annotate mlpdec.c @ 9379:d31c367da415 libavcodec

Make sure mpeg2 has its height rounded up to 32 as that is needed for interlaced stuff. This might have been exploitable when emu edge was not set though note this bug has been introduced just a few days ago.

/*
 * 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 libavcodec/mlpdec.c
 * MLP decoder
 */

#include <stdint.h>

#include "avcodec.h"
#include "libavutil/intreadwrite.h"
#include "bitstream.h"
#include "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];

    //! 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+2];
    //! Left shift to apply to noise values in 0x31eb substreams.
    uint8_t     matrix_noise_shift[MAX_MATRICES];
    //@}

    //! Left shift to apply to Huffman-decoded residuals.
    uint8_t     quant_step_size[MAX_CHANNELS];

    //! number of PCM samples in current audio block
    uint16_t    blocksize;
    //! Number of PCM samples decoded so far in this frame.
    uint16_t    blockpos;

    //! Left shift to apply to decoded PCM values to get final 24-bit output.
    int8_t      output_shift[MAX_CHANNELS];

    //! Running XOR of all output samples.
    int32_t     lossless_check_data;

} SubStream;

typedef struct MLPDecodeContext {
    AVCodecContext *avctx;

    //! Current access unit being read has a major sync.
    int         is_major_sync_unit;

    //! Set if a valid major sync block has been read. Otherwise no decoding is possible.
    uint8_t     params_valid;

    //! Number of substreams contained within this stream.
    uint8_t     num_substreams;

    //! Index of the last substream to decode - further substreams are skipped.
    uint8_t     max_decoded_substream;

    //! 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];

    ChannelParams channel_params[MAX_CHANNELS];

    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+2];
} MLPDecodeContext;

static VLC huff_vlc[3];

/** Initialize static data, constant between all invocations of the codec. */

static av_cold void init_static(void)
{
    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)
{
    ChannelParams *cp = &m->channel_params[ch];
    SubStream *s = &m->substream[substr];
    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 = &m->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 -1;

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

        result  += cp->sign_huff_offset;
        result <<= quant_step_size;

        m->sample_buffer[pos + s->blockpos][channel] = result;
    }

    return 0;
}

static av_cold int mlp_decode_init(AVCodecContext *avctx)
{
    MLPDecodeContext *m = avctx->priv_data;
    int substr;

    init_static();
    m->avctx = avctx;
    for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
        m->substream[substr].lossless_check_data = 0xffffffff;

    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;

    if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)
        return -1;

    if (mh.group1_bits == 0) {
        av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
        return -1;
    }
    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 -1;
    }

    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 -1;
    }

    if (mh.group1_samplerate == 0) {
        av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
        return -1;
    }
    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 -1;
    }
    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 -1;
    }
    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 -1;
    }

    if (mh.num_substreams == 0)
        return -1;
    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 -1;
    }
    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 -1;
    }

    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 = SAMPLE_FMT_S32;
    else
        m->avctx->sample_fmt = SAMPLE_FMT_S16;

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

    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);

    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 -1;
    }
    s->noise_type = get_bits1(gbp);

    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->min_channel > s->max_channel) {
        av_log(m->avctx, AV_LOG_ERROR,
               "Substream min channel cannot be greater than max channel.\n");
        return -1;
    }

    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_INFO,
               "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 -1;
        }
        s->ch_assign[ch_assign] = ch;
    }

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

    if (checksum != get_bits(gbp, 8))
        av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");

    /* Set default decoding parameters. */
    s->param_presence_flags   = 0xff;
    s->num_primitive_matrices = 0;
    s->blocksize              = 8;
    s->lossless_check_data    = 0;

    memset(s->output_shift   , 0, sizeof(s->output_shift   ));
    memset(s->quant_step_size, 0, sizeof(s->quant_step_size));

    for (ch = s->min_channel; ch <= s->max_channel; ch++) {
        ChannelParams *cp = &m->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 channel, unsigned int filter)
{
    FilterParams *fp = &m->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);

    m->filter_changed[channel][filter]++;

    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 -1;
    }
    fp->order = order;

    if (order > 0) {
        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 -1;
        }
        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 -1;
        }

        for (i = 0; i < order; i++)
            fp->coeff[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 -1;
            }

            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, SubStream *s, GetBitContext *gbp)
{
    unsigned int mat, ch;

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

    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 -1;
        }
        if (frac_bits > 14) {
            av_log(m->avctx, AV_LOG_ERROR,
                    "Too many fractional bits specified.\n");
            return -1;
        }

        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)
{
    ChannelParams *cp = &m->channel_params[ch];
    FilterParams *fir = &cp->filter_params[FIR];
    FilterParams *iir = &cp->filter_params[IIR];
    SubStream *s = &m->substream[substr];

    if (s->param_presence_flags & PARAM_FIR)
        if (get_bits1(gbp))
            if (read_filter_params(m, gbp, ch, FIR) < 0)
                return -1;

    if (s->param_presence_flags & PARAM_IIR)
        if (get_bits1(gbp))
            if (read_filter_params(m, gbp, ch, IIR) < 0)
                return -1;

    if (fir->order + iir->order > 8) {
        av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
        return -1;
    }

    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 -1;
    }
    /* 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 -1;
    }

    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;

    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 -1;
            }
        }

    if (s->param_presence_flags & PARAM_MATRIX)
        if (get_bits1(gbp)) {
            if (read_matrix_params(m, s, gbp) < 0)
                return -1;
        }

    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 = &m->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 (read_channel_params(m, substr, gbp, ch) < 0)
                return -1;
        }

    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];
    int32_t firbuf[MAX_BLOCKSIZE + MAX_FIR_ORDER];
    int32_t iirbuf[MAX_BLOCKSIZE + MAX_IIR_ORDER];
    FilterParams *fir = &m->channel_params[channel].filter_params[FIR];
    FilterParams *iir = &m->channel_params[channel].filter_params[IIR];
    unsigned int filter_shift = fir->shift;
    int32_t mask = MSB_MASK(s->quant_step_size[channel]);
    int index = MAX_BLOCKSIZE;
    int i;

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

    for (i = 0; i < s->blocksize; i++) {
        int32_t residual = m->sample_buffer[i + s->blockpos][channel];
        unsigned int order;
        int64_t accum = 0;
        int32_t result;

        /* TODO: Move this code to DSPContext? */

        for (order = 0; order < fir->order; order++)
            accum += (int64_t) firbuf[index + order] * fir->coeff[order];
        for (order = 0; order < iir->order; order++)
            accum += (int64_t) iirbuf[index + order] * iir->coeff[order];

        accum  = accum >> filter_shift;
        result = (accum + residual) & mask;

        --index;

        firbuf[index] = result;
        iirbuf[index] = result - accum;

        m->sample_buffer[i + s->blockpos][channel] = result;
    }

    memcpy(fir->state, &firbuf[index], MAX_FIR_ORDER * sizeof(int32_t));
    memcpy(iir->state, &iirbuf[index], 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;

    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 -1;
    }

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

    for (i = 0; i < s->blocksize; i++) {
        if (read_huff_channels(m, gbp, substr, i) < 0)
            return -1;
    }

    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]);

        /* TODO: DSPContext? */

        for (i = 0; i < s->blockpos; i++) {
            int64_t accum = 0;
            for (src_ch = 0; src_ch <= maxchan; src_ch++) {
                accum += (int64_t)m->sample_buffer[i][src_ch]
                                  * s->matrix_coeff[mat][src_ch];
            }
            if (matrix_noise_shift) {
                uint32_t index = s->num_primitive_matrices - mat;
                index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1);
                accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
            }
            m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask)
                                             + m->bypassed_lsbs[i][mat];
        }
    }
}

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

static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
                                uint8_t *data, unsigned int *data_size, int is32)
{
    SubStream *s = &m->substream[substr];
    unsigned int i, out_ch = 0;
    int32_t *data_32 = (int32_t*) data;
    int16_t *data_16 = (int16_t*) data;

    if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
        return -1;

    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;
        }
    }

    *data_size = i * out_ch * (is32 ? 4 : 2);

    return 0;
}

static int output_data(MLPDecodeContext *m, unsigned int substr,
                       uint8_t *data, unsigned int *data_size)
{
    if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
        return output_data_internal(m, substr, data, data_size, 1);
    else
        return output_data_internal(m, substr, data, data_size, 0);
}


/** Read an access unit from the stream.
 *  Returns < 0 on error, 0 if not enough data is present in the input stream
 *  otherwise returns the number of bytes consumed. */

static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
                            AVPacket *avpkt)
{
    const uint8_t *buf = avpkt->data;
    int buf_size = avpkt->size;
    MLPDecodeContext *m = avctx->priv_data;
    GetBitContext gb;
    unsigned int length, substr;
    unsigned int substream_start;
    unsigned int header_size = 4;
    unsigned int substr_header_size = 0;
    uint8_t substream_parity_present[MAX_SUBSTREAMS];
    uint16_t substream_data_len[MAX_SUBSTREAMS];
    uint8_t parity_bits;

    if (buf_size < 4)
        return 0;

    length = (AV_RB16(buf) & 0xfff) * 2;

    if (length > buf_size)
        return -1;

    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");
        *data_size = 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 {
            unsigned int ch;

            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 (m->matrix_changed > 1) {
                av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n");
                goto next_substr;
            }
            for (ch = 0; ch < s->max_channel; ch++)
                if (m->filter_changed[ch][FIR] > 1 ||
                    m->filter_changed[ch][IIR] > 1) {
                    av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n");
                    goto next_substr;
                }

            if (!s->restart_seen) {
                goto next_substr;
            }

            if (read_block_data(m, &gb, substr) < 0)
                return -1;

            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 -1;

            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 -1;

            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 (output_data(m, m->max_decoded_substream, data, data_size) < 0)
        return -1;

    return length;

substream_length_mismatch:
    av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);
    return -1;

error:
    m->params_valid = 0;
    return -1;
}

#if CONFIG_MLP_DECODER
AVCodec mlp_decoder = {
    "mlp",
    CODEC_TYPE_AUDIO,
    CODEC_ID_MLP,
    sizeof(MLPDecodeContext),
    mlp_decode_init,
    NULL,
    NULL,
    read_access_unit,
    .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
};
#endif /* CONFIG_MLP_DECODER */

#if CONFIG_TRUEHD_DECODER
AVCodec truehd_decoder = {
    "truehd",
    CODEC_TYPE_AUDIO,
    CODEC_ID_TRUEHD,
    sizeof(MLPDecodeContext),
    mlp_decode_init,
    NULL,
    NULL,
    read_access_unit,
    .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
};
#endif /* CONFIG_TRUEHD_DECODER */