PeerStreamer

/*

 * MLP decoder

 * Copyright (c) 2007-2008 Ian Caulfield

 *

 * This file is part of FFmpeg.

 *

 * FFmpeg is free software; you can redistribute it and/or

 * modify it under the terms of the GNU Lesser General Public

 * License as published by the Free Software Foundation; either

 * version 2.1 of the License, or (at your option) any later version.

 *

 * FFmpeg is distributed in the hope that it will be useful,

 * but WITHOUT ANY WARRANTY; without even the implied warranty of

 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU

 * Lesser General Public License for more details.

 *

 * You should have received a copy of the GNU Lesser General Public

 * License along with FFmpeg; if not, write to the Free Software

 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

 */

/**

 * @file

 * MLP decoder

 */

#include <stdint.h>

#include "avcodec.h"

#include "dsputil.h"

#include "libavutil/intreadwrite.h"

#include "get_bits.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];

    //! Channel coding parameters for channels in the substream

    ChannelParams channel_params[MAX_CHANNELS];

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

    uint8_t     noise_shift;

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

    uint32_t    noisegen_seed;

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

    uint8_t     data_check_present;

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

    uint8_t     param_presence_flags;

#define PARAM_BLOCKSIZE     (1 << 7)

#define PARAM_MATRIX        (1 << 6)

#define PARAM_OUTSHIFT      (1 << 5)

#define PARAM_QUANTSTEP     (1 << 4)

#define PARAM_FIR           (1 << 3)

#define PARAM_IIR           (1 << 2)

#define PARAM_HUFFOFFSET    (1 << 1)

#define PARAM_PRESENCE      (1 << 0)

    //@}

    //@{

    /** matrix data */

    //! Number of matrices to be applied.

    uint8_t     num_primitive_matrices;

    //! matrix output channel

    uint8_t     matrix_out_ch[MAX_MATRICES];

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

    uint8_t     lsb_bypass[MAX_MATRICES];

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

    int32_t     matrix_coeff[MAX_MATRICES][MAX_CHANNELS];

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

    uint8_t     matrix_noise_shift[MAX_MATRICES];

    //@}

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

    uint8_t     quant_step_size[MAX_CHANNELS];

    //! number of PCM samples in current audio block

    uint16_t    blocksize;

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

    uint16_t    blockpos;

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

    int8_t      output_shift[MAX_CHANNELS];

    //! Running XOR of all output samples.

    int32_t     lossless_check_data;

} SubStream;

typedef struct MLPDecodeContext {

    AVCodecContext *avctx;

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

    int         is_major_sync_unit;

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

    uint8_t     params_valid;

    //! Number of substreams contained within this stream.

    uint8_t     num_substreams;

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

    uint8_t     max_decoded_substream;

    //! number of PCM samples contained in each frame

    int         access_unit_size;

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

    int         access_unit_size_pow2;

    SubStream   substream[MAX_SUBSTREAMS];

    int         matrix_changed;

    int         filter_changed[MAX_CHANNELS][NUM_FILTERS];

    int8_t      noise_buffer[MAX_BLOCKSIZE_POW2];

    int8_t      bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];

    int32_t     sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS];

    DSPContext  dsp;

} MLPDecodeContext;

static VLC huff_vlc[3];

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

static av_cold void init_static(void)

{

    if (!huff_vlc[0].bits) {

        INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,

                    &ff_mlp_huffman_tables[0][0][1], 2, 1,

                    &ff_mlp_huffman_tables[0][0][0], 2, 1, 512);

        INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,

                    &ff_mlp_huffman_tables[1][0][1], 2, 1,

                    &ff_mlp_huffman_tables[1][0][0], 2, 1, 512);

        INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,

                    &ff_mlp_huffman_tables[2][0][1], 2, 1,

                    &ff_mlp_huffman_tables[2][0][0], 2, 1, 512);

    }

    ff_mlp_init_crc();

}

static inline int32_t calculate_sign_huff(MLPDecodeContext *m,

                                          unsigned int substr, unsigned int ch)

{

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

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

    int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];

    int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);

    int32_t sign_huff_offset = cp->huff_offset;

    if (cp->codebook > 0)

        sign_huff_offset -= 7 << lsb_bits;

    if (sign_shift >= 0)

        sign_huff_offset -= 1 << sign_shift;

    return sign_huff_offset;

}

/** Read a sample, consisting of either, both or neither of entropy-coded MSBs

 *  and plain LSBs. */

static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,

                                     unsigned int substr, unsigned int pos)

{

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

    unsigned int mat, channel;

    for (mat = 0; mat < s->num_primitive_matrices; mat++)

        if (s->lsb_bypass[mat])

            m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);

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

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

        int codebook = cp->codebook;

        int quant_step_size = s->quant_step_size[channel];

        int lsb_bits = cp->huff_lsbs - quant_step_size;

        int result = 0;

        if (codebook > 0)

            result = get_vlc2(gbp, huff_vlc[codebook-1].table,

                            VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);

        if (result < 0)

            return -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;

    dsputil_init(&m->dsp, avctx);

    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 = AV_SAMPLE_FMT_S32;

    else

        m->avctx->sample_fmt = AV_SAMPLE_FMT_S16;

    m->params_valid = 1;

    for (substr = 0; substr < MAX_SUBSTREAMS; substr++)

        m->substream[substr].restart_seen = 0;

    return 0;

}

/** Read a restart header from a block in a substream. This contains parameters

 *  required to decode the audio that do not change very often. Generally

 *  (always) present only in blocks following a major sync. */

static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,

                               const uint8_t *buf, unsigned int substr)

{

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

    unsigned int ch;

    int sync_word, tmp;

    uint8_t checksum;

    uint8_t lossless_check;

    int start_count = get_bits_count(gbp);

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

                                 ? MAX_MATRIX_CHANNEL_MLP

                                 : MAX_MATRIX_CHANNEL_TRUEHD;

    sync_word = get_bits(gbp, 13);

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

        av_log(m->avctx, AV_LOG_ERROR,

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

        return -1;

    }

    s->noise_type = get_bits1(gbp);

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

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

        return -1;

    }

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

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

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

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

    if (s->max_matrix_channel > max_matrix_channel) {

        av_log(m->avctx, AV_LOG_ERROR,

               "Max matrix channel cannot be greater than %d.\n",

               max_matrix_channel);

        return -1;

    }

    if (s->max_channel != s->max_matrix_channel) {

        av_log(m->avctx, AV_LOG_ERROR,

               "Max channel must be equal max matrix channel.\n");

        return -1;

    }

    /* This should happen for TrueHD streams with >6 channels and MLP's noise

     * type. It is not yet known if this is allowed. */

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

        av_log(m->avctx, AV_LOG_ERROR,

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

               "by the decoder. %s\n", s->max_channel+2, sample_message);

        return -1;

    }

    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_DEBUG,

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

               "Further substreams will be skipped.\n",

               s->max_channel + 1, substr);

        m->max_decoded_substream = substr;

    }

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

    s->noisegen_seed = get_bits(gbp, 23);

    skip_bits(gbp, 19);

    s->data_check_present = get_bits1(gbp);

    lossless_check = get_bits(gbp, 8);

    if (substr == m->max_decoded_substream

        && s->lossless_check_data != 0xffffffff) {

        tmp = xor_32_to_8(s->lossless_check_data);

        if (tmp != lossless_check)

            av_log(m->avctx, AV_LOG_WARNING,

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

                   lossless_check, tmp);

    }

    skip_bits(gbp, 16);

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

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

        int ch_assign = get_bits(gbp, 6);

        if (ch_assign > s->max_matrix_channel) {

            av_log(m->avctx, AV_LOG_ERROR,

                   "Assignment of matrix channel %d to invalid output channel %d. %s\n",

                   ch, ch_assign, sample_message);

            return -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 = &s->channel_params[ch];

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

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

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

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

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

        cp->huff_offset      = 0;

        cp->sign_huff_offset = (-1) << 23;

        cp->codebook         = 0;

        cp->huff_lsbs        = 24;

    }

    if (substr == m->max_decoded_substream)

        m->avctx->channels = s->max_matrix_channel + 1;

    return 0;

}

/** Read parameters for one of the prediction filters. */

static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,

                              unsigned int substr, unsigned int channel,

                              unsigned int filter)

{

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

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

    const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;

    const char fchar = filter ? 'I' : 'F';

    int i, order;

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

    assert(filter < 2);

    if (m->filter_changed[channel][filter]++ > 1) {

        av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n");

        return -1;

    }

    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) {

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

        int coeff_bits, coeff_shift;

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

        coeff_bits  = get_bits(gbp, 5);

        coeff_shift = get_bits(gbp, 3);

        if (coeff_bits < 1 || coeff_bits > 16) {

            av_log(m->avctx, AV_LOG_ERROR,

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

                   fchar);

            return -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++)

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

        if (get_bits1(gbp)) {

            int state_bits, state_shift;

            if (filter == FIR) {

                av_log(m->avctx, AV_LOG_ERROR,

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

                return -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, unsigned int substr, GetBitContext *gbp)

{

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

    unsigned int mat, ch;

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

                                     ? MAX_MATRICES_MLP

                                     : MAX_MATRICES_TRUEHD;

    if (m->matrix_changed++ > 1) {

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

        return -1;

    }

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

    if (s->num_primitive_matrices > max_primitive_matrices) {

        av_log(m->avctx, AV_LOG_ERROR,

               "Number of primitive matrices cannot be greater than %d.\n",

               max_primitive_matrices);

        return -1;

    }

    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)

{

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

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

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

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

    if (s->param_presence_flags & PARAM_FIR)

        if (get_bits1(gbp))

            if (read_filter_params(m, gbp, substr, ch, FIR) < 0)

                return -1;

    if (s->param_presence_flags & PARAM_IIR)

        if (get_bits1(gbp))

            if (read_filter_params(m, gbp, substr, 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, substr, 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 = &s->channel_params[ch];

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

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

            }

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

        if (get_bits1(gbp))

            if (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];

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

    int32_t state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FIR_ORDER];

    int32_t *firbuf = state_buffer[FIR] + MAX_BLOCKSIZE;

    int32_t *iirbuf = state_buffer[IIR] + MAX_BLOCKSIZE;

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

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

    unsigned int filter_shift = fir->shift;

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

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

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

    m->dsp.mlp_filter_channel(firbuf, fircoeff,

                              fir->order, iir->order,

                              filter_shift, mask, s->blocksize,

                              &m->sample_buffer[s->blockpos][channel]);

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

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

}

/** Read a block of PCM residual data (or actual if no filtering active). */

static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,

                           unsigned int substr)

{

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

    unsigned int i, ch, expected_stream_pos = 0;

    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]);

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

        int index  = s->num_primitive_matrices - mat;

        int index2 = 2 * index + 1;

        /* TODO: DSPContext? */

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

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

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

            int64_t accum = 0;

            for (src_ch = 0; src_ch <= maxchan; src_ch++)

                accum += (int64_t) samples[src_ch] * coeffs[src_ch];

            if (matrix_noise_shift) {

                index &= m->access_unit_size_pow2 - 1;

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

                index += index2;

            }

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

        }

    }

}

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

static int output_data_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 == AV_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.

 *  @return negative on error, 0 if not enough data is present in the input stream,

 *  otherwise the number of bytes consumed. */

static int read_access_unit(AVCodecContext *avctx, void* data, int *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 < 4 || 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 {

            if (get_bits1(&gb)) {

                if (get_bits1(&gb)) {

                    /* A restart header should be present. */

                    if (read_restart_header(m, &gb, buf, substr) < 0)

                        goto next_substr;

                    s->restart_seen = 1;

                }

                if (!s->restart_seen)

                    goto next_substr;

                if (read_decoding_params(m, &gb, substr) < 0)

                    goto next_substr;

            }

            if (!s->restart_seen)

                goto next_substr;

            if (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;

}

AVCodec ff_mlp_decoder = {

    "mlp",

    AVMEDIA_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)"),

};

#if CONFIG_TRUEHD_DECODER

AVCodec ff_truehd_decoder = {

    "truehd",

    AVMEDIA_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 */