text: /*
   * 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 "internal.h"
  #include "dsputil.h"
  #include "XMFFmpeg/libavutil/intreadwrite.h"
  #include "get_bits.h"
  #include "XMFFmpeg/libavutil/crc.h"
  #include "parser.h"
  #include "mlp_parser.h"
  #include "mlp.h"
  
  /** number of bits used for VLC lookup - longest Huffman code is 9 */
  #define VLC_BITS            9
  
  
  static const char* sample_message =
      "Please file a bug report following the instructions at "
      "http://ffmpeg.org/bugreports.html and include "
      "a sample of this file.";
  
  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];
  
      /// 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;
      AVFrame     frame;
  
      /// 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];
  
      DSPContext  dsp;
  } MLPDecodeContext;
  
  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 = (MLPDecodeContext *)avctx->priv_data;
      int substr;
  
      init_static();
      m->avctx = avctx;
      for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
          m->substream[substr].lossless_check_data = 0xffffffff;
      dsputil_init(&m->dsp, avctx);
  
      avcodec_get_frame_defaults(&m->frame);
      avctx->coded_frame = &m->frame;
  
      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 == 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(m->avctx, AV_LOG_ERROR,
                 "Number of substreams %d is larger than the maximum supported "
                 "by the decoder. %s\n", mh.num_substreams, sample_message);
          return AVERROR_INVALIDDATA;
      }
  
      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;
  
      if (mh.stream_type == 0xbb) {
          /* MLP stream */
          m->avctx->channel_layout = ff_mlp_layout[mh.channels_mlp];
      } else { /* mh.stream_type == 0xba */
          /* TrueHD stream */
          if (mh.channels_thd_stream2) {
              m->avctx->channel_layout = ff_truehd_layout(mh.channels_thd_stream2);
          } else {
              m->avctx->channel_layout = ff_truehd_layout(mh.channels_thd_stream1);
          }
          if (m->avctx->channels &&
              !m->avctx->request_channels && !m->avctx->request_channel_layout &&
              av_get_channel_layout_nb_channels(m->avctx->channel_layout) != m->avctx->channels) {
              m->avctx->channel_layout = 0;
              av_log_ask_for_sample(m->avctx, "Unknown channel layout.");
          }
      }
  
      m->needs_reordering = mh.channels_mlp >= 18 && mh.channels_mlp <= 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 == CODEC_ID_MLP
                                   ? MAX_MATRIX_CHANNEL_MLP
                                   : MAX_MATRIX_CHANNEL_TRUEHD;
  
      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 == 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 */
  
      s->min_channel        = get_bits(gbp, 4);
      s->max_channel        = get_bits(gbp, 4);
      s->max_matrix_channel = get_bits(gbp, 4);
  
      if (s->max_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 (s->max_channel != s->max_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 (s->max_channel > MAX_MATRIX_CHANNEL_MLP && !s->noise_type) {
          av_log(m->avctx, AV_LOG_ERROR,
                 "Number of channels %d is larger than the maximum supported "
                 "by the decoder. %s\n", s->max_channel+2, sample_message);
          return AVERROR_INVALIDDATA;
      }
  
      if (s->min_channel > s->max_channel) {
          av_log(m->avctx, AV_LOG_ERROR,
                 "Substream min channel cannot be greater than max channel.\n");
          return AVERROR_INVALIDDATA;
      }
  
      if (m->avctx->request_channels > 0
          && s->max_channel + 1 >= m->avctx->request_channels
          && substr < m->max_decoded_substream) {
          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;
      }
  
      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 (ch_assign > s->max_matrix_channel) {
              av_log(m->avctx, AV_LOG_ERROR,
                     "Assignment of matrix channel %d to invalid output channel %d. %s\n",
                     ch, ch_assign, sample_message);
              return AVERROR_INVALIDDATA;
          }
          s->ch_assign[ch_assign] = ch;
      }
  
      if (m->avctx->codec_id == 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]);
          }
      }
      if (m->avctx->codec_id == CODEC_ID_TRUEHD &&
          (m->avctx->channel_layout == AV_CH_LAYOUT_7POINT1 ||
          m->avctx->channel_layout == AV_CH_LAYOUT_7POINT1_WIDE)) {
          FFSWAP(int, s->ch_assign[4], s->ch_assign[6]);
          FFSWAP(int, s->ch_assign[5], s->ch_assign[7]);
      } else if (m->avctx->codec_id == CODEC_ID_TRUEHD &&
          (m->avctx->channel_layout == AV_CH_LAYOUT_6POINT1 ||
          m->avctx->channel_layout == (AV_CH_LAYOUT_6POINT1 | AV_CH_TOP_CENTER) ||
          m->avctx->channel_layout == (AV_CH_LAYOUT_6POINT1 | AV_CH_TOP_FRONT_CENTER))) {
          int i = s->ch_assign[6];
          s->ch_assign[6] = s->ch_assign[5];
          s->ch_assign[5] = s->ch_assign[4];
          s->ch_assign[4] = i;
      }
  
      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;
  
      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.
      assert(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 == 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");
          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.");
                  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(m->avctx, AV_LOG_WARNING, "This file contains some features "
                 "we have not tested yet. %s\n", sample_message);
      }
  
      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,
                         void *data, 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;
      }
  
      /* get output buffer */
      m->frame.nb_samples = s->blockpos;
      if ((ret = avctx->get_buffer(avctx, &m->frame)) < 0) {
          av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
          return ret;
      }
      data_32 = (int32_t *)m->frame.data[0];
      data_16 = (int16_t *)m->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;
      *(AVFrame *)data = m->frame;
  
      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 = (MLPDecodeContext *)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 == 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 == CODEC_ID_TRUEHD && shorten_by  & 0x2000)
                  s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
              else if (m->avctx->codec_id == 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;
  }
  
  AVCodec ff_mlp_decoder = {
      /*.name					= */	"mlp",
      /*.type					= */	AVMEDIA_TYPE_AUDIO,
      /*.id					= */	CODEC_ID_MLP,
      /*.priv_data_size		= */	sizeof(MLPDecodeContext),
      /*.init					= */	mlp_decode_init,
      /*.encode				= */	0,
      /*.close				= */	0,
      /*.decode				= */	read_access_unit,
      /*.capabilities			= */	CODEC_CAP_DR1,
  	/*.next					= */	0,
  	/*.flush				= */	0,
  	/*.supported_framerates	= */	0,
  	/*.pix_fmts				= */	0,
      /*.long_name			= */	NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
  };
  
  #if CONFIG_TRUEHD_DECODER
  AVCodec ff_truehd_decoder = {
      /*.name					= */	"truehd",
      /*.type					= */	AVMEDIA_TYPE_AUDIO,
      /*.id					= */	CODEC_ID_TRUEHD,
      /*.priv_data_size		= */	sizeof(MLPDecodeContext),
      /*.init					= */	mlp_decode_init,
      /*.encode				= */	0,
      /*.close				= */	0,
      /*.decode				= */	read_access_unit,
      /*.capabilities			= */	CODEC_CAP_DR1,
  	/*.next					= */	0,
  	/*.flush				= */	0,
  	/*.supported_framerates	= */	0,
  	/*.pix_fmts				= */	0,
      /*.long_name			= */	NULL_IF_CONFIG_SMALL("TrueHD"),
  };
  #endif /* CONFIG_TRUEHD_DECODER */
  