0200000,
     DCA_XXCH_REAR_HIGH_CENTER      = 0x0400000,
     DCA_XXCH_REAR_HIGH_LEFT        = 0x0800000,
     DCA_XXCH_REAR_HIGH_RIGHT       = 0x1000000,
     DCA_XXCH_REAR_LOW_CENTER       = 0x2000000,
     DCA_XXCH_REAR_LOW_LEFT         = 0x4000000,
     DCA_XXCH_REAR_LOW_RIGHT        = 0x8000000,
 };
 

 #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_NSYNCAUX        0x9A1105A0
 

 /** Bit allocation */

 typedef struct BitAlloc {

     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;

 }
 

 static float dca_dmix_code(unsigned code);
 

 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[ff_dca_vlc_offs[i]];
         dca_bitalloc_index.vlc[i].table_allocated = ff_dca_vlc_offs[i + 1] - ff_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[ff_dca_vlc_offs[i + 5]];
         dca_scalefactor.vlc[i].table_allocated = ff_dca_vlc_offs[i + 6] - ff_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[ff_dca_vlc_offs[i + 10]];
         dca_tmode.vlc[i].table_allocated = ff_dca_vlc_offs[i + 11] - ff_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[ff_dca_vlc_offs[c]];
             dca_smpl_bitalloc[i + 1].vlc[j].table_allocated = ff_dca_vlc_offs[c + 1] - ff_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 inline int dca_xxch2index(DCAContext *s, int xxch_ch)
 {
     int i, base, mask;
 
     /* locate channel set containing the channel */
     for (i = -1, base = 0, mask = (s->xxch_core_spkmask & ~DCA_XXCH_LFE1);
          i <= s->xxch_chset && !(mask & xxch_ch); mask = s->xxch_spk_masks[++i])
         base += av_popcount(mask);
 
     return base + av_popcount(mask & (xxch_ch - 1));
 }
 
 static int dca_parse_audio_coding_header(DCAContext *s, int base_channel,
                                          int xxch)

 {
     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 };

     int hdr_pos = 0, hdr_size = 0;

     float scale_factor;

     int this_chans, acc_mask;
     int embedded_downmix;
     int nchans, mask[8];
     int coeff, ichan;
 
     /* xxch has arbitrary sized audio coding headers */
     if (xxch) {
         hdr_pos  = get_bits_count(&s->gb);
         hdr_size = get_bits(&s->gb, 7) + 1;
     }

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

     if (xxch && nchans >= 3) {
         av_log(s->avctx, AV_LOG_ERROR, "nchans %d is too large\n", nchans);
         return AVERROR_INVALIDDATA;
     } else if (nchans + base_channel > DCA_PRIM_CHANNELS_MAX) {
         av_log(s->avctx, AV_LOG_ERROR, "channel sum %d + %d is too large\n", nchans, base_channel);
         return AVERROR_INVALIDDATA;
     }
 

     s->total_channels = nchans + base_channel;

     s->prim_channels  = s->total_channels;

     /* obtain speaker layout mask & downmix coefficients for XXCH */
     if (xxch) {
         acc_mask = s->xxch_core_spkmask;
 
         this_chans = get_bits(&s->gb, s->xxch_nbits_spk_mask - 6) << 6;
         s->xxch_spk_masks[s->xxch_chset] = this_chans;
         s->xxch_chset_nch[s->xxch_chset] = nchans;
 
         for (i = 0; i <= s->xxch_chset; i++)
             acc_mask |= s->xxch_spk_masks[i];
 
         /* check for downmixing information */
         if (get_bits1(&s->gb)) {
             embedded_downmix = get_bits1(&s->gb);

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

 
             if (coeff<1 || coeff>61) {
                 av_log(s->avctx, AV_LOG_ERROR, "6bit coeff %d is out of range\n", coeff);
                 return AVERROR_INVALIDDATA;
             }
 

             scale_factor     = -1.0f / dca_dmix_code((coeff<<2)-3);

 
             s->xxch_dmix_sf[s->xxch_chset] = scale_factor;
 
             for (i = base_channel; i < s->prim_channels; i++) {
                 mask[i] = get_bits(&s->gb, s->xxch_nbits_spk_mask);
             }
 
             for (j = base_channel; j < s->prim_channels; j++) {
                 memset(s->xxch_dmix_coeff[j], 0, sizeof(s->xxch_dmix_coeff[0]));
                 s->xxch_dmix_embedded |= (embedded_downmix << j);
                 for (i = 0; i < s->xxch_nbits_spk_mask; i++) {
                     if (mask[j] & (1 << i)) {
                         if ((1 << i) == DCA_XXCH_LFE1) {
                             av_log(s->avctx, AV_LOG_WARNING,
                                    "DCA-XXCH: dmix to LFE1 not supported.\n");
                             continue;
                         }
 
                         coeff = get_bits(&s->gb, 7);
                         ichan = dca_xxch2index(s, 1 << i);

                         if ((coeff&63)<1 || (coeff&63)>61) {
                             av_log(s->avctx, AV_LOG_ERROR, "7bit coeff %d is out of range\n", coeff);
                             return AVERROR_INVALIDDATA;
                         }

                         s->xxch_dmix_coeff[j][ichan] = dca_dmix_code((coeff<<2)-3);

                     }
                 }
             }
         }
     }
 

     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 (!xxch) {
         if (s->crc_present) {
             /* Audio header CRC check */
             get_bits(&s->gb, 16);
         }
     } else {
         /* Skip to the end of the header, also ignore CRC if present  */
         i = get_bits_count(&s->gb);
         if (hdr_pos + 8 * hdr_size > i)
             skip_bits_long(&s->gb, hdr_pos + 8 * hdr_size - i);

     }
 

     s->current_subframe    = 0;

     s->current_subsubframe = 0;
 

     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       = avpriv_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          = ff_dca_bit_rates[s->bit_rate_index];

     if (!s->bit_rate)

         return AVERROR_INVALIDDATA;

     skip_bits1(&s->gb); // always 0 (reserved, cf. ETSI TS 102 114 V1.4.1)

     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);
 

     if (s->lfe > 2) {

         s->lfe = 0;

         av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE value: %d\n", s->lfe);
         return AVERROR_INVALIDDATA;

     }
 

     /* 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;

 
     /* Primary audio coding header */

     s->subframes = get_bits(&s->gb, 4) + 1;

     return dca_parse_audio_coding_header(s, 0, 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;

         if (block_index + s->subsubframes[s->current_subframe] > s->sample_blocks/8) {
             s->subsubframes[s->current_subframe] = 1;
             return AVERROR_INVALIDDATA;
         }

         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) {

                 ff_dlog(s->avctx, "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 = ff_dca_scale_factor_quant7;

             log_size    = 7;

         } else {

             scale_table = ff_dca_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;
             }
         }
     }
 
     /* 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) {

         int quant7;

         /* 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 */

         quant7 = get_bits(&s->gb, 8);
         if (quant7 > 127) {

             avpriv_request_sample(s->avctx, "LFEScaleIndex larger than 127");

             return AVERROR_INVALIDDATA;
         }

         s->lfe_scale_factor = ff_dca_scale_factor_quant7[quant7];

 
         /* 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;
     }
 
     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 sb_act = s->subband_activity[chans];

     scale *= sqrt(1 / 8.0);

 
     /* Select filter */
     if (!s->multirate_inter)    /* Non-perfect reconstruction */

         prCoeff = ff_dca_fir_32bands_nonperfect;

     else                        /* Perfect reconstruction */

         prCoeff = ff_dca_fir_32bands_perfect;

     s->dcadsp.qmf_32_subbands(samples_in, sb_act, &s->synth, &s->imdct,
                               s->subband_fir_hist[chans],
                               &s->hist_index[chans],
                               s->subband_fir_noidea[chans], prCoeff,
                               samples_out, s->raXin, scale);

 }
 

 static QMF64_table *qmf64_precompute(void)
 {
     unsigned i, j;
     QMF64_table *table = av_malloc(sizeof(*table));
     if (!table)
         return NULL;
 
     for (i = 0; i < 32; i++)
         for (j = 0; j < 32; j++)
             table->dct4_coeff[i][j] = cos((2 * i + 1) * (2 * j + 1) * M_PI / 128);
     for (i = 0; i < 32; i++)
         for (j = 0; j < 32; j++)
             table->dct2_coeff[i][j] = cos((2 * i + 1) *      j      * M_PI /  64);
 
     /* FIXME: Is the factor 0.125 = 1/8 right? */
     for (i = 0; i < 32; i++)
         table->rcos[i] =  0.125 / cos((2 * i + 1) * M_PI / 256);
     for (i = 0; i < 32; i++)
         table->rsin[i] = -0.125 / sin((2 * i + 1) * M_PI / 256);
 
     return table;
 }
 
 /* FIXME: Totally unoptimized. Based on the reference code and
  * http://multimedia.cx/mirror/dca-transform.pdf, with guessed tweaks
  * for doubling the size. */
 static void qmf_64_subbands(DCAContext *s, int chans, float samples_in[64][8],
                             float *samples_out, float scale)
 {
     float raXin[64];
     float A[32], B[32];
     float *raX = s->subband_fir_hist[chans];
     float *raZ = s->subband_fir_noidea[chans];
     unsigned i, j, k, subindex;
 
     for (i = s->subband_activity[chans]; i < 64; i++)
         raXin[i] = 0.0;
     for (subindex = 0; subindex < 8; subindex++) {
         for (i = 0; i < s->subband_activity[chans]; i++)
             raXin[i] = samples_in[i][subindex];
 
         for (k = 0; k < 32; k++) {
             A[k] = 0.0;
             for (i = 0; i < 32; i++)
                 A[k] += (raXin[2 * i] + raXin[2 * i + 1]) * s->qmf64_table->dct4_coeff[k][i];
         }
         for (k = 0; k < 32; k++) {
             B[k] = raXin[0] * s->qmf64_table->dct2_coeff[k][0];
             for (i = 1; i < 32; i++)
                 B[k] += (raXin[2 * i] + raXin[2 * i - 1]) * s->qmf64_table->dct2_coeff[k][i];
         }
         for (k = 0; k < 32; k++) {
             raX[k]      = s->qmf64_table->rcos[k] * (A[k] + B[k]);
             raX[63 - k] = s->qmf64_table->rsin[k] * (A[k] - B[k]);
         }
 
         for (i = 0; i < 64; i++) {
             float out = raZ[i];
             for (j = 0; j < 1024; j += 128)
                 out += ff_dca_fir_64bands[j + i] * (raX[j + i] - raX[j + 63 - i]);
             *samples_out++ = out * scale;
         }
 
         for (i = 0; i < 64; i++) {
             float hist = 0.0;
             for (j = 0; j < 1024; j += 128)
                 hist += ff_dca_fir_64bands[64 + j + i] * (-raX[i + j] - raX[j + 63 - i]);
 
             raZ[i] = hist;
         }
 
         /* FIXME: Make buffer circular, to avoid this move. */
         memmove(raX + 64, raX, (1024 - 64) * sizeof(*raX));
     }
 }
 
 static void lfe_interpolation_fir(DCAContext *s, const float *samples_in,

                                   float *samples_out)

 {
     /* 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 idx;

     const float *prCoeff;
     int deciindex;
 
     /* Select decimation filter */

     if (s->lfe == 1) {

         idx     = 1;

         prCoeff = ff_dca_lfe_fir_128;

     } else {

         idx = 0;
         if (s->exss_ext_mask & DCA_EXT_EXSS_XLL)
             prCoeff = ff_dca_lfe_xll_fir_64;
         else
             prCoeff = ff_dca_lfe_fir_64;

     }
     /* Interpolation */

     for (deciindex = 0; deciindex < 2 * s->lfe; deciindex++) {

         s->dcadsp.lfe_fir[idx](samples_out, samples_in, prCoeff);

         samples_in++;

         samples_out += 2 * 32 * (1 + idx);

     }
 }
 
 /* downmixing routines */

 #define MIX_REAR1(samples, s1, rs, coef)            \
     samples[0][i] += samples[s1][i] * coef[rs][0];  \
     samples[1][i] += samples[s1][i] * coef[rs][1];

 #define MIX_REAR2(samples, s1, s2, rs, coef)                                          \
     samples[0][i] += samples[s1][i] * coef[rs][0] + samples[s2][i] * coef[rs + 1][0]; \
     samples[1][i] += samples[s1][i] * coef[rs][1] + samples[s2][i] * coef[rs + 1][1];

 
 #define MIX_FRONT3(samples, coef)                                      \

     t = samples[c][i];                                                 \
     u = samples[l][i];                                                 \
     v = samples[r][i];                                                 \
     samples[0][i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0];  \
     samples[1][i] = 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 lfe_present,
                         float coef[DCA_PRIM_CHANNELS_MAX + 1][2],

                         const int8_t *channel_mapping)

 {

     int c, l, r, sl, sr, s;

     int i;

     float t, u, v;

 
     switch (srcfmt) {
     case DCA_MONO:
     case DCA_4F2R:

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

         break;

     case DCA_CHANNEL:

     case DCA_STEREO:

     case DCA_STEREO_TOTAL:
     case DCA_STEREO_SUMDIFF:

         break;
     case DCA_3F:

         c = channel_mapping[0];
         l = channel_mapping[1];
         r = channel_mapping[2];

         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );

         break;
     case DCA_2F1R:

         s = channel_mapping[2];
         DOWNMIX_TO_STEREO(MIX_REAR1(samples, s, 2, coef), );

         break;
     case DCA_3F1R:

         c = channel_mapping[0];
         l = channel_mapping[1];
         r = channel_mapping[2];
         s = channel_mapping[3];

         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

                           MIX_REAR1(samples, s, 3, coef));

         break;
     case DCA_2F2R:

         sl = channel_mapping[2];
         sr = channel_mapping[3];
         DOWNMIX_TO_STEREO(MIX_REAR2(samples, sl, sr, 2, coef), );

         break;
     case DCA_3F2R:

         c  = channel_mapping[0];
         l  = channel_mapping[1];
         r  = channel_mapping[2];
         sl = channel_mapping[3];
         sr = channel_mapping[4];

         DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),

                           MIX_REAR2(samples, sl, sr, 3, coef));

         break;
     }

     if (lfe_present) {

         int lf_buf = ff_dca_lfe_index[srcfmt];
         int lf_idx =  ff_dca_channels[srcfmt];

         for (i = 0; i < 256; i++) {
             samples[0][i] += samples[lf_buf][i] * coef[lf_idx][0];
             samples[1][i] += samples[lf_buf][i] * coef[lf_idx][1];
         }
     }

 }
 

 #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, int32_t *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, int32_t *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 };

 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(int32_t, block, [8 * DCA_SUBBANDS]);

 
     /*
      * Audio data
      */
 
     /* Select quantization step size table */

     if (s->bit_rate_index == 0x1f)

         quant_step_table = ff_dca_lossless_quant_d;

     else

         quant_step_table = ff_dca_lossy_quant_d;

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

         float rscale[DCA_SUBBANDS];
 

         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) {

                 rscale[l] = 0;
                 memset(block + 8 * l, 0, 8 * sizeof(block[0]));

             } else {
                 /* Deal with transients */
                 int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];

                 rscale[l] = 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 + 8 * l);

                         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[8 * l + m] = get_sbits(&s->gb, abits - 3);

                     }

                 } else {

                     /* Huffman coded */

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

                         block[8 * l + m] = get_bitalloc(&s->gb,

                                                         &dca_smpl_bitalloc[abits], sel);

                 }

             }

         }
 
         s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[k][0],
                                                block, rscale, 8 * s->vq_start_subband[k]);

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

             /*
              * Inverse ADPCM if in prediction mode
              */
             if (s->prediction_mode[k][l]) {
                 int n;

                 if (s->predictor_history)

                     subband_samples[k][l][0] += (ff_dca_adpcm_vb[s->prediction_vq[k][l]][0] *

                                                  s->subband_samples_hist[k][l][3] +

                                                  ff_dca_adpcm_vb[s->prediction_vq[k][l]][1] *

                                                  s->subband_samples_hist[k][l][2] +

                                                  ff_dca_adpcm_vb[s->prediction_vq[k][l]][2] *

                                                  s->subband_samples_hist[k][l][1] +

                                                  ff_dca_adpcm_vb[s->prediction_vq[k][l]][3] *

                                                  s->subband_samples_hist[k][l][0]) *
                                                 (1.0f / 8192);
                 for (m = 1; m < 8; m++) {

                     float sum = ff_dca_adpcm_vb[s->prediction_vq[k][l]][0] *

                                 subband_samples[k][l][m - 1];
                     for (n = 2; n <= 4; n++)

                         if (m >= n)

                             sum += ff_dca_adpcm_vb[s->prediction_vq[k][l]][n - 1] *

                                    subband_samples[k][l][m - n];

                         else if (s->predictor_history)

                             sum += ff_dca_adpcm_vb[s->prediction_vq[k][l]][n - 1] *

                                    s->subband_samples_hist[k][l][m - n + 4];

                     subband_samples[k][l][m] += sum * (1.0f / 8192);

                 }
             }
         }
 
         /*
          * Decode VQ encoded high frequencies
          */

         if (s->subband_activity[k] > s->vq_start_subband[k]) {

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

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

                 s->debug_flag |= 0x01;
             }

             s->dcadsp.decode_hf(subband_samples[k], s->high_freq_vq[k],

                                 ff_dca_high_freq_vq, subsubframe * 8,

                                 s->scale_factor[k], s->vq_start_subband[k],
                                 s->subband_activity[k]);

         }
     }
 
     /* Check for DSYNC after subsubframe */

     if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {

         if (get_bits(&s->gb, 16) != 0xFFFF) {

             av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");

             return AVERROR_INVALIDDATA;

         }
     }
 
     /* Backup predictor history for adpcm */

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

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

             AV_COPY128(s->subband_samples_hist[k][l], &subband_samples[k][l][4]);

     return 0;
 }
 

 static int dca_filter_channels(DCAContext *s, int block_index, int upsample)

 {
     float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
     int k;
 

     if (upsample) {
         if (!s->qmf64_table) {
             s->qmf64_table = qmf64_precompute();
             if (!s->qmf64_table)
                 return AVERROR(ENOMEM);
         }
 
         /* 64 subbands QMF */
         for (k = 0; k < s->prim_channels; k++) {
             if (s->channel_order_tab[k] >= 0)
                 qmf_64_subbands(s, k, subband_samples[k],
                                 s->samples_chanptr[s->channel_order_tab[k]],
                                 /* Upsampling needs a factor 2 here. */
                                 M_SQRT2 / 32768.0);
         }
     } else {
         /* 32 subbands QMF */
         for (k = 0; k < s->prim_channels; k++) {
             if (s->channel_order_tab[k] >= 0)
                 qmf_32_subbands(s, k, subband_samples[k],
                                 s->samples_chanptr[s->channel_order_tab[k]],
                                 M_SQRT1_2 / 32768.0);
         }

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

     if (s->lfe) {

         float *samples = s->samples_chanptr[s->lfe_index];

         lfe_interpolation_fir(s,

                               s->lfe_data + 2 * s->lfe * (block_index + 4),

                               samples);
         if (upsample) {
             unsigned i;
             /* Should apply the filter in Table 6-11 when upsampling. For
              * now, just duplicate. */

             for (i = 255; i > 0; i--) {

                 samples[2 * i]     =
                 samples[2 * i + 1] = samples[i];
             }
             samples[1] = samples[0];
         }

     }
 

     /* FIXME: This downmixing is probably broken with upsample.
      * Probably totally broken also with XLL in general. */

     /* Downmixing to Stereo */
     if (s->prim_channels + !!s->lfe > 2 &&
         s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
         dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef,
                     s->channel_order_tab);
     }
 

     return 0;
 }
 

 static int dca_subframe_footer(DCAContext *s, int base_channel)

 {

     int in, out, aux_data_count, aux_data_end, reserved;

     uint32_t nsyncaux;

 
     /*
      * 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);

             // align (32-bit)
             skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
 
             aux_data_end = 8 * aux_data_count + get_bits_count(&s->gb);
 

             if ((nsyncaux = get_bits_long(&s->gb, 32)) != DCA_NSYNCAUX) {
                 av_log(s->avctx, AV_LOG_ERROR, "nSYNCAUX mismatch %#"PRIx32"\n",
                        nsyncaux);

                 return AVERROR_INVALIDDATA;

             }

 
             if (get_bits1(&s->gb)) { // bAUXTimeStampFlag
                 avpriv_request_sample(s->avctx,
                                       "Auxiliary Decode Time Stamp Flag");
                 // align (4-bit)
                 skip_bits(&s->gb, (-get_bits_count(&s->gb)) & 4);
                 // 44 bits: nMSByte (8), nMarker (4), nLSByte (28), nMarker (4)
                 skip_bits_long(&s->gb, 44);
             }
 
             if ((s->core_downmix = get_bits1(&s->gb))) {

                 int am = get_bits(&s->gb, 3);
                 switch (am) {

                 case 0:
                     s->core_downmix_amode = DCA_MONO;
                     break;
                 case 1:
                     s->core_downmix_amode = DCA_STEREO;
                     break;
                 case 2:
                     s->core_downmix_amode = DCA_STEREO_TOTAL;
                     break;
                 case 3:
                     s->core_downmix_amode = DCA_3F;
                     break;
                 case 4:
                     s->core_downmix_amode = DCA_2F1R;
                     break;
                 case 5:
                     s->core_downmix_amode = DCA_2F2R;
                     break;
                 case 6:
                     s->core_downmix_amode = DCA_3F1R;
                     break;
                 default:

                     av_log(s->avctx, AV_LOG_ERROR,
                            "Invalid mode %d for embedded downmix coefficients\n",
                            am);

                     return AVERROR_INVALIDDATA;
                 }

                 for (out = 0; out < ff_dca_channels[s->core_downmix_amode]; out++) {

                     for (in = 0; in < s->prim_channels + !!s->lfe; in++) {
                         uint16_t tmp = get_bits(&s->gb, 9);

                         if ((tmp & 0xFF) > 241) {
                             av_log(s->avctx, AV_LOG_ERROR,
                                    "Invalid downmix coefficient code %"PRIu16"\n",
                                    tmp);

                             return AVERROR_INVALIDDATA;

                         }

                         s->core_downmix_codes[in][out] = tmp;
                     }
                 }
             }
 
             align_get_bits(&s->gb); // byte align
             skip_bits(&s->gb, 16);  // nAUXCRC16
 
             // additional data (reserved, cf. ETSI TS 102 114 V1.4.1)

             if ((reserved = (aux_data_end - get_bits_count(&s->gb))) < 0) {

                 av_log(s->avctx, AV_LOG_ERROR,
                        "Overread auxiliary data by %d bits\n", -reserved);

                 return AVERROR_INVALIDDATA;

             } else if (reserved) {

                 avpriv_request_sample(s->avctx,
                                       "Core auxiliary data reserved content");
                 skip_bits_long(&s->gb, reserved);
             }
         }

         if (s->crc_present && 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) {
         /* Read subframe header */

         if ((ret = dca_subframe_header(s, base_channel, block_index)))
             return ret;

     }
 
     /* Read subsubframe */

     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) {
         /* Read subframe footer */

         if ((ret = dca_subframe_footer(s, base_channel)))
             return ret;

     }
 
     return 0;
 }
 

 int ff_dca_xbr_parse_frame(DCAContext *s)

 {
     int scale_table_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS][2];
     int active_bands[DCA_CHSETS_MAX][DCA_CHSET_CHANS_MAX];
     int abits_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS];
     int anctemp[DCA_CHSET_CHANS_MAX];
     int chset_fsize[DCA_CHSETS_MAX];
     int n_xbr_ch[DCA_CHSETS_MAX];
     int hdr_size, num_chsets, xbr_tmode, hdr_pos;
     int i, j, k, l, chset, chan_base;
 
     av_log(s->avctx, AV_LOG_DEBUG, "DTS-XBR: decoding XBR extension\n");
 
     /* get bit position of sync header */
     hdr_pos = get_bits_count(&s->gb) - 32;
 
     hdr_size = get_bits(&s->gb, 6) + 1;
     num_chsets = get_bits(&s->gb, 2) + 1;
 
     for(i = 0; i < num_chsets; i++)
         chset_fsize[i] = get_bits(&s->gb, 14) + 1;
 
     xbr_tmode = get_bits1(&s->gb);
 
     for(i = 0; i < num_chsets; i++) {
         n_xbr_ch[i] = get_bits(&s->gb, 3) + 1;
         k = get_bits(&s->gb, 2) + 5;

         for(j = 0; j < n_xbr_ch[i]; j++) {

             active_bands[i][j] = get_bits(&s->gb, k) + 1;

             if (active_bands[i][j] > DCA_SUBBANDS) {
                 av_log(s->avctx, AV_LOG_ERROR, "too many active subbands (%d)\n", active_bands[i][j]);
                 return AVERROR_INVALIDDATA;
             }
         }

     }
 
     /* skip to the end of the header */
     i = get_bits_count(&s->gb);
     if(hdr_pos + hdr_size * 8 > i)
         skip_bits_long(&s->gb, hdr_pos + hdr_size * 8 - i);
 
     /* loop over the channel data sets */
     /* only decode as many channels as we've decoded base data for */
     for(chset = 0, chan_base = 0;
         chset < num_chsets && chan_base + n_xbr_ch[chset] <= s->prim_channels;
         chan_base += n_xbr_ch[chset++]) {
         int start_posn = get_bits_count(&s->gb);
         int subsubframe = 0;
         int subframe = 0;
 
         /* loop over subframes */
         for (k = 0; k < (s->sample_blocks / 8); k++) {
             /* parse header if we're on first subsubframe of a block */
             if(subsubframe == 0) {
                 /* Parse subframe header */
                 for(i = 0; i < n_xbr_ch[chset]; i++) {
                     anctemp[i] = get_bits(&s->gb, 2) + 2;
                 }
 
                 for(i = 0; i < n_xbr_ch[chset]; i++) {
                     get_array(&s->gb, abits_high[i], active_bands[chset][i], anctemp[i]);
                 }
 
                 for(i = 0; i < n_xbr_ch[chset]; i++) {
                     anctemp[i] = get_bits(&s->gb, 3);
                     if(anctemp[i] < 1) {
                         av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: SYNC ERROR\n");
                         return AVERROR_INVALIDDATA;
                     }
                 }
 
                 /* generate scale factors */
                 for(i = 0; i < n_xbr_ch[chset]; i++) {
                     const uint32_t *scale_table;
                     int nbits;

                     int scale_table_size;

 
                     if (s->scalefactor_huffman[chan_base+i] == 6) {

                         scale_table = ff_dca_scale_factor_quant7;

                         scale_table_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7);

                     } else {

                         scale_table = ff_dca_scale_factor_quant6;

                         scale_table_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant6);

                     }
 
                     nbits = anctemp[i];
 
                     for(j = 0; j < active_bands[chset][i]; j++) {
                         if(abits_high[i][j] > 0) {

                             int index = get_bits(&s->gb, nbits);
                             if (index >= scale_table_size) {
                                 av_log(s->avctx, AV_LOG_ERROR, "scale table index %d invalid\n", index);
                                 return AVERROR_INVALIDDATA;
                             }
                             scale_table_high[i][j][0] = scale_table[index];

 
                             if(xbr_tmode && s->transition_mode[i][j]) {

                                 int index = get_bits(&s->gb, nbits);
                                 if (index >= scale_table_size) {
                                     av_log(s->avctx, AV_LOG_ERROR, "scale table index %d invalid\n", index);
                                     return AVERROR_INVALIDDATA;
                                 }
                                 scale_table_high[i][j][1] = scale_table[index];

                             }
                         }
                     }
                 }
             }
 
             /* decode audio array for this block */
             for(i = 0; i < n_xbr_ch[chset]; i++) {
                 for(j = 0; j < active_bands[chset][i]; j++) {
                     const int xbr_abits = abits_high[i][j];

                     const float quant_step_size = ff_dca_lossless_quant_d[xbr_abits];

                     const int sfi = xbr_tmode && s->transition_mode[i][j] && subsubframe >= s->transition_mode[i][j];
                     const float rscale = quant_step_size * scale_table_high[i][j][sfi];
                     float *subband_samples = s->subband_samples[k][chan_base+i][j];
                     int block[8];
 
                     if(xbr_abits <= 0)
                         continue;
 
                     if(xbr_abits > 7) {
                         get_array(&s->gb, block, 8, xbr_abits - 3);
                     } else {
                         int block_code1, block_code2, size, levels, err;
 
                         size   = abits_sizes[xbr_abits - 1];
                         levels = abits_levels[xbr_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: DTS-XBR: block code look-up failed\n");
                             return AVERROR_INVALIDDATA;
                         }
                     }
 
                     /* scale & sum into subband */
                     for(l = 0; l < 8; l++)
                         subband_samples[l] += (float)block[l] * rscale;
                 }
             }
 
             /* check DSYNC marker */
             if(s->aspf || subsubframe == s->subsubframes[subframe] - 1) {
                 if(get_bits(&s->gb, 16) != 0xffff) {
                     av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: Didn't get subframe DSYNC\n");
                     return AVERROR_INVALIDDATA;
                 }
             }
 
             /* advance sub-sub-frame index */
             if(++subsubframe >= s->subsubframes[subframe]) {
                 subsubframe = 0;
                 subframe++;
             }
         }
 
         /* skip to next channel set */
         i = get_bits_count(&s->gb);
         if(start_posn + chset_fsize[chset] * 8 != i) {
             j = start_posn + chset_fsize[chset] * 8 - i;
             if(j < 0 || j >= 8)
                 av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: end of channel set,"
                        " skipping further than expected (%d bits)\n", j);
             skip_bits_long(&s->gb, j);
         }
     }
 
     return 0;
 }
 

 /* parse initial header for XXCH and dump details */

 int ff_dca_xxch_decode_frame(DCAContext *s)

 {

     int hdr_size, spkmsk_bits, num_chsets, core_spk, hdr_pos;

     int i, chset, base_channel, chstart, fsize[8];
 
     /* assume header word has already been parsed */
     hdr_pos     = get_bits_count(&s->gb) - 32;
     hdr_size    = get_bits(&s->gb, 6) + 1;

   /*chhdr_crc   =*/ skip_bits1(&s->gb);

     spkmsk_bits = get_bits(&s->gb, 5) + 1;
     num_chsets  = get_bits(&s->gb, 2) + 1;
 
     for (i = 0; i < num_chsets; i++)
         fsize[i] = get_bits(&s->gb, 14) + 1;
 
     core_spk               = get_bits(&s->gb, spkmsk_bits);
     s->xxch_core_spkmask   = core_spk;
     s->xxch_nbits_spk_mask = spkmsk_bits;
     s->xxch_dmix_embedded  = 0;
 
     /* skip to the end of the header */
     i = get_bits_count(&s->gb);
     if (hdr_pos + hdr_size * 8 > i)
         skip_bits_long(&s->gb, hdr_pos + hdr_size * 8 - i);
 
     for (chset = 0; chset < num_chsets; chset++) {
         chstart       = get_bits_count(&s->gb);
         base_channel  = s->prim_channels;
         s->xxch_chset = chset;
 
         /* XXCH and Core headers differ, see 6.4.2 "XXCH Channel Set Header" vs.
            5.3.2 "Primary Audio Coding Header", DTS Spec 1.3.1 */
         dca_parse_audio_coding_header(s, base_channel, 1);
 
         /* decode channel data */
         for (i = 0; i < (s->sample_blocks / 8); i++) {
             if (dca_decode_block(s, base_channel, i)) {
                 av_log(s->avctx, AV_LOG_ERROR,
                        "Error decoding DTS-XXCH extension\n");
                 continue;
             }
         }
 
         /* skip to end of this section */
         i = get_bits_count(&s->gb);
         if (chstart + fsize[chset] * 8 > i)
             skip_bits_long(&s->gb, chstart + fsize[chset] * 8 - i);
     }
     s->xxch_chset = num_chsets;
 
     return 0;
 }
 

 static float dca_dmix_code(unsigned code)
 {
     int sign = (code >> 8) - 1;
     code &= 0xff;

     return ((ff_dca_dmixtable[code] ^ sign) - sign) * (1.0 / (1 << 15));

 }
 

 /**

  * Main frame decoding function
  * FIXME add arguments
  */

 static int dca_decode_frame(AVCodecContext *avctx, void *data,
                             int *got_frame_ptr, AVPacket *avpkt)

 {

     AVFrame *frame     = data;

     const uint8_t *buf = avpkt->data;

     int buf_size       = avpkt->size;

     int channel_mask;
     int channel_layout;

     int lfe_samples;

     int num_core_channels = 0;

     int i, ret;

     float **samples_flt;

     float *src_chan;
     float *dst_chan;

     DCAContext *s = avctx->priv_data;

     int core_ss_end;

     int channels, full_channels;

     float scale;
     int achan;
     int chset;
     int mask;
     int lavc;
     int posn;
     int j, k;
     int endch;

     int upsample = 0;

     s->exss_ext_mask = 0;
     s->xch_present   = 0;

     s->dca_buffer_size = AVERROR_INVALIDDATA;
     for (i = 0; i < buf_size - 3 && s->dca_buffer_size == AVERROR_INVALIDDATA; i++)
         s->dca_buffer_size = avpriv_dca_convert_bitstream(buf + i, buf_size - i, 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;

     }
 

     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;
 

     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->prim_channels + !!s->lfe > 2 &&

         avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {

             /* Stereo downmix coefficients
              *
              * The decoder can only downmix to 2-channel, so we need to ensure
              * embedded downmix coefficients are actually targeting 2-channel.
              */
             if (s->core_downmix && (s->core_downmix_amode == DCA_STEREO ||
                                     s->core_downmix_amode == DCA_STEREO_TOTAL)) {

                 for (i = 0; i < num_core_channels + !!s->lfe; i++) {

                     /* Range checked earlier */
                     s->downmix_coef[i][0] = dca_dmix_code(s->core_downmix_codes[i][0]);
                     s->downmix_coef[i][1] = dca_dmix_code(s->core_downmix_codes[i][1]);

                 }

                 s->output = s->core_downmix_amode;

             } else {
                 int am = s->amode & DCA_CHANNEL_MASK;

                 if (am >= FF_ARRAY_ELEMS(ff_dca_default_coeffs)) {

                     av_log(s->avctx, AV_LOG_ERROR,
                            "Invalid channel mode %d\n", am);
                     return AVERROR_INVALIDDATA;
                 }

                 if (num_core_channels + !!s->lfe >

                     FF_ARRAY_ELEMS(ff_dca_default_coeffs[0])) {

                     avpriv_request_sample(s->avctx, "Downmixing %d channels",
                                           s->prim_channels + !!s->lfe);
                     return AVERROR_PATCHWELCOME;
                 }

                 for (i = 0; i < num_core_channels + !!s->lfe; i++) {

                     s->downmix_coef[i][0] = ff_dca_default_coeffs[am][i][0];
                     s->downmix_coef[i][1] = ff_dca_default_coeffs[am][i][1];

                 }

             }

             ff_dlog(s->avctx, "Stereo downmix coeffs:\n");

             for (i = 0; i < num_core_channels + !!s->lfe; i++) {

                 ff_dlog(s->avctx, "L, input channel %d = %f\n", i,

                         s->downmix_coef[i][0]);

                 ff_dlog(s->avctx, "R, input channel %d = %f\n", i,

                         s->downmix_coef[i][1]);

             }

             ff_dlog(s->avctx, "\n");

     }
 

     if (s->ext_coding)

         s->core_ext_mask = ff_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 | DCA_EXT_XXCH)) {

         /* 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 DCA_SYNCWORD_XCH: {

                 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;
                 }

                 if (s->xch_base_channel < 2) {

                     avpriv_request_sample(avctx, "XCh with fewer than 2 base channels");

                     continue;
                 }
 

                 /* much like core primary audio coding header */

                 dca_parse_audio_coding_header(s, s->xch_base_channel, 0);

 
                 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 DCA_SYNCWORD_XXCH:

                 /* 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;

                 ff_dca_xxch_decode_frame(s);

                 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_SYNCWORD_SUBSTREAM)

         ff_dca_exss_parse_header(s);

 
     avctx->profile = s->profile;
 

     full_channels = channels = s->prim_channels + !!s->lfe;

     /* If we have XXCH then the channel layout is managed differently */
     /* note that XLL will also have another way to do things */

 #if FF_API_REQUEST_CHANNELS
 FF_DISABLE_DEPRECATION_WARNINGS

     if (!(s->core_ext_mask & DCA_EXT_XXCH)
         || (s->core_ext_mask & DCA_EXT_XXCH && avctx->request_channels > 0
             && avctx->request_channels
             < num_core_channels + !!s->lfe + s->xxch_chset_nch[0]))

     {
 FF_ENABLE_DEPRECATION_WARNINGS
 #else
     if (!(s->core_ext_mask & DCA_EXT_XXCH)) {
 #endif
         /* xxx should also do MA extensions */

         if (s->amode < 16) {

             avctx->channel_layout = ff_dca_core_channel_layout[s->amode];

 
             if (s->prim_channels + !!s->lfe > 2 &&
                 avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
                 /*
                  * Neither the core's auxiliary data nor our default tables contain
                  * downmix coefficients for the additional channel coded in the XCh
                  * extension, so when we're doing a Stereo downmix, don't decode it.
                  */
                 s->xch_disable = 1;
             }

 #if FF_API_REQUEST_CHANNELS
 FF_DISABLE_DEPRECATION_WARNINGS

             if (s->xch_present && !s->xch_disable &&
                 (!avctx->request_channels ||
                  avctx->request_channels > num_core_channels + !!s->lfe)) {

 FF_ENABLE_DEPRECATION_WARNINGS
 #else

             if (s->xch_present && !s->xch_disable) {

 #endif

                 if (avctx->channel_layout & AV_CH_BACK_CENTER) {
                     avpriv_request_sample(avctx, "XCh with Back center channel");
                     return AVERROR_INVALIDDATA;
                 }

                 avctx->channel_layout |= AV_CH_BACK_CENTER;
                 if (s->lfe) {
                     avctx->channel_layout |= AV_CH_LOW_FREQUENCY;

                     s->channel_order_tab = ff_dca_channel_reorder_lfe_xch[s->amode];

                 } else {

                     s->channel_order_tab = ff_dca_channel_reorder_nolfe_xch[s->amode];

                 }

                 if (s->channel_order_tab[s->xch_base_channel] < 0)
                     return AVERROR_INVALIDDATA;

             } 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 = ff_dca_channel_reorder_lfe[s->amode];

                 } else

                     s->channel_order_tab = ff_dca_channel_reorder_nolfe[s->amode];

             }

 
             if (channels > !!s->lfe &&
                 s->channel_order_tab[channels - 1 - !!s->lfe] < 0)
                 return AVERROR_INVALIDDATA;
 

             if (av_get_channel_layout_nb_channels(avctx->channel_layout) != channels) {
                 av_log(avctx, AV_LOG_ERROR, "Number of channels %d mismatches layout %d\n", channels, av_get_channel_layout_nb_channels(avctx->channel_layout));
                 return AVERROR_INVALIDDATA;
             }
 

             if (num_core_channels + !!s->lfe > 2 &&

                 avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {

                 channels              = 2;
                 s->output             = s->prim_channels == 2 ? s->amode : DCA_STEREO;

                 avctx->channel_layout = AV_CH_LAYOUT_STEREO;
             }
             else if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) {
                 static const int8_t dca_channel_order_native[9] = { 0, 1, 2, 3, 4, 5, 6, 7, 8 };
                 s->channel_order_tab = dca_channel_order_native;
             }

             s->lfe_index = ff_dca_lfe_index[s->amode];

         } else {

             av_log(avctx, AV_LOG_ERROR,
                    "Non standard configuration %d !\n", s->amode);
             return AVERROR_INVALIDDATA;
         }
 

         s->xxch_dmix_embedded = 0;

     } else {
         /* we only get here if an XXCH channel set can be added to the mix */
         channel_mask = s->xxch_core_spkmask;
 

 #if FF_API_REQUEST_CHANNELS
 FF_DISABLE_DEPRECATION_WARNINGS

         if (avctx->request_channels > 0
             && avctx->request_channels < s->prim_channels) {

             channels = num_core_channels + !!s->lfe;

             for (i = 0; i < s->xxch_chset && channels + s->xxch_chset_nch[i]
                                               <= avctx->request_channels; i++) {
                 channels += s->xxch_chset_nch[i];
                 channel_mask |= s->xxch_spk_masks[i];
             }

 FF_ENABLE_DEPRECATION_WARNINGS
         } else
 #endif
         {

             channels = s->prim_channels + !!s->lfe;
             for (i = 0; i < s->xxch_chset; i++) {
                 channel_mask |= s->xxch_spk_masks[i];
             }

         }

         /* Given the DTS spec'ed channel mask, generate an avcodec version */
         channel_layout = 0;
         for (i = 0; i < s->xxch_nbits_spk_mask; ++i) {
             if (channel_mask & (1 << i)) {

                 channel_layout |= ff_dca_map_xxch_to_native[i];

             }
         }

         /* make sure that we have managed to get equivalent dts/avcodec channel

          * masks in some sense -- unfortunately some channels could overlap */
         if (av_popcount(channel_mask) != av_popcount(channel_layout)) {
             av_log(avctx, AV_LOG_DEBUG,

                    "DTS-XXCH: Inconsistent avcodec/dts channel layouts\n");

             return AVERROR_INVALIDDATA;

         }
 

         avctx->channel_layout = channel_layout;
 
         if (!(avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE)) {
             /* Estimate DTS --> avcodec ordering table */
             for (chset = -1, j = 0; chset < s->xxch_chset; ++chset) {
                 mask = chset >= 0 ? s->xxch_spk_masks[chset]
                                   : s->xxch_core_spkmask;
                 for (i = 0; i < s->xxch_nbits_spk_mask; i++) {
                     if (mask & ~(DCA_XXCH_LFE1 | DCA_XXCH_LFE2) & (1 << i)) {

                         lavc = ff_dca_map_xxch_to_native[i];

                         posn = av_popcount(channel_layout & (lavc - 1));
                         s->xxch_order_tab[j++] = posn;
                     }
                 }

             }

             s->lfe_index = av_popcount(channel_layout & (AV_CH_LOW_FREQUENCY-1));
         } else { /* native ordering */
             for (i = 0; i < channels; i++)
                 s->xxch_order_tab[i] = i;
 
             s->lfe_index = channels - 1;
         }
 
         s->channel_order_tab = s->xxch_order_tab;

     }

     /* get output buffer */

     frame->nb_samples = 256 * (s->sample_blocks / 8);

     if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) {
         int xll_nb_samples = s->xll_segments * s->xll_smpl_in_seg;
         /* Check for invalid/unsupported conditions first */
         if (s->xll_residual_channels > channels) {
             av_log(s->avctx, AV_LOG_WARNING,
                    "DCA: too many residual channels (%d, core channels %d). Disabling XLL\n",
                    s->xll_residual_channels, channels);
             s->exss_ext_mask &= ~DCA_EXT_EXSS_XLL;
         } else if (xll_nb_samples != frame->nb_samples &&
                    2 * frame->nb_samples != xll_nb_samples) {
             av_log(s->avctx, AV_LOG_WARNING,
                    "DCA: unsupported upsampling (%d XLL samples, %d core samples). Disabling XLL\n",
                    xll_nb_samples, frame->nb_samples);
             s->exss_ext_mask &= ~DCA_EXT_EXSS_XLL;
         } else {
             if (2 * frame->nb_samples == xll_nb_samples) {
                 av_log(s->avctx, AV_LOG_INFO,
                        "XLL: upsampling core channels by a factor of 2\n");
                 upsample = 1;
 
                 frame->nb_samples = xll_nb_samples;
                 // FIXME: Is it good enough to copy from the first channel set?
                 avctx->sample_rate = s->xll_chsets[0].sampling_frequency;
             }
             /* If downmixing to stereo, don't decode additional channels.
              * FIXME: Using the xch_disable flag for this doesn't seem right. */
             if (!s->xch_disable)

                 channels = s->xll_channels;

         }
     }
 

     if (avctx->channels != channels) {

         if (avctx->channels)
             av_log(avctx, AV_LOG_INFO, "Number of channels changed in DCA decoder (%d -> %d)\n", avctx->channels, channels);
         avctx->channels = channels;

     }

     /* FIXME: This is an ugly hack, to just revert to the default
      * layout if we have additional channels. Need to convert the XLL

      * channel masks to ffmpeg channel_layout mask. */

     if (av_get_channel_layout_nb_channels(avctx->channel_layout) != avctx->channels)
         avctx->channel_layout = 0;
 

     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)

         return ret;

     samples_flt = (float **) frame->extended_data;

     /* allocate buffer for extra channels if downmixing */
     if (avctx->channels < full_channels) {
         ret = av_samples_get_buffer_size(NULL, full_channels - channels,

                                          frame->nb_samples,

                                          avctx->sample_fmt, 0);
         if (ret < 0)
             return ret;
 
         av_fast_malloc(&s->extra_channels_buffer,
                        &s->extra_channels_buffer_size, ret);
         if (!s->extra_channels_buffer)
             return AVERROR(ENOMEM);
 

         ret = av_samples_fill_arrays((uint8_t **) s->extra_channels, NULL,

                                      s->extra_channels_buffer,
                                      full_channels - channels,

                                      frame->nb_samples, avctx->sample_fmt, 0);

         if (ret < 0)
             return ret;
     }
 

     /* filter to get final output */

     for (i = 0; i < (s->sample_blocks / 8); i++) {

         int ch;

         unsigned block = upsample ? 512 : 256;

         for (ch = 0; ch < channels; ch++)

             s->samples_chanptr[ch] = samples_flt[ch] + i * block;

         for (; ch < full_channels; ch++)

             s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * block;

         dca_filter_channels(s, i, upsample);

 
         /* 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_chanptr[s->channel_order_tab[s->xch_base_channel]];
             float *lt_chan   = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 2]];
             float *rt_chan   = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 1]];

             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 stream contains XXCH, we might need to undo an embedded downmix */
         if (s->xxch_dmix_embedded) {
             /* Loop over channel sets in turn */
             ch = num_core_channels;
             for (chset = 0; chset < s->xxch_chset; chset++) {
                 endch = ch + s->xxch_chset_nch[chset];
                 mask = s->xxch_dmix_embedded;
 
                 /* undo downmix */
                 for (j = ch; j < endch; j++) {
                     if (mask & (1 << j)) { /* this channel has been mixed-out */

                         src_chan = s->samples_chanptr[s->channel_order_tab[j]];

                         for (k = 0; k < endch; k++) {
                             achan = s->channel_order_tab[k];
                             scale = s->xxch_dmix_coeff[j][k];
                             if (scale != 0.0) {

                                 dst_chan = s->samples_chanptr[achan];

                                 s->fdsp->vector_fmac_scalar(dst_chan, src_chan,

                                                            -scale, 256);
                             }
                         }
                     }
                 }
 
                 /* if a downmix has been embedded then undo the pre-scaling */
                 if ((mask & (1 << ch)) && s->xxch_dmix_sf[chset] != 1.0f) {
                     scale = s->xxch_dmix_sf[chset];
 
                     for (j = 0; j < ch; j++) {

                         src_chan = s->samples_chanptr[s->channel_order_tab[j]];

                         for (k = 0; k < 256; k++)
                             src_chan[k] *= scale;
                     }
 
                     /* LFE channel is always part of core, scale if it exists */
                     if (s->lfe) {

                         src_chan = s->samples_chanptr[s->lfe_index];

                         for (k = 0; k < 256; k++)
                             src_chan[k] *= scale;
                     }
                 }
 
                 ch = endch;
             }
 
         }

     }
 

     /* 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];
 

     if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) {
         ret = ff_dca_xll_decode_audio(s, frame);
         if (ret < 0)
             return ret;
     }

     /* AVMatrixEncoding
      *
      * DCA_STEREO_TOTAL (Lt/Rt) is equivalent to Dolby Surround */
     ret = ff_side_data_update_matrix_encoding(frame,
                                               (s->output & ~DCA_LFE) == DCA_STEREO_TOTAL ?
                                               AV_MATRIX_ENCODING_DOLBY : AV_MATRIX_ENCODING_NONE);
     if (ret < 0)
         return ret;
 

     if (   avctx->profile != FF_PROFILE_DTS_HD_MA
         && avctx->profile != FF_PROFILE_DTS_HD_HRA)
         avctx->bit_rate = s->bit_rate;

     *got_frame_ptr = 1;

     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;
 
     s->avctx = avctx;
     dca_init_vlcs();
 

     s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);

     if (!s->fdsp)
         return AVERROR(ENOMEM);
 

     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);

     avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

     /* allow downmixing to stereo */

 #if FF_API_REQUEST_CHANNELS
 FF_DISABLE_DEPRECATION_WARNINGS
     if (avctx->request_channels == 2)
         avctx->request_channel_layout = AV_CH_LAYOUT_STEREO;
 FF_ENABLE_DEPRECATION_WARNINGS
 #endif
     if (avctx->channels > 2 &&
         avctx->request_channel_layout == AV_CH_LAYOUT_STEREO)
         avctx->channels = 2;

     return 0;
 }
 

 static av_cold int dca_decode_end(AVCodecContext *avctx)

 {
     DCAContext *s = avctx->priv_data;
     ff_mdct_end(&s->imdct);

     av_freep(&s->extra_channels_buffer);

     av_freep(&s->fdsp);

     av_freep(&s->xll_sample_buf);
     av_freep(&s->qmf64_table);

     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 },
 };
 

 static const AVOption options[] = {

     { "disable_xch", "disable decoding of the XCh extension", offsetof(DCAContext, xch_disable), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },

     { "disable_xll", "disable decoding of the XLL extension", offsetof(DCAContext, xll_disable), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },

     { NULL },
 };
 
 static const AVClass dca_decoder_class = {
     .class_name = "DCA decoder",
     .item_name  = av_default_item_name,
     .option     = options,
     .version    = LIBAVUTIL_VERSION_INT,

     .category   = AV_CLASS_CATEGORY_DECODER,

 };
 

 AVCodec ff_dca_decoder = {

     .name            = "dca",

     .long_name       = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),

     .type            = AVMEDIA_TYPE_AUDIO,

     .id              = AV_CODEC_ID_DTS,

     .priv_data_size  = sizeof(DCAContext),
     .init            = dca_decode_init,
     .decode          = dca_decode_frame,
     .close           = dca_decode_end,

     .capabilities    = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,

     .sample_fmts     = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,

                                                        AV_SAMPLE_FMT_NONE },
     .profiles        = NULL_IF_CONFIG_SMALL(profiles),

     .priv_class      = &dca_decoder_class,

 };