/*
  * MLP decoder
  * Copyright (c) 2007-2008 Ian Caulfield
  *
  * 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

  * MLP decoder
  */
 

 #include <stdint.h>
 

 #include "avcodec.h"
 #include "libavutil/intreadwrite.h"

 #include "libavutil/channel_layout.h"

 #include "get_bits.h"

 #include "internal.h"

 #include "libavutil/crc.h"
 #include "parser.h"
 #include "mlp_parser.h"

 #include "mlpdsp.h"

 #include "mlp.h"

 /** number of bits used for VLC lookup - longest Huffman code is 9 */

 #define VLC_BITS            9
 
 typedef struct SubStream {

     /// Set if a valid restart header has been read. Otherwise the substream cannot be decoded.

     uint8_t     restart_seen;
 
     //@{
     /** restart header data */

     /// The type of noise to be used in the rematrix stage.

     uint16_t    noise_type;
 

     /// The index of the first channel coded in this substream.

     uint8_t     min_channel;

     /// The index of the last channel coded in this substream.

     uint8_t     max_channel;

     /// The number of channels input into the rematrix stage.

     uint8_t     max_matrix_channel;

     /// For each channel output by the matrix, the output channel to map it to

     uint8_t     ch_assign[MAX_CHANNELS];

     /// The channel layout for this substream
     uint64_t    ch_layout;

     /// Channel coding parameters for channels in the substream

     ChannelParams channel_params[MAX_CHANNELS];
 

     /// The left shift applied to random noise in 0x31ea substreams.

     uint8_t     noise_shift;

     /// The current seed value for the pseudorandom noise generator(s).

     uint32_t    noisegen_seed;
 

     /// Set if the substream contains extra info to check the size of VLC blocks.

     uint8_t     data_check_present;
 

     /// Bitmask of which parameter sets are conveyed in a decoding parameter block.

     uint8_t     param_presence_flags;
 #define PARAM_BLOCKSIZE     (1 << 7)
 #define PARAM_MATRIX        (1 << 6)
 #define PARAM_OUTSHIFT      (1 << 5)
 #define PARAM_QUANTSTEP     (1 << 4)
 #define PARAM_FIR           (1 << 3)
 #define PARAM_IIR           (1 << 2)
 #define PARAM_HUFFOFFSET    (1 << 1)

 #define PARAM_PRESENCE      (1 << 0)

     //@}
 
     //@{
     /** matrix data */
 

     /// Number of matrices to be applied.

     uint8_t     num_primitive_matrices;
 

     /// matrix output channel

     uint8_t     matrix_out_ch[MAX_MATRICES];
 

     /// Whether the LSBs of the matrix output are encoded in the bitstream.

     uint8_t     lsb_bypass[MAX_MATRICES];

     /// Matrix coefficients, stored as 2.14 fixed point.

     int32_t     matrix_coeff[MAX_MATRICES][MAX_CHANNELS];

     /// Left shift to apply to noise values in 0x31eb substreams.

     uint8_t     matrix_noise_shift[MAX_MATRICES];
     //@}
 

     /// Left shift to apply to Huffman-decoded residuals.

     uint8_t     quant_step_size[MAX_CHANNELS];
 

     /// number of PCM samples in current audio block

     uint16_t    blocksize;

     /// Number of PCM samples decoded so far in this frame.

     uint16_t    blockpos;
 

     /// Left shift to apply to decoded PCM values to get final 24-bit output.

     int8_t      output_shift[MAX_CHANNELS];
 

     /// Running XOR of all output samples.

     int32_t     lossless_check_data;
 
 } SubStream;
 
 typedef struct MLPDecodeContext {
     AVCodecContext *avctx;
 

     /// Current access unit being read has a major sync.

     int         is_major_sync_unit;
 

     /// Set if a valid major sync block has been read. Otherwise no decoding is possible.

     uint8_t     params_valid;
 

     /// Number of substreams contained within this stream.

     uint8_t     num_substreams;
 

     /// Index of the last substream to decode - further substreams are skipped.

     uint8_t     max_decoded_substream;
 

     /// Stream needs channel reordering to comply with FFmpeg's channel order

     uint8_t     needs_reordering;
 

     /// number of PCM samples contained in each frame

     int         access_unit_size;

     /// next power of two above the number of samples in each frame

     int         access_unit_size_pow2;
 
     SubStream   substream[MAX_SUBSTREAMS];
 

     int         matrix_changed;
     int         filter_changed[MAX_CHANNELS][NUM_FILTERS];
 

     int8_t      noise_buffer[MAX_BLOCKSIZE_POW2];
     int8_t      bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];

     int32_t     sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS];

     MLPDSPContext dsp;

 } MLPDecodeContext;
 

 static const uint64_t thd_channel_order[] = {
     AV_CH_FRONT_LEFT, AV_CH_FRONT_RIGHT,                     // LR
     AV_CH_FRONT_CENTER,                                      // C
     AV_CH_LOW_FREQUENCY,                                     // LFE
     AV_CH_SIDE_LEFT, AV_CH_SIDE_RIGHT,                       // LRs
     AV_CH_TOP_FRONT_LEFT, AV_CH_TOP_FRONT_RIGHT,             // LRvh
     AV_CH_FRONT_LEFT_OF_CENTER, AV_CH_FRONT_RIGHT_OF_CENTER, // LRc
     AV_CH_BACK_LEFT, AV_CH_BACK_RIGHT,                       // LRrs
     AV_CH_BACK_CENTER,                                       // Cs
     AV_CH_TOP_CENTER,                                        // Ts
     AV_CH_SURROUND_DIRECT_LEFT, AV_CH_SURROUND_DIRECT_RIGHT, // LRsd
     AV_CH_WIDE_LEFT, AV_CH_WIDE_RIGHT,                       // LRw
     AV_CH_TOP_FRONT_CENTER,                                  // Cvh
     AV_CH_LOW_FREQUENCY_2,                                   // LFE2
 };
 
 static uint64_t thd_channel_layout_extract_channel(uint64_t channel_layout,
                                                    int index)
 {
     int i;
 
     if (av_get_channel_layout_nb_channels(channel_layout) <= index)
         return 0;
 
     for (i = 0; i < FF_ARRAY_ELEMS(thd_channel_order); i++)
         if (channel_layout & thd_channel_order[i] && !index--)
             return thd_channel_order[i];
     return 0;
 }
 

 static VLC huff_vlc[3];
 
 /** Initialize static data, constant between all invocations of the codec. */
 

 static av_cold void init_static(void)

 {

     if (!huff_vlc[0].bits) {

         INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
                     &ff_mlp_huffman_tables[0][0][1], 2, 1,
                     &ff_mlp_huffman_tables[0][0][0], 2, 1, 512);
         INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
                     &ff_mlp_huffman_tables[1][0][1], 2, 1,
                     &ff_mlp_huffman_tables[1][0][0], 2, 1, 512);
         INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
                     &ff_mlp_huffman_tables[2][0][1], 2, 1,
                     &ff_mlp_huffman_tables[2][0][0], 2, 1, 512);

     }

     ff_mlp_init_crc();

 }
 
 static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
                                           unsigned int substr, unsigned int ch)
 {
     SubStream *s = &m->substream[substr];

     ChannelParams *cp = &s->channel_params[ch];

     int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
     int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
     int32_t sign_huff_offset = cp->huff_offset;

     if (cp->codebook > 0)

         sign_huff_offset -= 7 << lsb_bits;
 
     if (sign_shift >= 0)
         sign_huff_offset -= 1 << sign_shift;
 
     return sign_huff_offset;
 }
 
 /** Read a sample, consisting of either, both or neither of entropy-coded MSBs
  *  and plain LSBs. */
 
 static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
                                      unsigned int substr, unsigned int pos)
 {
     SubStream *s = &m->substream[substr];
     unsigned int mat, channel;
 
     for (mat = 0; mat < s->num_primitive_matrices; mat++)
         if (s->lsb_bypass[mat])
             m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
 
     for (channel = s->min_channel; channel <= s->max_channel; channel++) {

         ChannelParams *cp = &s->channel_params[channel];

         int codebook = cp->codebook;

         int quant_step_size = s->quant_step_size[channel];

         int lsb_bits = cp->huff_lsbs - quant_step_size;

         int result = 0;
 
         if (codebook > 0)
             result = get_vlc2(gbp, huff_vlc[codebook-1].table,
                             VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
 
         if (result < 0)

             return AVERROR_INVALIDDATA;

 
         if (lsb_bits > 0)
             result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
 

         result  += cp->sign_huff_offset;

         result <<= quant_step_size;
 
         m->sample_buffer[pos + s->blockpos][channel] = result;
     }
 
     return 0;
 }
 
 static av_cold int mlp_decode_init(AVCodecContext *avctx)
 {
     MLPDecodeContext *m = avctx->priv_data;
     int substr;
 
     init_static();
     m->avctx = avctx;
     for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
         m->substream[substr].lossless_check_data = 0xffffffff;

     ff_mlpdsp_init(&m->dsp);

     return 0;
 }
 
 /** Read a major sync info header - contains high level information about
  *  the stream - sample rate, channel arrangement etc. Most of this
  *  information is not actually necessary for decoding, only for playback.
  */
 
 static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
 {
     MLPHeaderInfo mh;

     int substr, ret;

     if ((ret = ff_mlp_read_major_sync(m->avctx, &mh, gb)) != 0)
         return ret;

 
     if (mh.group1_bits == 0) {

         av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");

         return AVERROR_INVALIDDATA;

     }
     if (mh.group2_bits > mh.group1_bits) {
         av_log(m->avctx, AV_LOG_ERROR,

                "Channel group 2 cannot have more bits per sample than group 1.\n");

         return AVERROR_INVALIDDATA;

     }
 
     if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
         av_log(m->avctx, AV_LOG_ERROR,

                "Channel groups with differing sample rates are not currently supported.\n");

         return AVERROR_INVALIDDATA;

     }
 
     if (mh.group1_samplerate == 0) {

         av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");

         return AVERROR_INVALIDDATA;

     }
     if (mh.group1_samplerate > MAX_SAMPLERATE) {
         av_log(m->avctx, AV_LOG_ERROR,

                "Sampling rate %d is greater than the supported maximum (%d).\n",

                mh.group1_samplerate, MAX_SAMPLERATE);

         return AVERROR_INVALIDDATA;

     }
     if (mh.access_unit_size > MAX_BLOCKSIZE) {
         av_log(m->avctx, AV_LOG_ERROR,

                "Block size %d is greater than the supported maximum (%d).\n",

                mh.access_unit_size, MAX_BLOCKSIZE);

         return AVERROR_INVALIDDATA;

     }
     if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
         av_log(m->avctx, AV_LOG_ERROR,

                "Block size pow2 %d is greater than the supported maximum (%d).\n",

                mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);

         return AVERROR_INVALIDDATA;

     }
 
     if (mh.num_substreams == 0)

         return AVERROR_INVALIDDATA;

     if (m->avctx->codec_id == AV_CODEC_ID_MLP && mh.num_substreams > 2) {

         av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n");

         return AVERROR_INVALIDDATA;

     }

     if (mh.num_substreams > MAX_SUBSTREAMS) {

         av_log_ask_for_sample(m->avctx,

                "Number of substreams %d is larger than the maximum supported "

                "by the decoder.\n", mh.num_substreams);
         return AVERROR_PATCHWELCOME;

     }
 
     m->access_unit_size      = mh.access_unit_size;
     m->access_unit_size_pow2 = mh.access_unit_size_pow2;
 
     m->num_substreams        = mh.num_substreams;
     m->max_decoded_substream = m->num_substreams - 1;
 
     m->avctx->sample_rate    = mh.group1_samplerate;
     m->avctx->frame_size     = mh.access_unit_size;
 

     m->avctx->bits_per_raw_sample = mh.group1_bits;

     if (mh.group1_bits > 16)

         m->avctx->sample_fmt = AV_SAMPLE_FMT_S32;

     else

         m->avctx->sample_fmt = AV_SAMPLE_FMT_S16;

 
     m->params_valid = 1;
     for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
         m->substream[substr].restart_seen = 0;
 

     /* Set the layout for each substream. When there's more than one, the first
      * substream is Stereo. Subsequent substreams' layouts are indicated in the
      * major sync. */
     if (m->avctx->codec_id == AV_CODEC_ID_MLP) {
         if ((substr = (mh.num_substreams > 1)))
             m->substream[0].ch_layout = AV_CH_LAYOUT_STEREO;
         m->substream[substr].ch_layout = mh.channel_layout_mlp;
     } else {
         if ((substr = (mh.num_substreams > 1)))
             m->substream[0].ch_layout = AV_CH_LAYOUT_STEREO;
         if (mh.num_substreams > 2)
             if (mh.channel_layout_thd_stream2)
                 m->substream[2].ch_layout = mh.channel_layout_thd_stream2;
             else
                 m->substream[2].ch_layout = mh.channel_layout_thd_stream1;
         m->substream[substr].ch_layout = mh.channel_layout_thd_stream1;

 
         if (m->avctx->channels<=2 && m->substream[substr].ch_layout == AV_CH_LAYOUT_MONO && m->max_decoded_substream == 1) {
             av_log(m->avctx, AV_LOG_DEBUG, "Mono stream with 2 substreams, ignoring 2nd\n");
             m->max_decoded_substream = 0;
             if (m->avctx->channels==2)
                 m->avctx->channel_layout = AV_CH_LAYOUT_STEREO;
         }

     }
 

     m->needs_reordering = mh.channel_arrangement >= 18 && mh.channel_arrangement <= 20;

     return 0;
 }
 
 /** Read a restart header from a block in a substream. This contains parameters
  *  required to decode the audio that do not change very often. Generally
  *  (always) present only in blocks following a major sync. */
 
 static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
                                const uint8_t *buf, unsigned int substr)
 {
     SubStream *s = &m->substream[substr];
     unsigned int ch;
     int sync_word, tmp;
     uint8_t checksum;
     uint8_t lossless_check;
     int start_count = get_bits_count(gbp);

     const int max_matrix_channel = m->avctx->codec_id == AV_CODEC_ID_MLP

                                  ? MAX_MATRIX_CHANNEL_MLP
                                  : MAX_MATRIX_CHANNEL_TRUEHD;

     int max_channel, min_channel, matrix_channel;

 
     sync_word = get_bits(gbp, 13);
 

     if (sync_word != 0x31ea >> 1) {

         av_log(m->avctx, AV_LOG_ERROR,

                "restart header sync incorrect (got 0x%04x)\n", sync_word);

         return AVERROR_INVALIDDATA;

     }
 

     s->noise_type = get_bits1(gbp);
 

     if (m->avctx->codec_id == AV_CODEC_ID_MLP && s->noise_type) {

         av_log(m->avctx, AV_LOG_ERROR, "MLP must have 0x31ea sync word.\n");

         return AVERROR_INVALIDDATA;

     }
 

     skip_bits(gbp, 16); /* Output timestamp */
 

     min_channel    = get_bits(gbp, 4);
     max_channel    = get_bits(gbp, 4);
     matrix_channel = get_bits(gbp, 4);

     if (matrix_channel > max_matrix_channel) {

         av_log(m->avctx, AV_LOG_ERROR,
                "Max matrix channel cannot be greater than %d.\n",
                max_matrix_channel);

         return AVERROR_INVALIDDATA;

     }
 

     if (max_channel != matrix_channel) {

         av_log(m->avctx, AV_LOG_ERROR,
                "Max channel must be equal max matrix channel.\n");

         return AVERROR_INVALIDDATA;

     }
 

     /* This should happen for TrueHD streams with >6 channels and MLP's noise
      * type. It is not yet known if this is allowed. */

     if (max_channel > MAX_MATRIX_CHANNEL_MLP && !s->noise_type) {

         av_log_ask_for_sample(m->avctx,

                "Number of channels %d is larger than the maximum supported "

                "by the decoder.\n", max_channel + 2);

         return AVERROR_PATCHWELCOME;

     }
 

     if (min_channel > max_channel) {

         av_log(m->avctx, AV_LOG_ERROR,
                "Substream min channel cannot be greater than max channel.\n");

         return AVERROR_INVALIDDATA;

     }
 

     s->min_channel = min_channel;
     s->max_channel = max_channel;
     s->max_matrix_channel = matrix_channel;
 

 #if FF_API_REQUEST_CHANNELS
     if (m->avctx->request_channels > 0 &&
         m->avctx->request_channels <= s->max_channel + 1 &&
         m->max_decoded_substream > substr) {

         av_log(m->avctx, AV_LOG_DEBUG,

                "Extracting %d-channel downmix from substream %d. "

                "Further substreams will be skipped.\n",
                s->max_channel + 1, substr);
         m->max_decoded_substream = substr;

     } else
 #endif
     if (m->avctx->request_channel_layout == s->ch_layout &&
         m->max_decoded_substream > substr) {
         av_log(m->avctx, AV_LOG_DEBUG,
                "Extracting %d-channel downmix (0x%"PRIx64") from substream %d. "
                "Further substreams will be skipped.\n",
                s->max_channel + 1, s->ch_layout, substr);
         m->max_decoded_substream = substr;

     }
 
     s->noise_shift   = get_bits(gbp,  4);
     s->noisegen_seed = get_bits(gbp, 23);
 
     skip_bits(gbp, 19);
 
     s->data_check_present = get_bits1(gbp);
     lossless_check = get_bits(gbp, 8);
     if (substr == m->max_decoded_substream
         && s->lossless_check_data != 0xffffffff) {

         tmp = xor_32_to_8(s->lossless_check_data);

         if (tmp != lossless_check)
             av_log(m->avctx, AV_LOG_WARNING,

                    "Lossless check failed - expected %02x, calculated %02x.\n",

                    lossless_check, tmp);
     }
 
     skip_bits(gbp, 16);
 

     memset(s->ch_assign, 0, sizeof(s->ch_assign));
 

     for (ch = 0; ch <= s->max_matrix_channel; ch++) {
         int ch_assign = get_bits(gbp, 6);

         if (m->avctx->codec_id == AV_CODEC_ID_TRUEHD) {
             uint64_t channel = thd_channel_layout_extract_channel(s->ch_layout,
                                                                   ch_assign);
             ch_assign = av_get_channel_layout_channel_index(s->ch_layout,
                                                             channel);
         }

         if ((unsigned)ch_assign > s->max_matrix_channel) {

             av_log_ask_for_sample(m->avctx,
                    "Assignment of matrix channel %d to invalid output channel %d.\n",
                    ch, ch_assign);
             return AVERROR_PATCHWELCOME;

         }

         s->ch_assign[ch_assign] = ch;

     }
 

     checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);

 
     if (checksum != get_bits(gbp, 8))

         av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");

     /* Set default decoding parameters. */

     s->param_presence_flags   = 0xff;
     s->num_primitive_matrices = 0;
     s->blocksize              = 8;
     s->lossless_check_data    = 0;
 
     memset(s->output_shift   , 0, sizeof(s->output_shift   ));
     memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
 
     for (ch = s->min_channel; ch <= s->max_channel; ch++) {

         ChannelParams *cp = &s->channel_params[ch];

         cp->filter_params[FIR].order = 0;
         cp->filter_params[IIR].order = 0;
         cp->filter_params[FIR].shift = 0;
         cp->filter_params[IIR].shift = 0;

         /* Default audio coding is 24-bit raw PCM. */

         cp->huff_offset      = 0;
         cp->sign_huff_offset = (-1) << 23;
         cp->codebook         = 0;
         cp->huff_lsbs        = 24;

     }
 

     if (substr == m->max_decoded_substream) {
         m->avctx->channels       = s->max_matrix_channel + 1;
         m->avctx->channel_layout = s->ch_layout;

 
         if (m->avctx->codec_id == AV_CODEC_ID_MLP && m->needs_reordering) {
             if (m->avctx->channel_layout == (AV_CH_LAYOUT_QUAD|AV_CH_LOW_FREQUENCY) ||
                 m->avctx->channel_layout == AV_CH_LAYOUT_5POINT0_BACK) {
                 int i = s->ch_assign[4];
                 s->ch_assign[4] = s->ch_assign[3];
                 s->ch_assign[3] = s->ch_assign[2];
                 s->ch_assign[2] = i;
             } else if (m->avctx->channel_layout == AV_CH_LAYOUT_5POINT1_BACK) {
                 FFSWAP(int, s->ch_assign[2], s->ch_assign[4]);
                 FFSWAP(int, s->ch_assign[3], s->ch_assign[5]);
             }
         }
 

     }

 
     return 0;
 }
 
 /** Read parameters for one of the prediction filters. */
 
 static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,

                               unsigned int substr, unsigned int channel,
                               unsigned int filter)

 {

     SubStream *s = &m->substream[substr];
     FilterParams *fp = &s->channel_params[channel].filter_params[filter];

     const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;

     const char fchar = filter ? 'I' : 'F';
     int i, order;
 

     // Filter is 0 for FIR, 1 for IIR.

     av_assert0(filter < 2);

     if (m->filter_changed[channel][filter]++ > 1) {
         av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n");

         return AVERROR_INVALIDDATA;

     }

     order = get_bits(gbp, 4);

     if (order > max_order) {

         av_log(m->avctx, AV_LOG_ERROR,

                "%cIR filter order %d is greater than maximum %d.\n",

                fchar, order, max_order);

         return AVERROR_INVALIDDATA;

     }

     fp->order = order;

 
     if (order > 0) {

         int32_t *fcoeff = s->channel_params[channel].coeff[filter];

         int coeff_bits, coeff_shift;
 

         fp->shift = get_bits(gbp, 4);

 
         coeff_bits  = get_bits(gbp, 5);
         coeff_shift = get_bits(gbp, 3);
         if (coeff_bits < 1 || coeff_bits > 16) {
             av_log(m->avctx, AV_LOG_ERROR,

                    "%cIR filter coeff_bits must be between 1 and 16.\n",

                    fchar);

             return AVERROR_INVALIDDATA;

         }
         if (coeff_bits + coeff_shift > 16) {
             av_log(m->avctx, AV_LOG_ERROR,

                    "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",

                    fchar);

             return AVERROR_INVALIDDATA;

         }
 
         for (i = 0; i < order; i++)

             fcoeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;

 
         if (get_bits1(gbp)) {
             int state_bits, state_shift;
 
             if (filter == FIR) {
                 av_log(m->avctx, AV_LOG_ERROR,

                        "FIR filter has state data specified.\n");

                 return AVERROR_INVALIDDATA;

             }
 
             state_bits  = get_bits(gbp, 4);
             state_shift = get_bits(gbp, 4);
 

             /* TODO: Check validity of state data. */

 
             for (i = 0; i < order; i++)

                 fp->state[i] = get_sbits(gbp, state_bits) << state_shift;

         }
     }
 
     return 0;
 }
 

 /** Read parameters for primitive matrices. */
 

 static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp)

 {

     SubStream *s = &m->substream[substr];

     unsigned int mat, ch;

     const int max_primitive_matrices = m->avctx->codec_id == AV_CODEC_ID_MLP

                                      ? MAX_MATRICES_MLP
                                      : MAX_MATRICES_TRUEHD;

     if (m->matrix_changed++ > 1) {
         av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n");

         return AVERROR_INVALIDDATA;

     }
 

     s->num_primitive_matrices = get_bits(gbp, 4);
 

     if (s->num_primitive_matrices > max_primitive_matrices) {
         av_log(m->avctx, AV_LOG_ERROR,
                "Number of primitive matrices cannot be greater than %d.\n",
                max_primitive_matrices);

         return AVERROR_INVALIDDATA;

     }
 

     for (mat = 0; mat < s->num_primitive_matrices; mat++) {
         int frac_bits, max_chan;
         s->matrix_out_ch[mat] = get_bits(gbp, 4);
         frac_bits             = get_bits(gbp, 4);
         s->lsb_bypass   [mat] = get_bits1(gbp);
 

         if (s->matrix_out_ch[mat] > s->max_matrix_channel) {

             av_log(m->avctx, AV_LOG_ERROR,
                     "Invalid channel %d specified as output from matrix.\n",
                     s->matrix_out_ch[mat]);

             return AVERROR_INVALIDDATA;

         }
         if (frac_bits > 14) {
             av_log(m->avctx, AV_LOG_ERROR,
                     "Too many fractional bits specified.\n");

             return AVERROR_INVALIDDATA;

         }
 
         max_chan = s->max_matrix_channel;
         if (!s->noise_type)
             max_chan+=2;
 
         for (ch = 0; ch <= max_chan; ch++) {
             int coeff_val = 0;
             if (get_bits1(gbp))
                 coeff_val = get_sbits(gbp, frac_bits + 2);
 
             s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
         }
 
         if (s->noise_type)
             s->matrix_noise_shift[mat] = get_bits(gbp, 4);
         else
             s->matrix_noise_shift[mat] = 0;
     }
 
     return 0;
 }
 

 /** Read channel parameters. */
 
 static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
                                GetBitContext *gbp, unsigned int ch)
 {

     SubStream *s = &m->substream[substr];
     ChannelParams *cp = &s->channel_params[ch];

     FilterParams *fir = &cp->filter_params[FIR];
     FilterParams *iir = &cp->filter_params[IIR];

     int ret;

 
     if (s->param_presence_flags & PARAM_FIR)
         if (get_bits1(gbp))

             if ((ret = read_filter_params(m, gbp, substr, ch, FIR)) < 0)
                 return ret;

 
     if (s->param_presence_flags & PARAM_IIR)
         if (get_bits1(gbp))

             if ((ret = read_filter_params(m, gbp, substr, ch, IIR)) < 0)
                 return ret;

     if (fir->order + iir->order > 8) {
         av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");

         return AVERROR_INVALIDDATA;

     }
 

     if (fir->order && iir->order &&
         fir->shift != iir->shift) {
         av_log(m->avctx, AV_LOG_ERROR,
                 "FIR and IIR filters must use the same precision.\n");

         return AVERROR_INVALIDDATA;

     }
     /* The FIR and IIR filters must have the same precision.

      * To simplify the filtering code, only the precision of the
      * FIR filter is considered. If only the IIR filter is employed,
      * the FIR filter precision is set to that of the IIR filter, so
      * that the filtering code can use it. */

     if (!fir->order && iir->order)
         fir->shift = iir->shift;
 
     if (s->param_presence_flags & PARAM_HUFFOFFSET)
         if (get_bits1(gbp))
             cp->huff_offset = get_sbits(gbp, 15);
 
     cp->codebook  = get_bits(gbp, 2);
     cp->huff_lsbs = get_bits(gbp, 5);
 

     if (cp->huff_lsbs > 24) {
         av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");

         cp->huff_lsbs = 0;

         return AVERROR_INVALIDDATA;

     }

     cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);

 
     return 0;
 }
 

 /** Read decoding parameters that change more often than those in the restart
  *  header. */
 
 static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
                                 unsigned int substr)
 {
     SubStream *s = &m->substream[substr];

     unsigned int ch;

     int ret;

     if (s->param_presence_flags & PARAM_PRESENCE)

         if (get_bits1(gbp))
             s->param_presence_flags = get_bits(gbp, 8);

 
     if (s->param_presence_flags & PARAM_BLOCKSIZE)
         if (get_bits1(gbp)) {
             s->blocksize = get_bits(gbp, 9);

             if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {

                 av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.\n");

                 s->blocksize = 0;

                 return AVERROR_INVALIDDATA;

             }
         }
 
     if (s->param_presence_flags & PARAM_MATRIX)

         if (get_bits1(gbp))

             if ((ret = read_matrix_params(m, substr, gbp)) < 0)
                 return ret;

 
     if (s->param_presence_flags & PARAM_OUTSHIFT)
         if (get_bits1(gbp))

             for (ch = 0; ch <= s->max_matrix_channel; ch++)

                 s->output_shift[ch] = get_sbits(gbp, 4);

 
     if (s->param_presence_flags & PARAM_QUANTSTEP)
         if (get_bits1(gbp))
             for (ch = 0; ch <= s->max_channel; ch++) {

                 ChannelParams *cp = &s->channel_params[ch];

                 s->quant_step_size[ch] = get_bits(gbp, 4);
 

                 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);

             }
 
     for (ch = s->min_channel; ch <= s->max_channel; ch++)

         if (get_bits1(gbp))

             if ((ret = read_channel_params(m, substr, gbp, ch)) < 0)
                 return ret;

 
     return 0;
 }
 
 #define MSB_MASK(bits)  (-1u << bits)
 
 /** Generate PCM samples using the prediction filters and residual values
  *  read from the data stream, and update the filter state. */
 
 static void filter_channel(MLPDecodeContext *m, unsigned int substr,
                            unsigned int channel)
 {
     SubStream *s = &m->substream[substr];

     const int32_t *fircoeff = s->channel_params[channel].coeff[FIR];

     int32_t state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FIR_ORDER];
     int32_t *firbuf = state_buffer[FIR] + MAX_BLOCKSIZE;
     int32_t *iirbuf = state_buffer[IIR] + MAX_BLOCKSIZE;

     FilterParams *fir = &s->channel_params[channel].filter_params[FIR];
     FilterParams *iir = &s->channel_params[channel].filter_params[IIR];

     unsigned int filter_shift = fir->shift;

     int32_t mask = MSB_MASK(s->quant_step_size[channel]);
 

     memcpy(firbuf, fir->state, MAX_FIR_ORDER * sizeof(int32_t));
     memcpy(iirbuf, iir->state, MAX_IIR_ORDER * sizeof(int32_t));

     m->dsp.mlp_filter_channel(firbuf, fircoeff,
                               fir->order, iir->order,

                               filter_shift, mask, s->blocksize,
                               &m->sample_buffer[s->blockpos][channel]);

     memcpy(fir->state, firbuf - s->blocksize, MAX_FIR_ORDER * sizeof(int32_t));
     memcpy(iir->state, iirbuf - s->blocksize, MAX_IIR_ORDER * sizeof(int32_t));

 }
 
 /** Read a block of PCM residual data (or actual if no filtering active). */
 
 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
                            unsigned int substr)
 {
     SubStream *s = &m->substream[substr];
     unsigned int i, ch, expected_stream_pos = 0;

     int ret;

 
     if (s->data_check_present) {
         expected_stream_pos  = get_bits_count(gbp);
         expected_stream_pos += get_bits(gbp, 16);

         av_log_ask_for_sample(m->avctx, "This file contains some features "
                               "we have not tested yet.\n");

     }
 
     if (s->blockpos + s->blocksize > m->access_unit_size) {

         av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");

         return AVERROR_INVALIDDATA;

     }
 
     memset(&m->bypassed_lsbs[s->blockpos][0], 0,
            s->blocksize * sizeof(m->bypassed_lsbs[0]));
 

     for (i = 0; i < s->blocksize; i++)

         if ((ret = read_huff_channels(m, gbp, substr, i)) < 0)
             return ret;

     for (ch = s->min_channel; ch <= s->max_channel; ch++)

         filter_channel(m, substr, ch);
 
     s->blockpos += s->blocksize;
 
     if (s->data_check_present) {
         if (get_bits_count(gbp) != expected_stream_pos)

             av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");

         skip_bits(gbp, 8);
     }
 
     return 0;
 }
 

 /** Data table used for TrueHD noise generation function. */

 
 static const int8_t noise_table[256] = {
      30,  51,  22,  54,   3,   7,  -4,  38,  14,  55,  46,  81,  22,  58,  -3,   2,
      52,  31,  -7,  51,  15,  44,  74,  30,  85, -17,  10,  33,  18,  80,  28,  62,
      10,  32,  23,  69,  72,  26,  35,  17,  73,  60,   8,  56,   2,   6,  -2,  -5,
      51,   4,  11,  50,  66,  76,  21,  44,  33,  47,   1,  26,  64,  48,  57,  40,
      38,  16, -10, -28,  92,  22, -18,  29, -10,   5, -13,  49,  19,  24,  70,  34,
      61,  48,  30,  14,  -6,  25,  58,  33,  42,  60,  67,  17,  54,  17,  22,  30,
      67,  44,  -9,  50, -11,  43,  40,  32,  59,  82,  13,  49, -14,  55,  60,  36,
      48,  49,  31,  47,  15,  12,   4,  65,   1,  23,  29,  39,  45,  -2,  84,  69,
       0,  72,  37,  57,  27,  41, -15, -16,  35,  31,  14,  61,  24,   0,  27,  24,
      16,  41,  55,  34,  53,   9,  56,  12,  25,  29,  53,   5,  20, -20,  -8,  20,
      13,  28,  -3,  78,  38,  16,  11,  62,  46,  29,  21,  24,  46,  65,  43, -23,
      89,  18,  74,  21,  38, -12,  19,  12, -19,   8,  15,  33,   4,  57,   9,  -8,
      36,  35,  26,  28,   7,  83,  63,  79,  75,  11,   3,  87,  37,  47,  34,  40,
      39,  19,  20,  42,  27,  34,  39,  77,  13,  42,  59,  64,  45,  -1,  32,  37,
      45,  -5,  53,  -6,   7,  36,  50,  23,   6,  32,   9, -21,  18,  71,  27,  52,
     -25,  31,  35,  42,  -1,  68,  63,  52,  26,  43,  66,  37,  41,  25,  40,  70,
 };
 
 /** Noise generation functions.
  *  I'm not sure what these are for - they seem to be some kind of pseudorandom
  *  sequence generators, used to generate noise data which is used when the
  *  channels are rematrixed. I'm not sure if they provide a practical benefit
  *  to compression, or just obfuscate the decoder. Are they for some kind of
  *  dithering? */
 
 /** Generate two channels of noise, used in the matrix when
  *  restart sync word == 0x31ea. */
 
 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
 {
     SubStream *s = &m->substream[substr];
     unsigned int i;
     uint32_t seed = s->noisegen_seed;
     unsigned int maxchan = s->max_matrix_channel;
 
     for (i = 0; i < s->blockpos; i++) {
         uint16_t seed_shr7 = seed >> 7;
         m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
         m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7)   << s->noise_shift;
 
         seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
     }
 
     s->noisegen_seed = seed;
 }
 
 /** Generate a block of noise, used when restart sync word == 0x31eb. */
 
 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
 {
     SubStream *s = &m->substream[substr];
     unsigned int i;
     uint32_t seed = s->noisegen_seed;
 
     for (i = 0; i < m->access_unit_size_pow2; i++) {
         uint8_t seed_shr15 = seed >> 15;
         m->noise_buffer[i] = noise_table[seed_shr15];
         seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
     }
 
     s->noisegen_seed = seed;
 }
 
 
 /** Apply the channel matrices in turn to reconstruct the original audio
  *  samples. */
 
 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
 {
     SubStream *s = &m->substream[substr];
     unsigned int mat, src_ch, i;
     unsigned int maxchan;
 
     maxchan = s->max_matrix_channel;
     if (!s->noise_type) {
         generate_2_noise_channels(m, substr);
         maxchan += 2;
     } else {
         fill_noise_buffer(m, substr);
     }
 
     for (mat = 0; mat < s->num_primitive_matrices; mat++) {
         int matrix_noise_shift = s->matrix_noise_shift[mat];
         unsigned int dest_ch = s->matrix_out_ch[mat];
         int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);

         int32_t *coeffs = s->matrix_coeff[mat];

         int index  = s->num_primitive_matrices - mat;
         int index2 = 2 * index + 1;

 
         /* TODO: DSPContext? */
 
         for (i = 0; i < s->blockpos; i++) {

             int32_t bypassed_lsb = m->bypassed_lsbs[i][mat];

             int32_t *samples = m->sample_buffer[i];

             int64_t accum = 0;

 
             for (src_ch = 0; src_ch <= maxchan; src_ch++)
                 accum += (int64_t) samples[src_ch] * coeffs[src_ch];
 

             if (matrix_noise_shift) {

                 index &= m->access_unit_size_pow2 - 1;

                 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);

                 index += index2;

             }

             samples[dest_ch] = ((accum >> 14) & mask) + bypassed_lsb;

         }
     }
 }
 
 /** Write the audio data into the output buffer. */
 

 static int output_data(MLPDecodeContext *m, unsigned int substr,

                        AVFrame *frame, int *got_frame_ptr)

 {

     AVCodecContext *avctx = m->avctx;

     SubStream *s = &m->substream[substr];

     unsigned int i, out_ch = 0;

     int32_t *data_32;
     int16_t *data_16;
     int ret;

     int is32 = (m->avctx->sample_fmt == AV_SAMPLE_FMT_S32);

     if (m->avctx->channels != s->max_matrix_channel + 1) {
         av_log(m->avctx, AV_LOG_ERROR, "channel count mismatch\n");
         return AVERROR_INVALIDDATA;
     }
 

     if (!s->blockpos) {
         av_log(avctx, AV_LOG_ERROR, "No samples to output.\n");
         return AVERROR_INVALIDDATA;
     }

     /* get output buffer */

     frame->nb_samples = s->blockpos;
     if ((ret = ff_get_buffer(avctx, frame)) < 0) {

         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
         return ret;
     }

     data_32 = (int32_t *)frame->data[0];
     data_16 = (int16_t *)frame->data[0];

 
     for (i = 0; i < s->blockpos; i++) {

         for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
             int mat_ch = s->ch_assign[out_ch];
             int32_t sample = m->sample_buffer[i][mat_ch]
                           << s->output_shift[mat_ch];
             s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;

             if (is32) *data_32++ = sample << 8;
             else      *data_16++ = sample >> 8;
         }
     }
 

     *got_frame_ptr = 1;

 
     return 0;
 }
 
 /** Read an access unit from the stream.

  *  @return negative on error, 0 if not enough data is present in the input stream,
  *  otherwise the number of bytes consumed. */

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

 {

     const uint8_t *buf = avpkt->data;
     int buf_size = avpkt->size;

     MLPDecodeContext *m = avctx->priv_data;
     GetBitContext gb;
     unsigned int length, substr;
     unsigned int substream_start;
     unsigned int header_size = 4;
     unsigned int substr_header_size = 0;
     uint8_t substream_parity_present[MAX_SUBSTREAMS];
     uint16_t substream_data_len[MAX_SUBSTREAMS];
     uint8_t parity_bits;

     int ret;

 
     if (buf_size < 4)
         return 0;
 
     length = (AV_RB16(buf) & 0xfff) * 2;
 

     if (length < 4 || length > buf_size)

         return AVERROR_INVALIDDATA;

 
     init_get_bits(&gb, (buf + 4), (length - 4) * 8);
 

     m->is_major_sync_unit = 0;

     if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
         if (read_major_sync(m, &gb) < 0)
             goto error;

         m->is_major_sync_unit = 1;

         header_size += 28;
     }
 
     if (!m->params_valid) {
         av_log(m->avctx, AV_LOG_WARNING,

                "Stream parameters not seen; skipping frame.\n");

         *got_frame_ptr = 0;

         return length;
     }
 
     substream_start = 0;
 
     for (substr = 0; substr < m->num_substreams; substr++) {

         int extraword_present, checkdata_present, end, nonrestart_substr;

 
         extraword_present = get_bits1(&gb);

         nonrestart_substr = get_bits1(&gb);

         checkdata_present = get_bits1(&gb);
         skip_bits1(&gb);
 
         end = get_bits(&gb, 12) * 2;
 
         substr_header_size += 2;
 
         if (extraword_present) {

             if (m->avctx->codec_id == AV_CODEC_ID_MLP) {

                 av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n");
                 goto error;
             }

             skip_bits(&gb, 16);
             substr_header_size += 2;
         }
 

         if (!(nonrestart_substr ^ m->is_major_sync_unit)) {
             av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n");
             goto error;
         }
 

         if (end + header_size + substr_header_size > length) {
             av_log(m->avctx, AV_LOG_ERROR,
                    "Indicated length of substream %d data goes off end of "
                    "packet.\n", substr);
             end = length - header_size - substr_header_size;
         }
 
         if (end < substream_start) {
             av_log(avctx, AV_LOG_ERROR,
                    "Indicated end offset of substream %d data "
                    "is smaller than calculated start offset.\n",
                    substr);
             goto error;
         }
 
         if (substr > m->max_decoded_substream)
             continue;
 
         substream_parity_present[substr] = checkdata_present;
         substream_data_len[substr] = end - substream_start;
         substream_start = end;
     }
 

     parity_bits  = ff_mlp_calculate_parity(buf, 4);
     parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);

 
     if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
         av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
         goto error;
     }
 
     buf += header_size + substr_header_size;
 
     for (substr = 0; substr <= m->max_decoded_substream; substr++) {
         SubStream *s = &m->substream[substr];
         init_get_bits(&gb, buf, substream_data_len[substr] * 8);
 

         m->matrix_changed = 0;
         memset(m->filter_changed, 0, sizeof(m->filter_changed));
 

         s->blockpos = 0;
         do {
             if (get_bits1(&gb)) {
                 if (get_bits1(&gb)) {

                     /* A restart header should be present. */

                     if (read_restart_header(m, &gb, buf, substr) < 0)
                         goto next_substr;
                     s->restart_seen = 1;
                 }
 

                 if (!s->restart_seen)

                     goto next_substr;
                 if (read_decoding_params(m, &gb, substr) < 0)
                     goto next_substr;
             }
 

             if (!s->restart_seen)

                 goto next_substr;
 

             if ((ret = read_block_data(m, &gb, substr)) < 0)
                 return ret;

             if (get_bits_count(&gb) >= substream_data_len[substr] * 8)
                 goto substream_length_mismatch;
 
         } while (!get_bits1(&gb));

 
         skip_bits(&gb, (-get_bits_count(&gb)) & 15);

         if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) {
             int shorten_by;
 
             if (get_bits(&gb, 16) != 0xD234)

                 return AVERROR_INVALIDDATA;

 
             shorten_by = get_bits(&gb, 16);

             if      (m->avctx->codec_id == AV_CODEC_ID_TRUEHD && shorten_by  & 0x2000)

                 s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);

             else if (m->avctx->codec_id == AV_CODEC_ID_MLP    && shorten_by != 0xD234)

                 return AVERROR_INVALIDDATA;

             if (substr == m->max_decoded_substream)

                 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");

         }

         if (substream_parity_present[substr]) {

             uint8_t parity, checksum;
 

             if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16)
                 goto substream_length_mismatch;
 

             parity   = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
             checksum = ff_mlp_checksum8       (buf, substream_data_len[substr] - 2);

             if ((get_bits(&gb, 8) ^ parity) != 0xa9    )
                 av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr);
             if ( get_bits(&gb, 8)           != checksum)
                 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n"    , substr);

         }

 
         if (substream_data_len[substr] * 8 != get_bits_count(&gb))

             goto substream_length_mismatch;

 
 next_substr:

         if (!s->restart_seen)

             av_log(m->avctx, AV_LOG_ERROR,
                    "No restart header present in substream %d.\n", substr);
 

         buf += substream_data_len[substr];
     }
 
     rematrix_channels(m, m->max_decoded_substream);
 

     if ((ret = output_data(m, m->max_decoded_substream, data, got_frame_ptr)) < 0)

         return ret;

 
     return length;
 

 substream_length_mismatch:
     av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);

     return AVERROR_INVALIDDATA;

 error:
     m->params_valid = 0;

     return AVERROR_INVALIDDATA;

 }
 

 #if CONFIG_MLP_DECODER

 AVCodec ff_mlp_decoder = {

     .name           = "mlp",
     .type           = AVMEDIA_TYPE_AUDIO,

     .id             = AV_CODEC_ID_MLP,

     .priv_data_size = sizeof(MLPDecodeContext),
     .init           = mlp_decode_init,
     .decode         = read_access_unit,

     .capabilities   = CODEC_CAP_DR1,

     .long_name      = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),

 };

 #endif

 #if CONFIG_TRUEHD_DECODER

 AVCodec ff_truehd_decoder = {

     .name           = "truehd",
     .type           = AVMEDIA_TYPE_AUDIO,

     .id             = AV_CODEC_ID_TRUEHD,

     .priv_data_size = sizeof(MLPDecodeContext),
     .init           = mlp_decode_init,
     .decode         = read_access_unit,

     .capabilities   = CODEC_CAP_DR1,

     .long_name      = NULL_IF_CONFIG_SMALL("TrueHD"),

 };
 #endif /* CONFIG_TRUEHD_DECODER */