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  1. /*
  2.  * MLP decoder
  3.  * Copyright (c) 2007-2008 Ian Caulfield
  4.  *
  5.  * This file is part of FFmpeg.
  6.  *
  7.  * FFmpeg is free software; you can redistribute it and/or
  8.  * modify it under the terms of the GNU Lesser General Public
  9.  * License as published by the Free Software Foundation; either
  10.  * version 2.1 of the License, or (at your option) any later version.
  11.  *
  12.  * FFmpeg is distributed in the hope that it will be useful,
  13.  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15.  * Lesser General Public License for more details.
  16.  *
  17.  * You should have received a copy of the GNU Lesser General Public
  18.  * License along with FFmpeg; if not, write to the Free Software
  19.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20.  */
  21.  
  22. /**
  23.  * @file
  24.  * MLP decoder
  25.  */
  26.  
  27. #include <stdint.h>
  28.  
  29. #include "avcodec.h"
  30. #include "libavutil/internal.h"
  31. #include "libavutil/intreadwrite.h"
  32. #include "libavutil/channel_layout.h"
  33. #include "get_bits.h"
  34. #include "internal.h"
  35. #include "libavutil/crc.h"
  36. #include "parser.h"
  37. #include "mlp_parser.h"
  38. #include "mlpdsp.h"
  39. #include "mlp.h"
  40.  
  41. /** number of bits used for VLC lookup - longest Huffman code is 9 */
  42. #define VLC_BITS            9
  43.  
  44. typedef struct SubStream {
  45.     /// Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
  46.     uint8_t     restart_seen;
  47.  
  48.     //@{
  49.     /** restart header data */
  50.     /// The type of noise to be used in the rematrix stage.
  51.     uint16_t    noise_type;
  52.  
  53.     /// The index of the first channel coded in this substream.
  54.     uint8_t     min_channel;
  55.     /// The index of the last channel coded in this substream.
  56.     uint8_t     max_channel;
  57.     /// The number of channels input into the rematrix stage.
  58.     uint8_t     max_matrix_channel;
  59.     /// For each channel output by the matrix, the output channel to map it to
  60.     uint8_t     ch_assign[MAX_CHANNELS];
  61.     /// The channel layout for this substream
  62.     uint64_t    ch_layout;
  63.  
  64.     /// Channel coding parameters for channels in the substream
  65.     ChannelParams channel_params[MAX_CHANNELS];
  66.  
  67.     /// The left shift applied to random noise in 0x31ea substreams.
  68.     uint8_t     noise_shift;
  69.     /// The current seed value for the pseudorandom noise generator(s).
  70.     uint32_t    noisegen_seed;
  71.  
  72.     /// Set if the substream contains extra info to check the size of VLC blocks.
  73.     uint8_t     data_check_present;
  74.  
  75.     /// Bitmask of which parameter sets are conveyed in a decoding parameter block.
  76.     uint8_t     param_presence_flags;
  77. #define PARAM_BLOCKSIZE     (1 << 7)
  78. #define PARAM_MATRIX        (1 << 6)
  79. #define PARAM_OUTSHIFT      (1 << 5)
  80. #define PARAM_QUANTSTEP     (1 << 4)
  81. #define PARAM_FIR           (1 << 3)
  82. #define PARAM_IIR           (1 << 2)
  83. #define PARAM_HUFFOFFSET    (1 << 1)
  84. #define PARAM_PRESENCE      (1 << 0)
  85.     //@}
  86.  
  87.     //@{
  88.     /** matrix data */
  89.  
  90.     /// Number of matrices to be applied.
  91.     uint8_t     num_primitive_matrices;
  92.  
  93.     /// matrix output channel
  94.     uint8_t     matrix_out_ch[MAX_MATRICES];
  95.  
  96.     /// Whether the LSBs of the matrix output are encoded in the bitstream.
  97.     uint8_t     lsb_bypass[MAX_MATRICES];
  98.     /// Matrix coefficients, stored as 2.14 fixed point.
  99.     int32_t     matrix_coeff[MAX_MATRICES][MAX_CHANNELS];
  100.     /// Left shift to apply to noise values in 0x31eb substreams.
  101.     uint8_t     matrix_noise_shift[MAX_MATRICES];
  102.     //@}
  103.  
  104.     /// Left shift to apply to Huffman-decoded residuals.
  105.     uint8_t     quant_step_size[MAX_CHANNELS];
  106.  
  107.     /// number of PCM samples in current audio block
  108.     uint16_t    blocksize;
  109.     /// Number of PCM samples decoded so far in this frame.
  110.     uint16_t    blockpos;
  111.  
  112.     /// Left shift to apply to decoded PCM values to get final 24-bit output.
  113.     int8_t      output_shift[MAX_CHANNELS];
  114.  
  115.     /// Running XOR of all output samples.
  116.     int32_t     lossless_check_data;
  117.  
  118. } SubStream;
  119.  
  120. typedef struct MLPDecodeContext {
  121.     AVCodecContext *avctx;
  122.  
  123.     /// Current access unit being read has a major sync.
  124.     int         is_major_sync_unit;
  125.  
  126.     /// Set if a valid major sync block has been read. Otherwise no decoding is possible.
  127.     uint8_t     params_valid;
  128.  
  129.     /// Number of substreams contained within this stream.
  130.     uint8_t     num_substreams;
  131.  
  132.     /// Index of the last substream to decode - further substreams are skipped.
  133.     uint8_t     max_decoded_substream;
  134.  
  135.     /// Stream needs channel reordering to comply with FFmpeg's channel order
  136.     uint8_t     needs_reordering;
  137.  
  138.     /// number of PCM samples contained in each frame
  139.     int         access_unit_size;
  140.     /// next power of two above the number of samples in each frame
  141.     int         access_unit_size_pow2;
  142.  
  143.     SubStream   substream[MAX_SUBSTREAMS];
  144.  
  145.     int         matrix_changed;
  146.     int         filter_changed[MAX_CHANNELS][NUM_FILTERS];
  147.  
  148.     int8_t      noise_buffer[MAX_BLOCKSIZE_POW2];
  149.     int8_t      bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
  150.     int32_t     sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS];
  151.  
  152.     MLPDSPContext dsp;
  153. } MLPDecodeContext;
  154.  
  155. static const uint64_t thd_channel_order[] = {
  156.     AV_CH_FRONT_LEFT, AV_CH_FRONT_RIGHT,                     // LR
  157.     AV_CH_FRONT_CENTER,                                      // C
  158.     AV_CH_LOW_FREQUENCY,                                     // LFE
  159.     AV_CH_SIDE_LEFT, AV_CH_SIDE_RIGHT,                       // LRs
  160.     AV_CH_TOP_FRONT_LEFT, AV_CH_TOP_FRONT_RIGHT,             // LRvh
  161.     AV_CH_FRONT_LEFT_OF_CENTER, AV_CH_FRONT_RIGHT_OF_CENTER, // LRc
  162.     AV_CH_BACK_LEFT, AV_CH_BACK_RIGHT,                       // LRrs
  163.     AV_CH_BACK_CENTER,                                       // Cs
  164.     AV_CH_TOP_CENTER,                                        // Ts
  165.     AV_CH_SURROUND_DIRECT_LEFT, AV_CH_SURROUND_DIRECT_RIGHT, // LRsd
  166.     AV_CH_WIDE_LEFT, AV_CH_WIDE_RIGHT,                       // LRw
  167.     AV_CH_TOP_FRONT_CENTER,                                  // Cvh
  168.     AV_CH_LOW_FREQUENCY_2,                                   // LFE2
  169. };
  170.  
  171. static uint64_t thd_channel_layout_extract_channel(uint64_t channel_layout,
  172.                                                    int index)
  173. {
  174.     int i;
  175.  
  176.     if (av_get_channel_layout_nb_channels(channel_layout) <= index)
  177.         return 0;
  178.  
  179.     for (i = 0; i < FF_ARRAY_ELEMS(thd_channel_order); i++)
  180.         if (channel_layout & thd_channel_order[i] && !index--)
  181.             return thd_channel_order[i];
  182.     return 0;
  183. }
  184.  
  185. static VLC huff_vlc[3];
  186.  
  187. /** Initialize static data, constant between all invocations of the codec. */
  188.  
  189. static av_cold void init_static(void)
  190. {
  191.     if (!huff_vlc[0].bits) {
  192.         INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
  193.                     &ff_mlp_huffman_tables[0][0][1], 2, 1,
  194.                     &ff_mlp_huffman_tables[0][0][0], 2, 1, 512);
  195.         INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
  196.                     &ff_mlp_huffman_tables[1][0][1], 2, 1,
  197.                     &ff_mlp_huffman_tables[1][0][0], 2, 1, 512);
  198.         INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
  199.                     &ff_mlp_huffman_tables[2][0][1], 2, 1,
  200.                     &ff_mlp_huffman_tables[2][0][0], 2, 1, 512);
  201.     }
  202.  
  203.     ff_mlp_init_crc();
  204. }
  205.  
  206. static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
  207.                                           unsigned int substr, unsigned int ch)
  208. {
  209.     SubStream *s = &m->substream[substr];
  210.     ChannelParams *cp = &s->channel_params[ch];
  211.     int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
  212.     int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
  213.     int32_t sign_huff_offset = cp->huff_offset;
  214.  
  215.     if (cp->codebook > 0)
  216.         sign_huff_offset -= 7 << lsb_bits;
  217.  
  218.     if (sign_shift >= 0)
  219.         sign_huff_offset -= 1 << sign_shift;
  220.  
  221.     return sign_huff_offset;
  222. }
  223.  
  224. /** Read a sample, consisting of either, both or neither of entropy-coded MSBs
  225.  *  and plain LSBs. */
  226.  
  227. static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
  228.                                      unsigned int substr, unsigned int pos)
  229. {
  230.     SubStream *s = &m->substream[substr];
  231.     unsigned int mat, channel;
  232.  
  233.     for (mat = 0; mat < s->num_primitive_matrices; mat++)
  234.         if (s->lsb_bypass[mat])
  235.             m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
  236.  
  237.     for (channel = s->min_channel; channel <= s->max_channel; channel++) {
  238.         ChannelParams *cp = &s->channel_params[channel];
  239.         int codebook = cp->codebook;
  240.         int quant_step_size = s->quant_step_size[channel];
  241.         int lsb_bits = cp->huff_lsbs - quant_step_size;
  242.         int result = 0;
  243.  
  244.         if (codebook > 0)
  245.             result = get_vlc2(gbp, huff_vlc[codebook-1].table,
  246.                             VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
  247.  
  248.         if (result < 0)
  249.             return AVERROR_INVALIDDATA;
  250.  
  251.         if (lsb_bits > 0)
  252.             result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
  253.  
  254.         result  += cp->sign_huff_offset;
  255.         result <<= quant_step_size;
  256.  
  257.         m->sample_buffer[pos + s->blockpos][channel] = result;
  258.     }
  259.  
  260.     return 0;
  261. }
  262.  
  263. static av_cold int mlp_decode_init(AVCodecContext *avctx)
  264. {
  265.     MLPDecodeContext *m = avctx->priv_data;
  266.     int substr;
  267.  
  268.     init_static();
  269.     m->avctx = avctx;
  270.     for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
  271.         m->substream[substr].lossless_check_data = 0xffffffff;
  272.     ff_mlpdsp_init(&m->dsp);
  273.  
  274.     return 0;
  275. }
  276.  
  277. /** Read a major sync info header - contains high level information about
  278.  *  the stream - sample rate, channel arrangement etc. Most of this
  279.  *  information is not actually necessary for decoding, only for playback.
  280.  */
  281.  
  282. static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
  283. {
  284.     MLPHeaderInfo mh;
  285.     int substr, ret;
  286.  
  287.     if ((ret = ff_mlp_read_major_sync(m->avctx, &mh, gb)) != 0)
  288.         return ret;
  289.  
  290.     if (mh.group1_bits == 0) {
  291.         av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
  292.         return AVERROR_INVALIDDATA;
  293.     }
  294.     if (mh.group2_bits > mh.group1_bits) {
  295.         av_log(m->avctx, AV_LOG_ERROR,
  296.                "Channel group 2 cannot have more bits per sample than group 1.\n");
  297.         return AVERROR_INVALIDDATA;
  298.     }
  299.  
  300.     if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
  301.         av_log(m->avctx, AV_LOG_ERROR,
  302.                "Channel groups with differing sample rates are not currently supported.\n");
  303.         return AVERROR_INVALIDDATA;
  304.     }
  305.  
  306.     if (mh.group1_samplerate == 0) {
  307.         av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
  308.         return AVERROR_INVALIDDATA;
  309.     }
  310.     if (mh.group1_samplerate > MAX_SAMPLERATE) {
  311.         av_log(m->avctx, AV_LOG_ERROR,
  312.                "Sampling rate %d is greater than the supported maximum (%d).\n",
  313.                mh.group1_samplerate, MAX_SAMPLERATE);
  314.         return AVERROR_INVALIDDATA;
  315.     }
  316.     if (mh.access_unit_size > MAX_BLOCKSIZE) {
  317.         av_log(m->avctx, AV_LOG_ERROR,
  318.                "Block size %d is greater than the supported maximum (%d).\n",
  319.                mh.access_unit_size, MAX_BLOCKSIZE);
  320.         return AVERROR_INVALIDDATA;
  321.     }
  322.     if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
  323.         av_log(m->avctx, AV_LOG_ERROR,
  324.                "Block size pow2 %d is greater than the supported maximum (%d).\n",
  325.                mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
  326.         return AVERROR_INVALIDDATA;
  327.     }
  328.  
  329.     if (mh.num_substreams == 0)
  330.         return AVERROR_INVALIDDATA;
  331.     if (m->avctx->codec_id == AV_CODEC_ID_MLP && mh.num_substreams > 2) {
  332.         av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n");
  333.         return AVERROR_INVALIDDATA;
  334.     }
  335.     if (mh.num_substreams > MAX_SUBSTREAMS) {
  336.         avpriv_request_sample(m->avctx,
  337.                               "%d substreams (more than the "
  338.                               "maximum supported by the decoder)",
  339.                               mh.num_substreams);
  340.         return AVERROR_PATCHWELCOME;
  341.     }
  342.  
  343.     m->access_unit_size      = mh.access_unit_size;
  344.     m->access_unit_size_pow2 = mh.access_unit_size_pow2;
  345.  
  346.     m->num_substreams        = mh.num_substreams;
  347.     m->max_decoded_substream = m->num_substreams - 1;
  348.  
  349.     m->avctx->sample_rate    = mh.group1_samplerate;
  350.     m->avctx->frame_size     = mh.access_unit_size;
  351.  
  352.     m->avctx->bits_per_raw_sample = mh.group1_bits;
  353.     if (mh.group1_bits > 16)
  354.         m->avctx->sample_fmt = AV_SAMPLE_FMT_S32;
  355.     else
  356.         m->avctx->sample_fmt = AV_SAMPLE_FMT_S16;
  357.  
  358.     m->params_valid = 1;
  359.     for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
  360.         m->substream[substr].restart_seen = 0;
  361.  
  362.     /* Set the layout for each substream. When there's more than one, the first
  363.      * substream is Stereo. Subsequent substreams' layouts are indicated in the
  364.      * major sync. */
  365.     if (m->avctx->codec_id == AV_CODEC_ID_MLP) {
  366.         if ((substr = (mh.num_substreams > 1)))
  367.             m->substream[0].ch_layout = AV_CH_LAYOUT_STEREO;
  368.         m->substream[substr].ch_layout = mh.channel_layout_mlp;
  369.     } else {
  370.         if ((substr = (mh.num_substreams > 1)))
  371.             m->substream[0].ch_layout = AV_CH_LAYOUT_STEREO;
  372.         if (mh.num_substreams > 2)
  373.             if (mh.channel_layout_thd_stream2)
  374.                 m->substream[2].ch_layout = mh.channel_layout_thd_stream2;
  375.             else
  376.                 m->substream[2].ch_layout = mh.channel_layout_thd_stream1;
  377.         m->substream[substr].ch_layout = mh.channel_layout_thd_stream1;
  378.  
  379.         if (m->avctx->channels<=2 && m->substream[substr].ch_layout == AV_CH_LAYOUT_MONO && m->max_decoded_substream == 1) {
  380.             av_log(m->avctx, AV_LOG_DEBUG, "Mono stream with 2 substreams, ignoring 2nd\n");
  381.             m->max_decoded_substream = 0;
  382.             if (m->avctx->channels==2)
  383.                 m->avctx->channel_layout = AV_CH_LAYOUT_STEREO;
  384.         }
  385.     }
  386.  
  387.     m->needs_reordering = mh.channel_arrangement >= 18 && mh.channel_arrangement <= 20;
  388.  
  389.     return 0;
  390. }
  391.  
  392. /** Read a restart header from a block in a substream. This contains parameters
  393.  *  required to decode the audio that do not change very often. Generally
  394.  *  (always) present only in blocks following a major sync. */
  395.  
  396. static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
  397.                                const uint8_t *buf, unsigned int substr)
  398. {
  399.     SubStream *s = &m->substream[substr];
  400.     unsigned int ch;
  401.     int sync_word, tmp;
  402.     uint8_t checksum;
  403.     uint8_t lossless_check;
  404.     int start_count = get_bits_count(gbp);
  405.     int min_channel, max_channel, max_matrix_channel;
  406.     const int std_max_matrix_channel = m->avctx->codec_id == AV_CODEC_ID_MLP
  407.                                      ? MAX_MATRIX_CHANNEL_MLP
  408.                                      : MAX_MATRIX_CHANNEL_TRUEHD;
  409.  
  410.     sync_word = get_bits(gbp, 13);
  411.  
  412.     if (sync_word != 0x31ea >> 1) {
  413.         av_log(m->avctx, AV_LOG_ERROR,
  414.                "restart header sync incorrect (got 0x%04x)\n", sync_word);
  415.         return AVERROR_INVALIDDATA;
  416.     }
  417.  
  418.     s->noise_type = get_bits1(gbp);
  419.  
  420.     if (m->avctx->codec_id == AV_CODEC_ID_MLP && s->noise_type) {
  421.         av_log(m->avctx, AV_LOG_ERROR, "MLP must have 0x31ea sync word.\n");
  422.         return AVERROR_INVALIDDATA;
  423.     }
  424.  
  425.     skip_bits(gbp, 16); /* Output timestamp */
  426.  
  427.     min_channel        = get_bits(gbp, 4);
  428.     max_channel        = get_bits(gbp, 4);
  429.     max_matrix_channel = get_bits(gbp, 4);
  430.  
  431.     if (max_matrix_channel > std_max_matrix_channel) {
  432.         av_log(m->avctx, AV_LOG_ERROR,
  433.                "Max matrix channel cannot be greater than %d.\n",
  434.                std_max_matrix_channel);
  435.         return AVERROR_INVALIDDATA;
  436.     }
  437.  
  438.     if (max_channel != max_matrix_channel) {
  439.         av_log(m->avctx, AV_LOG_ERROR,
  440.                "Max channel must be equal max matrix channel.\n");
  441.         return AVERROR_INVALIDDATA;
  442.     }
  443.  
  444.     /* This should happen for TrueHD streams with >6 channels and MLP's noise
  445.      * type. It is not yet known if this is allowed. */
  446.     if (max_channel > MAX_MATRIX_CHANNEL_MLP && !s->noise_type) {
  447.         avpriv_request_sample(m->avctx,
  448.                               "%d channels (more than the "
  449.                               "maximum supported by the decoder)",
  450.                               max_channel + 2);
  451.         return AVERROR_PATCHWELCOME;
  452.     }
  453.  
  454.     if (min_channel > max_channel) {
  455.         av_log(m->avctx, AV_LOG_ERROR,
  456.                "Substream min channel cannot be greater than max channel.\n");
  457.         return AVERROR_INVALIDDATA;
  458.     }
  459.  
  460.     s->min_channel        = min_channel;
  461.     s->max_channel        = max_channel;
  462.     s->max_matrix_channel = max_matrix_channel;
  463.  
  464. #if FF_API_REQUEST_CHANNELS
  465. FF_DISABLE_DEPRECATION_WARNINGS
  466.     if (m->avctx->request_channels > 0 &&
  467.         m->avctx->request_channels <= s->max_channel + 1 &&
  468.         m->max_decoded_substream > substr) {
  469.         av_log(m->avctx, AV_LOG_DEBUG,
  470.                "Extracting %d-channel downmix from substream %d. "
  471.                "Further substreams will be skipped.\n",
  472.                s->max_channel + 1, substr);
  473.         m->max_decoded_substream = substr;
  474. FF_ENABLE_DEPRECATION_WARNINGS
  475.     } else
  476. #endif
  477.     if (m->avctx->request_channel_layout == s->ch_layout &&
  478.         m->max_decoded_substream > substr) {
  479.         av_log(m->avctx, AV_LOG_DEBUG,
  480.                "Extracting %d-channel downmix (0x%"PRIx64") from substream %d. "
  481.                "Further substreams will be skipped.\n",
  482.                s->max_channel + 1, s->ch_layout, substr);
  483.         m->max_decoded_substream = substr;
  484.     }
  485.  
  486.     s->noise_shift   = get_bits(gbp,  4);
  487.     s->noisegen_seed = get_bits(gbp, 23);
  488.  
  489.     skip_bits(gbp, 19);
  490.  
  491.     s->data_check_present = get_bits1(gbp);
  492.     lossless_check = get_bits(gbp, 8);
  493.     if (substr == m->max_decoded_substream
  494.         && s->lossless_check_data != 0xffffffff) {
  495.         tmp = xor_32_to_8(s->lossless_check_data);
  496.         if (tmp != lossless_check)
  497.             av_log(m->avctx, AV_LOG_WARNING,
  498.                    "Lossless check failed - expected %02x, calculated %02x.\n",
  499.                    lossless_check, tmp);
  500.     }
  501.  
  502.     skip_bits(gbp, 16);
  503.  
  504.     memset(s->ch_assign, 0, sizeof(s->ch_assign));
  505.  
  506.     for (ch = 0; ch <= s->max_matrix_channel; ch++) {
  507.         int ch_assign = get_bits(gbp, 6);
  508.         if (m->avctx->codec_id == AV_CODEC_ID_TRUEHD) {
  509.             uint64_t channel = thd_channel_layout_extract_channel(s->ch_layout,
  510.                                                                   ch_assign);
  511.             ch_assign = av_get_channel_layout_channel_index(s->ch_layout,
  512.                                                             channel);
  513.         }
  514.         if ((unsigned)ch_assign > s->max_matrix_channel) {
  515.             avpriv_request_sample(m->avctx,
  516.                                   "Assignment of matrix channel %d to invalid output channel %d",
  517.                                   ch, ch_assign);
  518.             return AVERROR_PATCHWELCOME;
  519.         }
  520.         s->ch_assign[ch_assign] = ch;
  521.     }
  522.  
  523.     checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
  524.  
  525.     if (checksum != get_bits(gbp, 8))
  526.         av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
  527.  
  528.     /* Set default decoding parameters. */
  529.     s->param_presence_flags   = 0xff;
  530.     s->num_primitive_matrices = 0;
  531.     s->blocksize              = 8;
  532.     s->lossless_check_data    = 0;
  533.  
  534.     memset(s->output_shift   , 0, sizeof(s->output_shift   ));
  535.     memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
  536.  
  537.     for (ch = s->min_channel; ch <= s->max_channel; ch++) {
  538.         ChannelParams *cp = &s->channel_params[ch];
  539.         cp->filter_params[FIR].order = 0;
  540.         cp->filter_params[IIR].order = 0;
  541.         cp->filter_params[FIR].shift = 0;
  542.         cp->filter_params[IIR].shift = 0;
  543.  
  544.         /* Default audio coding is 24-bit raw PCM. */
  545.         cp->huff_offset      = 0;
  546.         cp->sign_huff_offset = (-1) << 23;
  547.         cp->codebook         = 0;
  548.         cp->huff_lsbs        = 24;
  549.     }
  550.  
  551.     if (substr == m->max_decoded_substream) {
  552.         m->avctx->channels       = s->max_matrix_channel + 1;
  553.         m->avctx->channel_layout = s->ch_layout;
  554.  
  555.         if (m->avctx->codec_id == AV_CODEC_ID_MLP && m->needs_reordering) {
  556.             if (m->avctx->channel_layout == (AV_CH_LAYOUT_QUAD|AV_CH_LOW_FREQUENCY) ||
  557.                 m->avctx->channel_layout == AV_CH_LAYOUT_5POINT0_BACK) {
  558.                 int i = s->ch_assign[4];
  559.                 s->ch_assign[4] = s->ch_assign[3];
  560.                 s->ch_assign[3] = s->ch_assign[2];
  561.                 s->ch_assign[2] = i;
  562.             } else if (m->avctx->channel_layout == AV_CH_LAYOUT_5POINT1_BACK) {
  563.                 FFSWAP(int, s->ch_assign[2], s->ch_assign[4]);
  564.                 FFSWAP(int, s->ch_assign[3], s->ch_assign[5]);
  565.             }
  566.         }
  567.  
  568.     }
  569.  
  570.     return 0;
  571. }
  572.  
  573. /** Read parameters for one of the prediction filters. */
  574.  
  575. static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
  576.                               unsigned int substr, unsigned int channel,
  577.                               unsigned int filter)
  578. {
  579.     SubStream *s = &m->substream[substr];
  580.     FilterParams *fp = &s->channel_params[channel].filter_params[filter];
  581.     const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;
  582.     const char fchar = filter ? 'I' : 'F';
  583.     int i, order;
  584.  
  585.     // Filter is 0 for FIR, 1 for IIR.
  586.     av_assert0(filter < 2);
  587.  
  588.     if (m->filter_changed[channel][filter]++ > 1) {
  589.         av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n");
  590.         return AVERROR_INVALIDDATA;
  591.     }
  592.  
  593.     order = get_bits(gbp, 4);
  594.     if (order > max_order) {
  595.         av_log(m->avctx, AV_LOG_ERROR,
  596.                "%cIR filter order %d is greater than maximum %d.\n",
  597.                fchar, order, max_order);
  598.         return AVERROR_INVALIDDATA;
  599.     }
  600.     fp->order = order;
  601.  
  602.     if (order > 0) {
  603.         int32_t *fcoeff = s->channel_params[channel].coeff[filter];
  604.         int coeff_bits, coeff_shift;
  605.  
  606.         fp->shift = get_bits(gbp, 4);
  607.  
  608.         coeff_bits  = get_bits(gbp, 5);
  609.         coeff_shift = get_bits(gbp, 3);
  610.         if (coeff_bits < 1 || coeff_bits > 16) {
  611.             av_log(m->avctx, AV_LOG_ERROR,
  612.                    "%cIR filter coeff_bits must be between 1 and 16.\n",
  613.                    fchar);
  614.             return AVERROR_INVALIDDATA;
  615.         }
  616.         if (coeff_bits + coeff_shift > 16) {
  617.             av_log(m->avctx, AV_LOG_ERROR,
  618.                    "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
  619.                    fchar);
  620.             return AVERROR_INVALIDDATA;
  621.         }
  622.  
  623.         for (i = 0; i < order; i++)
  624.             fcoeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
  625.  
  626.         if (get_bits1(gbp)) {
  627.             int state_bits, state_shift;
  628.  
  629.             if (filter == FIR) {
  630.                 av_log(m->avctx, AV_LOG_ERROR,
  631.                        "FIR filter has state data specified.\n");
  632.                 return AVERROR_INVALIDDATA;
  633.             }
  634.  
  635.             state_bits  = get_bits(gbp, 4);
  636.             state_shift = get_bits(gbp, 4);
  637.  
  638.             /* TODO: Check validity of state data. */
  639.  
  640.             for (i = 0; i < order; i++)
  641.                 fp->state[i] = state_bits ? get_sbits(gbp, state_bits) << state_shift : 0;
  642.         }
  643.     }
  644.  
  645.     return 0;
  646. }
  647.  
  648. /** Read parameters for primitive matrices. */
  649.  
  650. static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp)
  651. {
  652.     SubStream *s = &m->substream[substr];
  653.     unsigned int mat, ch;
  654.     const int max_primitive_matrices = m->avctx->codec_id == AV_CODEC_ID_MLP
  655.                                      ? MAX_MATRICES_MLP
  656.                                      : MAX_MATRICES_TRUEHD;
  657.  
  658.     if (m->matrix_changed++ > 1) {
  659.         av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n");
  660.         return AVERROR_INVALIDDATA;
  661.     }
  662.  
  663.     s->num_primitive_matrices = get_bits(gbp, 4);
  664.  
  665.     if (s->num_primitive_matrices > max_primitive_matrices) {
  666.         av_log(m->avctx, AV_LOG_ERROR,
  667.                "Number of primitive matrices cannot be greater than %d.\n",
  668.                max_primitive_matrices);
  669.         return AVERROR_INVALIDDATA;
  670.     }
  671.  
  672.     for (mat = 0; mat < s->num_primitive_matrices; mat++) {
  673.         int frac_bits, max_chan;
  674.         s->matrix_out_ch[mat] = get_bits(gbp, 4);
  675.         frac_bits             = get_bits(gbp, 4);
  676.         s->lsb_bypass   [mat] = get_bits1(gbp);
  677.  
  678.         if (s->matrix_out_ch[mat] > s->max_matrix_channel) {
  679.             av_log(m->avctx, AV_LOG_ERROR,
  680.                     "Invalid channel %d specified as output from matrix.\n",
  681.                     s->matrix_out_ch[mat]);
  682.             return AVERROR_INVALIDDATA;
  683.         }
  684.         if (frac_bits > 14) {
  685.             av_log(m->avctx, AV_LOG_ERROR,
  686.                     "Too many fractional bits specified.\n");
  687.             return AVERROR_INVALIDDATA;
  688.         }
  689.  
  690.         max_chan = s->max_matrix_channel;
  691.         if (!s->noise_type)
  692.             max_chan+=2;
  693.  
  694.         for (ch = 0; ch <= max_chan; ch++) {
  695.             int coeff_val = 0;
  696.             if (get_bits1(gbp))
  697.                 coeff_val = get_sbits(gbp, frac_bits + 2);
  698.  
  699.             s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
  700.         }
  701.  
  702.         if (s->noise_type)
  703.             s->matrix_noise_shift[mat] = get_bits(gbp, 4);
  704.         else
  705.             s->matrix_noise_shift[mat] = 0;
  706.     }
  707.  
  708.     return 0;
  709. }
  710.  
  711. /** Read channel parameters. */
  712.  
  713. static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
  714.                                GetBitContext *gbp, unsigned int ch)
  715. {
  716.     SubStream *s = &m->substream[substr];
  717.     ChannelParams *cp = &s->channel_params[ch];
  718.     FilterParams *fir = &cp->filter_params[FIR];
  719.     FilterParams *iir = &cp->filter_params[IIR];
  720.     int ret;
  721.  
  722.     if (s->param_presence_flags & PARAM_FIR)
  723.         if (get_bits1(gbp))
  724.             if ((ret = read_filter_params(m, gbp, substr, ch, FIR)) < 0)
  725.                 return ret;
  726.  
  727.     if (s->param_presence_flags & PARAM_IIR)
  728.         if (get_bits1(gbp))
  729.             if ((ret = read_filter_params(m, gbp, substr, ch, IIR)) < 0)
  730.                 return ret;
  731.  
  732.     if (fir->order + iir->order > 8) {
  733.         av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
  734.         return AVERROR_INVALIDDATA;
  735.     }
  736.  
  737.     if (fir->order && iir->order &&
  738.         fir->shift != iir->shift) {
  739.         av_log(m->avctx, AV_LOG_ERROR,
  740.                 "FIR and IIR filters must use the same precision.\n");
  741.         return AVERROR_INVALIDDATA;
  742.     }
  743.     /* The FIR and IIR filters must have the same precision.
  744.      * To simplify the filtering code, only the precision of the
  745.      * FIR filter is considered. If only the IIR filter is employed,
  746.      * the FIR filter precision is set to that of the IIR filter, so
  747.      * that the filtering code can use it. */
  748.     if (!fir->order && iir->order)
  749.         fir->shift = iir->shift;
  750.  
  751.     if (s->param_presence_flags & PARAM_HUFFOFFSET)
  752.         if (get_bits1(gbp))
  753.             cp->huff_offset = get_sbits(gbp, 15);
  754.  
  755.     cp->codebook  = get_bits(gbp, 2);
  756.     cp->huff_lsbs = get_bits(gbp, 5);
  757.  
  758.     if (cp->huff_lsbs > 24) {
  759.         av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
  760.         cp->huff_lsbs = 0;
  761.         return AVERROR_INVALIDDATA;
  762.     }
  763.  
  764.     cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
  765.  
  766.     return 0;
  767. }
  768.  
  769. /** Read decoding parameters that change more often than those in the restart
  770.  *  header. */
  771.  
  772. static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
  773.                                 unsigned int substr)
  774. {
  775.     SubStream *s = &m->substream[substr];
  776.     unsigned int ch;
  777.     int ret;
  778.  
  779.     if (s->param_presence_flags & PARAM_PRESENCE)
  780.         if (get_bits1(gbp))
  781.             s->param_presence_flags = get_bits(gbp, 8);
  782.  
  783.     if (s->param_presence_flags & PARAM_BLOCKSIZE)
  784.         if (get_bits1(gbp)) {
  785.             s->blocksize = get_bits(gbp, 9);
  786.             if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
  787.                 av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.\n");
  788.                 s->blocksize = 0;
  789.                 return AVERROR_INVALIDDATA;
  790.             }
  791.         }
  792.  
  793.     if (s->param_presence_flags & PARAM_MATRIX)
  794.         if (get_bits1(gbp))
  795.             if ((ret = read_matrix_params(m, substr, gbp)) < 0)
  796.                 return ret;
  797.  
  798.     if (s->param_presence_flags & PARAM_OUTSHIFT)
  799.         if (get_bits1(gbp))
  800.             for (ch = 0; ch <= s->max_matrix_channel; ch++)
  801.                 s->output_shift[ch] = get_sbits(gbp, 4);
  802.  
  803.     if (s->param_presence_flags & PARAM_QUANTSTEP)
  804.         if (get_bits1(gbp))
  805.             for (ch = 0; ch <= s->max_channel; ch++) {
  806.                 ChannelParams *cp = &s->channel_params[ch];
  807.  
  808.                 s->quant_step_size[ch] = get_bits(gbp, 4);
  809.  
  810.                 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
  811.             }
  812.  
  813.     for (ch = s->min_channel; ch <= s->max_channel; ch++)
  814.         if (get_bits1(gbp))
  815.             if ((ret = read_channel_params(m, substr, gbp, ch)) < 0)
  816.                 return ret;
  817.  
  818.     return 0;
  819. }
  820.  
  821. #define MSB_MASK(bits)  (-1u << bits)
  822.  
  823. /** Generate PCM samples using the prediction filters and residual values
  824.  *  read from the data stream, and update the filter state. */
  825.  
  826. static void filter_channel(MLPDecodeContext *m, unsigned int substr,
  827.                            unsigned int channel)
  828. {
  829.     SubStream *s = &m->substream[substr];
  830.     const int32_t *fircoeff = s->channel_params[channel].coeff[FIR];
  831.     int32_t state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FIR_ORDER];
  832.     int32_t *firbuf = state_buffer[FIR] + MAX_BLOCKSIZE;
  833.     int32_t *iirbuf = state_buffer[IIR] + MAX_BLOCKSIZE;
  834.     FilterParams *fir = &s->channel_params[channel].filter_params[FIR];
  835.     FilterParams *iir = &s->channel_params[channel].filter_params[IIR];
  836.     unsigned int filter_shift = fir->shift;
  837.     int32_t mask = MSB_MASK(s->quant_step_size[channel]);
  838.  
  839.     memcpy(firbuf, fir->state, MAX_FIR_ORDER * sizeof(int32_t));
  840.     memcpy(iirbuf, iir->state, MAX_IIR_ORDER * sizeof(int32_t));
  841.  
  842.     m->dsp.mlp_filter_channel(firbuf, fircoeff,
  843.                               fir->order, iir->order,
  844.                               filter_shift, mask, s->blocksize,
  845.                               &m->sample_buffer[s->blockpos][channel]);
  846.  
  847.     memcpy(fir->state, firbuf - s->blocksize, MAX_FIR_ORDER * sizeof(int32_t));
  848.     memcpy(iir->state, iirbuf - s->blocksize, MAX_IIR_ORDER * sizeof(int32_t));
  849. }
  850.  
  851. /** Read a block of PCM residual data (or actual if no filtering active). */
  852.  
  853. static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
  854.                            unsigned int substr)
  855. {
  856.     SubStream *s = &m->substream[substr];
  857.     unsigned int i, ch, expected_stream_pos = 0;
  858.     int ret;
  859.  
  860.     if (s->data_check_present) {
  861.         expected_stream_pos  = get_bits_count(gbp);
  862.         expected_stream_pos += get_bits(gbp, 16);
  863.         avpriv_request_sample(m->avctx,
  864.                               "Substreams with VLC block size check info");
  865.     }
  866.  
  867.     if (s->blockpos + s->blocksize > m->access_unit_size) {
  868.         av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
  869.         return AVERROR_INVALIDDATA;
  870.     }
  871.  
  872.     memset(&m->bypassed_lsbs[s->blockpos][0], 0,
  873.            s->blocksize * sizeof(m->bypassed_lsbs[0]));
  874.  
  875.     for (i = 0; i < s->blocksize; i++)
  876.         if ((ret = read_huff_channels(m, gbp, substr, i)) < 0)
  877.             return ret;
  878.  
  879.     for (ch = s->min_channel; ch <= s->max_channel; ch++)
  880.         filter_channel(m, substr, ch);
  881.  
  882.     s->blockpos += s->blocksize;
  883.  
  884.     if (s->data_check_present) {
  885.         if (get_bits_count(gbp) != expected_stream_pos)
  886.             av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
  887.         skip_bits(gbp, 8);
  888.     }
  889.  
  890.     return 0;
  891. }
  892.  
  893. /** Data table used for TrueHD noise generation function. */
  894.  
  895. static const int8_t noise_table[256] = {
  896.      30,  51,  22,  54,   3,   7,  -4,  38,  14,  55,  46,  81,  22,  58,  -3,   2,
  897.      52,  31,  -7,  51,  15,  44,  74,  30,  85, -17,  10,  33,  18,  80,  28,  62,
  898.      10,  32,  23,  69,  72,  26,  35,  17,  73,  60,   8,  56,   2,   6,  -2,  -5,
  899.      51,   4,  11,  50,  66,  76,  21,  44,  33,  47,   1,  26,  64,  48,  57,  40,
  900.      38,  16, -10, -28,  92,  22, -18,  29, -10,   5, -13,  49,  19,  24,  70,  34,
  901.      61,  48,  30,  14,  -6,  25,  58,  33,  42,  60,  67,  17,  54,  17,  22,  30,
  902.      67,  44,  -9,  50, -11,  43,  40,  32,  59,  82,  13,  49, -14,  55,  60,  36,
  903.      48,  49,  31,  47,  15,  12,   4,  65,   1,  23,  29,  39,  45,  -2,  84,  69,
  904.       0,  72,  37,  57,  27,  41, -15, -16,  35,  31,  14,  61,  24,   0,  27,  24,
  905.      16,  41,  55,  34,  53,   9,  56,  12,  25,  29,  53,   5,  20, -20,  -8,  20,
  906.      13,  28,  -3,  78,  38,  16,  11,  62,  46,  29,  21,  24,  46,  65,  43, -23,
  907.      89,  18,  74,  21,  38, -12,  19,  12, -19,   8,  15,  33,   4,  57,   9,  -8,
  908.      36,  35,  26,  28,   7,  83,  63,  79,  75,  11,   3,  87,  37,  47,  34,  40,
  909.      39,  19,  20,  42,  27,  34,  39,  77,  13,  42,  59,  64,  45,  -1,  32,  37,
  910.      45,  -5,  53,  -6,   7,  36,  50,  23,   6,  32,   9, -21,  18,  71,  27,  52,
  911.     -25,  31,  35,  42,  -1,  68,  63,  52,  26,  43,  66,  37,  41,  25,  40,  70,
  912. };
  913.  
  914. /** Noise generation functions.
  915.  *  I'm not sure what these are for - they seem to be some kind of pseudorandom
  916.  *  sequence generators, used to generate noise data which is used when the
  917.  *  channels are rematrixed. I'm not sure if they provide a practical benefit
  918.  *  to compression, or just obfuscate the decoder. Are they for some kind of
  919.  *  dithering? */
  920.  
  921. /** Generate two channels of noise, used in the matrix when
  922.  *  restart sync word == 0x31ea. */
  923.  
  924. static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
  925. {
  926.     SubStream *s = &m->substream[substr];
  927.     unsigned int i;
  928.     uint32_t seed = s->noisegen_seed;
  929.     unsigned int maxchan = s->max_matrix_channel;
  930.  
  931.     for (i = 0; i < s->blockpos; i++) {
  932.         uint16_t seed_shr7 = seed >> 7;
  933.         m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
  934.         m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7)   << s->noise_shift;
  935.  
  936.         seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
  937.     }
  938.  
  939.     s->noisegen_seed = seed;
  940. }
  941.  
  942. /** Generate a block of noise, used when restart sync word == 0x31eb. */
  943.  
  944. static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
  945. {
  946.     SubStream *s = &m->substream[substr];
  947.     unsigned int i;
  948.     uint32_t seed = s->noisegen_seed;
  949.  
  950.     for (i = 0; i < m->access_unit_size_pow2; i++) {
  951.         uint8_t seed_shr15 = seed >> 15;
  952.         m->noise_buffer[i] = noise_table[seed_shr15];
  953.         seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
  954.     }
  955.  
  956.     s->noisegen_seed = seed;
  957. }
  958.  
  959.  
  960. /** Apply the channel matrices in turn to reconstruct the original audio
  961.  *  samples. */
  962.  
  963. static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
  964. {
  965.     SubStream *s = &m->substream[substr];
  966.     unsigned int mat, src_ch, i;
  967.     unsigned int maxchan;
  968.  
  969.     maxchan = s->max_matrix_channel;
  970.     if (!s->noise_type) {
  971.         generate_2_noise_channels(m, substr);
  972.         maxchan += 2;
  973.     } else {
  974.         fill_noise_buffer(m, substr);
  975.     }
  976.  
  977.     for (mat = 0; mat < s->num_primitive_matrices; mat++) {
  978.         int matrix_noise_shift = s->matrix_noise_shift[mat];
  979.         unsigned int dest_ch = s->matrix_out_ch[mat];
  980.         int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
  981.         int32_t *coeffs = s->matrix_coeff[mat];
  982.         int index  = s->num_primitive_matrices - mat;
  983.         int index2 = 2 * index + 1;
  984.  
  985.         /* TODO: DSPContext? */
  986.  
  987.         for (i = 0; i < s->blockpos; i++) {
  988.             int32_t bypassed_lsb = m->bypassed_lsbs[i][mat];
  989.             int32_t *samples = m->sample_buffer[i];
  990.             int64_t accum = 0;
  991.  
  992.             for (src_ch = 0; src_ch <= maxchan; src_ch++)
  993.                 accum += (int64_t) samples[src_ch] * coeffs[src_ch];
  994.  
  995.             if (matrix_noise_shift) {
  996.                 index &= m->access_unit_size_pow2 - 1;
  997.                 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
  998.                 index += index2;
  999.             }
  1000.  
  1001.             samples[dest_ch] = ((accum >> 14) & mask) + bypassed_lsb;
  1002.         }
  1003.     }
  1004. }
  1005.  
  1006. /** Write the audio data into the output buffer. */
  1007.  
  1008. static int output_data(MLPDecodeContext *m, unsigned int substr,
  1009.                        AVFrame *frame, int *got_frame_ptr)
  1010. {
  1011.     AVCodecContext *avctx = m->avctx;
  1012.     SubStream *s = &m->substream[substr];
  1013.     unsigned int i, out_ch = 0;
  1014.     int32_t *data_32;
  1015.     int16_t *data_16;
  1016.     int ret;
  1017.     int is32 = (m->avctx->sample_fmt == AV_SAMPLE_FMT_S32);
  1018.  
  1019.     if (m->avctx->channels != s->max_matrix_channel + 1) {
  1020.         av_log(m->avctx, AV_LOG_ERROR, "channel count mismatch\n");
  1021.         return AVERROR_INVALIDDATA;
  1022.     }
  1023.  
  1024.     if (!s->blockpos) {
  1025.         av_log(avctx, AV_LOG_ERROR, "No samples to output.\n");
  1026.         return AVERROR_INVALIDDATA;
  1027.     }
  1028.  
  1029.     /* get output buffer */
  1030.     frame->nb_samples = s->blockpos;
  1031.     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
  1032.         return ret;
  1033.     data_32 = (int32_t *)frame->data[0];
  1034.     data_16 = (int16_t *)frame->data[0];
  1035.  
  1036.     for (i = 0; i < s->blockpos; i++) {
  1037.         for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
  1038.             int mat_ch = s->ch_assign[out_ch];
  1039.             int32_t sample = m->sample_buffer[i][mat_ch]
  1040.                           << s->output_shift[mat_ch];
  1041.             s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
  1042.             if (is32) *data_32++ = sample << 8;
  1043.             else      *data_16++ = sample >> 8;
  1044.         }
  1045.     }
  1046.  
  1047.     *got_frame_ptr = 1;
  1048.  
  1049.     return 0;
  1050. }
  1051.  
  1052. /** Read an access unit from the stream.
  1053.  *  @return negative on error, 0 if not enough data is present in the input stream,
  1054.  *  otherwise the number of bytes consumed. */
  1055.  
  1056. static int read_access_unit(AVCodecContext *avctx, void* data,
  1057.                             int *got_frame_ptr, AVPacket *avpkt)
  1058. {
  1059.     const uint8_t *buf = avpkt->data;
  1060.     int buf_size = avpkt->size;
  1061.     MLPDecodeContext *m = avctx->priv_data;
  1062.     GetBitContext gb;
  1063.     unsigned int length, substr;
  1064.     unsigned int substream_start;
  1065.     unsigned int header_size = 4;
  1066.     unsigned int substr_header_size = 0;
  1067.     uint8_t substream_parity_present[MAX_SUBSTREAMS];
  1068.     uint16_t substream_data_len[MAX_SUBSTREAMS];
  1069.     uint8_t parity_bits;
  1070.     int ret;
  1071.  
  1072.     if (buf_size < 4)
  1073.         return AVERROR_INVALIDDATA;
  1074.  
  1075.     length = (AV_RB16(buf) & 0xfff) * 2;
  1076.  
  1077.     if (length < 4 || length > buf_size)
  1078.         return AVERROR_INVALIDDATA;
  1079.  
  1080.     init_get_bits(&gb, (buf + 4), (length - 4) * 8);
  1081.  
  1082.     m->is_major_sync_unit = 0;
  1083.     if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
  1084.         if (read_major_sync(m, &gb) < 0)
  1085.             goto error;
  1086.         m->is_major_sync_unit = 1;
  1087.         header_size += 28;
  1088.     }
  1089.  
  1090.     if (!m->params_valid) {
  1091.         av_log(m->avctx, AV_LOG_WARNING,
  1092.                "Stream parameters not seen; skipping frame.\n");
  1093.         *got_frame_ptr = 0;
  1094.         return length;
  1095.     }
  1096.  
  1097.     substream_start = 0;
  1098.  
  1099.     for (substr = 0; substr < m->num_substreams; substr++) {
  1100.         int extraword_present, checkdata_present, end, nonrestart_substr;
  1101.  
  1102.         extraword_present = get_bits1(&gb);
  1103.         nonrestart_substr = get_bits1(&gb);
  1104.         checkdata_present = get_bits1(&gb);
  1105.         skip_bits1(&gb);
  1106.  
  1107.         end = get_bits(&gb, 12) * 2;
  1108.  
  1109.         substr_header_size += 2;
  1110.  
  1111.         if (extraword_present) {
  1112.             if (m->avctx->codec_id == AV_CODEC_ID_MLP) {
  1113.                 av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n");
  1114.                 goto error;
  1115.             }
  1116.             skip_bits(&gb, 16);
  1117.             substr_header_size += 2;
  1118.         }
  1119.  
  1120.         if (!(nonrestart_substr ^ m->is_major_sync_unit)) {
  1121.             av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n");
  1122.             goto error;
  1123.         }
  1124.  
  1125.         if (end + header_size + substr_header_size > length) {
  1126.             av_log(m->avctx, AV_LOG_ERROR,
  1127.                    "Indicated length of substream %d data goes off end of "
  1128.                    "packet.\n", substr);
  1129.             end = length - header_size - substr_header_size;
  1130.         }
  1131.  
  1132.         if (end < substream_start) {
  1133.             av_log(avctx, AV_LOG_ERROR,
  1134.                    "Indicated end offset of substream %d data "
  1135.                    "is smaller than calculated start offset.\n",
  1136.                    substr);
  1137.             goto error;
  1138.         }
  1139.  
  1140.         if (substr > m->max_decoded_substream)
  1141.             continue;
  1142.  
  1143.         substream_parity_present[substr] = checkdata_present;
  1144.         substream_data_len[substr] = end - substream_start;
  1145.         substream_start = end;
  1146.     }
  1147.  
  1148.     parity_bits  = ff_mlp_calculate_parity(buf, 4);
  1149.     parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
  1150.  
  1151.     if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
  1152.         av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
  1153.         goto error;
  1154.     }
  1155.  
  1156.     buf += header_size + substr_header_size;
  1157.  
  1158.     for (substr = 0; substr <= m->max_decoded_substream; substr++) {
  1159.         SubStream *s = &m->substream[substr];
  1160.         init_get_bits(&gb, buf, substream_data_len[substr] * 8);
  1161.  
  1162.         m->matrix_changed = 0;
  1163.         memset(m->filter_changed, 0, sizeof(m->filter_changed));
  1164.  
  1165.         s->blockpos = 0;
  1166.         do {
  1167.             if (get_bits1(&gb)) {
  1168.                 if (get_bits1(&gb)) {
  1169.                     /* A restart header should be present. */
  1170.                     if (read_restart_header(m, &gb, buf, substr) < 0)
  1171.                         goto next_substr;
  1172.                     s->restart_seen = 1;
  1173.                 }
  1174.  
  1175.                 if (!s->restart_seen)
  1176.                     goto next_substr;
  1177.                 if (read_decoding_params(m, &gb, substr) < 0)
  1178.                     goto next_substr;
  1179.             }
  1180.  
  1181.             if (!s->restart_seen)
  1182.                 goto next_substr;
  1183.  
  1184.             if ((ret = read_block_data(m, &gb, substr)) < 0)
  1185.                 return ret;
  1186.  
  1187.             if (get_bits_count(&gb) >= substream_data_len[substr] * 8)
  1188.                 goto substream_length_mismatch;
  1189.  
  1190.         } while (!get_bits1(&gb));
  1191.  
  1192.         skip_bits(&gb, (-get_bits_count(&gb)) & 15);
  1193.  
  1194.         if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) {
  1195.             int shorten_by;
  1196.  
  1197.             if (get_bits(&gb, 16) != 0xD234)
  1198.                 return AVERROR_INVALIDDATA;
  1199.  
  1200.             shorten_by = get_bits(&gb, 16);
  1201.             if      (m->avctx->codec_id == AV_CODEC_ID_TRUEHD && shorten_by  & 0x2000)
  1202.                 s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
  1203.             else if (m->avctx->codec_id == AV_CODEC_ID_MLP    && shorten_by != 0xD234)
  1204.                 return AVERROR_INVALIDDATA;
  1205.  
  1206.             if (substr == m->max_decoded_substream)
  1207.                 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
  1208.         }
  1209.  
  1210.         if (substream_parity_present[substr]) {
  1211.             uint8_t parity, checksum;
  1212.  
  1213.             if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16)
  1214.                 goto substream_length_mismatch;
  1215.  
  1216.             parity   = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
  1217.             checksum = ff_mlp_checksum8       (buf, substream_data_len[substr] - 2);
  1218.  
  1219.             if ((get_bits(&gb, 8) ^ parity) != 0xa9    )
  1220.                 av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr);
  1221.             if ( get_bits(&gb, 8)           != checksum)
  1222.                 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n"    , substr);
  1223.         }
  1224.  
  1225.         if (substream_data_len[substr] * 8 != get_bits_count(&gb))
  1226.             goto substream_length_mismatch;
  1227.  
  1228. next_substr:
  1229.         if (!s->restart_seen)
  1230.             av_log(m->avctx, AV_LOG_ERROR,
  1231.                    "No restart header present in substream %d.\n", substr);
  1232.  
  1233.         buf += substream_data_len[substr];
  1234.     }
  1235.  
  1236.     rematrix_channels(m, m->max_decoded_substream);
  1237.  
  1238.     if ((ret = output_data(m, m->max_decoded_substream, data, got_frame_ptr)) < 0)
  1239.         return ret;
  1240.  
  1241.     return length;
  1242.  
  1243. substream_length_mismatch:
  1244.     av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);
  1245.     return AVERROR_INVALIDDATA;
  1246.  
  1247. error:
  1248.     m->params_valid = 0;
  1249.     return AVERROR_INVALIDDATA;
  1250. }
  1251.  
  1252. #if CONFIG_MLP_DECODER
  1253. AVCodec ff_mlp_decoder = {
  1254.     .name           = "mlp",
  1255.     .long_name      = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
  1256.     .type           = AVMEDIA_TYPE_AUDIO,
  1257.     .id             = AV_CODEC_ID_MLP,
  1258.     .priv_data_size = sizeof(MLPDecodeContext),
  1259.     .init           = mlp_decode_init,
  1260.     .decode         = read_access_unit,
  1261.     .capabilities   = CODEC_CAP_DR1,
  1262. };
  1263. #endif
  1264. #if CONFIG_TRUEHD_DECODER
  1265. AVCodec ff_truehd_decoder = {
  1266.     .name           = "truehd",
  1267.     .long_name      = NULL_IF_CONFIG_SMALL("TrueHD"),
  1268.     .type           = AVMEDIA_TYPE_AUDIO,
  1269.     .id             = AV_CODEC_ID_TRUEHD,
  1270.     .priv_data_size = sizeof(MLPDecodeContext),
  1271.     .init           = mlp_decode_init,
  1272.     .decode         = read_access_unit,
  1273.     .capabilities   = CODEC_CAP_DR1,
  1274. };
  1275. #endif /* CONFIG_TRUEHD_DECODER */
  1276.