adding ffmpeg specific version

1 files changed, 1272 insertions, 0 deletions

diff --git a/ffmpeg1/libavcodec/mlpdec.c b/ffmpeg1/libavcodec/mlpdec.c
new file mode 100644
index 0000000..a7c79a4
--- /dev/null
+++ b/ffmpeg1/libavcodec/mlpdec.c
@@ -0,0 +1,1272 @@
+/*
+ * MLP decoder
+ * Copyright (c) 2007-2008 Ian Caulfield
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * MLP decoder
+ */
+
+#include <stdint.h>
+
+#include "avcodec.h"
+#include "libavutil/intreadwrite.h"
+#include "libavutil/channel_layout.h"
+#include "get_bits.h"
+#include "internal.h"
+#include "libavutil/crc.h"
+#include "parser.h"
+#include "mlp_parser.h"
+#include "mlpdsp.h"
+#include "mlp.h"
+
+/** number of bits used for VLC lookup - longest Huffman code is 9 */
+#define VLC_BITS            9
+
+typedef struct SubStream {
+    /// Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
+    uint8_t     restart_seen;
+
+    //@{
+    /** restart header data */
+    /// The type of noise to be used in the rematrix stage.
+    uint16_t    noise_type;
+
+    /// The index of the first channel coded in this substream.
+    uint8_t     min_channel;
+    /// The index of the last channel coded in this substream.
+    uint8_t     max_channel;
+    /// The number of channels input into the rematrix stage.
+    uint8_t     max_matrix_channel;
+    /// For each channel output by the matrix, the output channel to map it to
+    uint8_t     ch_assign[MAX_CHANNELS];
+    /// The channel layout for this substream
+    uint64_t    ch_layout;
+
+    /// Channel coding parameters for channels in the substream
+    ChannelParams channel_params[MAX_CHANNELS];
+
+    /// The left shift applied to random noise in 0x31ea substreams.
+    uint8_t     noise_shift;
+    /// The current seed value for the pseudorandom noise generator(s).
+    uint32_t    noisegen_seed;
+
+    /// Set if the substream contains extra info to check the size of VLC blocks.
+    uint8_t     data_check_present;
+
+    /// Bitmask of which parameter sets are conveyed in a decoding parameter block.
+    uint8_t     param_presence_flags;
+#define PARAM_BLOCKSIZE     (1 << 7)
+#define PARAM_MATRIX        (1 << 6)
+#define PARAM_OUTSHIFT      (1 << 5)
+#define PARAM_QUANTSTEP     (1 << 4)
+#define PARAM_FIR           (1 << 3)
+#define PARAM_IIR           (1 << 2)
+#define PARAM_HUFFOFFSET    (1 << 1)
+#define PARAM_PRESENCE      (1 << 0)
+    //@}
+
+    //@{
+    /** matrix data */
+
+    /// Number of matrices to be applied.
+    uint8_t     num_primitive_matrices;
+
+    /// matrix output channel
+    uint8_t     matrix_out_ch[MAX_MATRICES];
+
+    /// Whether the LSBs of the matrix output are encoded in the bitstream.
+    uint8_t     lsb_bypass[MAX_MATRICES];
+    /// Matrix coefficients, stored as 2.14 fixed point.
+    int32_t     matrix_coeff[MAX_MATRICES][MAX_CHANNELS];
+    /// Left shift to apply to noise values in 0x31eb substreams.
+    uint8_t     matrix_noise_shift[MAX_MATRICES];
+    //@}
+
+    /// Left shift to apply to Huffman-decoded residuals.
+    uint8_t     quant_step_size[MAX_CHANNELS];
+
+    /// number of PCM samples in current audio block
+    uint16_t    blocksize;
+    /// Number of PCM samples decoded so far in this frame.
+    uint16_t    blockpos;
+
+    /// Left shift to apply to decoded PCM values to get final 24-bit output.
+    int8_t      output_shift[MAX_CHANNELS];
+
+    /// Running XOR of all output samples.
+    int32_t     lossless_check_data;
+
+} SubStream;
+
+typedef struct MLPDecodeContext {
+    AVCodecContext *avctx;
+
+    /// Current access unit being read has a major sync.
+    int         is_major_sync_unit;
+
+    /// Set if a valid major sync block has been read. Otherwise no decoding is possible.
+    uint8_t     params_valid;
+
+    /// Number of substreams contained within this stream.
+    uint8_t     num_substreams;
+
+    /// Index of the last substream to decode - further substreams are skipped.
+    uint8_t     max_decoded_substream;
+
+    /// Stream needs channel reordering to comply with FFmpeg's channel order
+    uint8_t     needs_reordering;
+
+    /// number of PCM samples contained in each frame
+    int         access_unit_size;
+    /// next power of two above the number of samples in each frame
+    int         access_unit_size_pow2;
+
+    SubStream   substream[MAX_SUBSTREAMS];
+
+    int         matrix_changed;
+    int         filter_changed[MAX_CHANNELS][NUM_FILTERS];
+
+    int8_t      noise_buffer[MAX_BLOCKSIZE_POW2];
+    int8_t      bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
+    int32_t     sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS];
+
+    MLPDSPContext dsp;
+} MLPDecodeContext;
+
+static const uint64_t thd_channel_order[] = {
+    AV_CH_FRONT_LEFT, AV_CH_FRONT_RIGHT,                     // LR
+    AV_CH_FRONT_CENTER,                                      // C
+    AV_CH_LOW_FREQUENCY,                                     // LFE
+    AV_CH_SIDE_LEFT, AV_CH_SIDE_RIGHT,                       // LRs
+    AV_CH_TOP_FRONT_LEFT, AV_CH_TOP_FRONT_RIGHT,             // LRvh
+    AV_CH_FRONT_LEFT_OF_CENTER, AV_CH_FRONT_RIGHT_OF_CENTER, // LRc
+    AV_CH_BACK_LEFT, AV_CH_BACK_RIGHT,                       // LRrs
+    AV_CH_BACK_CENTER,                                       // Cs
+    AV_CH_TOP_CENTER,                                        // Ts
+    AV_CH_SURROUND_DIRECT_LEFT, AV_CH_SURROUND_DIRECT_RIGHT, // LRsd
+    AV_CH_WIDE_LEFT, AV_CH_WIDE_RIGHT,                       // LRw
+    AV_CH_TOP_FRONT_CENTER,                                  // Cvh
+    AV_CH_LOW_FREQUENCY_2,                                   // LFE2
+};
+
+static uint64_t thd_channel_layout_extract_channel(uint64_t channel_layout,
+                                                   int index)
+{
+    int i;
+
+    if (av_get_channel_layout_nb_channels(channel_layout) <= index)
+        return 0;
+
+    for (i = 0; i < FF_ARRAY_ELEMS(thd_channel_order); i++)
+        if (channel_layout & thd_channel_order[i] && !index--)
+            return thd_channel_order[i];
+    return 0;
+}
+
+static VLC huff_vlc[3];
+
+/** Initialize static data, constant between all invocations of the codec. */
+
+static av_cold void init_static(void)
+{
+    if (!huff_vlc[0].bits) {
+        INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
+                    &ff_mlp_huffman_tables[0][0][1], 2, 1,
+                    &ff_mlp_huffman_tables[0][0][0], 2, 1, 512);
+        INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
+                    &ff_mlp_huffman_tables[1][0][1], 2, 1,
+                    &ff_mlp_huffman_tables[1][0][0], 2, 1, 512);
+        INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
+                    &ff_mlp_huffman_tables[2][0][1], 2, 1,
+                    &ff_mlp_huffman_tables[2][0][0], 2, 1, 512);
+    }
+
+    ff_mlp_init_crc();
+}
+
+static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
+                                          unsigned int substr, unsigned int ch)
+{
+    SubStream *s = &m->substream[substr];
+    ChannelParams *cp = &s->channel_params[ch];
+    int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
+    int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
+    int32_t sign_huff_offset = cp->huff_offset;
+
+    if (cp->codebook > 0)
+        sign_huff_offset -= 7 << lsb_bits;
+
+    if (sign_shift >= 0)
+        sign_huff_offset -= 1 << sign_shift;
+
+    return sign_huff_offset;
+}
+
+/** Read a sample, consisting of either, both or neither of entropy-coded MSBs
+ *  and plain LSBs. */
+
+static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
+                                     unsigned int substr, unsigned int pos)
+{
+    SubStream *s = &m->substream[substr];
+    unsigned int mat, channel;
+
+    for (mat = 0; mat < s->num_primitive_matrices; mat++)
+        if (s->lsb_bypass[mat])
+            m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
+
+    for (channel = s->min_channel; channel <= s->max_channel; channel++) {
+        ChannelParams *cp = &s->channel_params[channel];
+        int codebook = cp->codebook;
+        int quant_step_size = s->quant_step_size[channel];
+        int lsb_bits = cp->huff_lsbs - quant_step_size;
+        int result = 0;
+
+        if (codebook > 0)
+            result = get_vlc2(gbp, huff_vlc[codebook-1].table,
+                            VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
+
+        if (result < 0)
+            return AVERROR_INVALIDDATA;
+
+        if (lsb_bits > 0)
+            result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
+
+        result  += cp->sign_huff_offset;
+        result <<= quant_step_size;
+
+        m->sample_buffer[pos + s->blockpos][channel] = result;
+    }
+
+    return 0;
+}
+
+static av_cold int mlp_decode_init(AVCodecContext *avctx)
+{
+    MLPDecodeContext *m = avctx->priv_data;
+    int substr;
+
+    init_static();
+    m->avctx = avctx;
+    for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
+        m->substream[substr].lossless_check_data = 0xffffffff;
+    ff_mlpdsp_init(&m->dsp);
+
+    return 0;
+}
+
+/** Read a major sync info header - contains high level information about
+ *  the stream - sample rate, channel arrangement etc. Most of this
+ *  information is not actually necessary for decoding, only for playback.
+ */
+
+static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
+{
+    MLPHeaderInfo mh;
+    int substr, ret;
+
+    if ((ret = ff_mlp_read_major_sync(m->avctx, &mh, gb)) != 0)
+        return ret;
+
+    if (mh.group1_bits == 0) {
+        av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
+        return AVERROR_INVALIDDATA;
+    }
+    if (mh.group2_bits > mh.group1_bits) {
+        av_log(m->avctx, AV_LOG_ERROR,
+               "Channel group 2 cannot have more bits per sample than group 1.\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
+        av_log(m->avctx, AV_LOG_ERROR,
+               "Channel groups with differing sample rates are not currently supported.\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (mh.group1_samplerate == 0) {
+        av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
+        return AVERROR_INVALIDDATA;
+    }
+    if (mh.group1_samplerate > MAX_SAMPLERATE) {
+        av_log(m->avctx, AV_LOG_ERROR,
+               "Sampling rate %d is greater than the supported maximum (%d).\n",
+               mh.group1_samplerate, MAX_SAMPLERATE);
+        return AVERROR_INVALIDDATA;
+    }
+    if (mh.access_unit_size > MAX_BLOCKSIZE) {
+        av_log(m->avctx, AV_LOG_ERROR,
+               "Block size %d is greater than the supported maximum (%d).\n",
+               mh.access_unit_size, MAX_BLOCKSIZE);
+        return AVERROR_INVALIDDATA;
+    }
+    if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
+        av_log(m->avctx, AV_LOG_ERROR,
+               "Block size pow2 %d is greater than the supported maximum (%d).\n",
+               mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (mh.num_substreams == 0)
+        return AVERROR_INVALIDDATA;
+    if (m->avctx->codec_id == AV_CODEC_ID_MLP && mh.num_substreams > 2) {
+        av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n");
+        return AVERROR_INVALIDDATA;
+    }
+    if (mh.num_substreams > MAX_SUBSTREAMS) {
+        avpriv_request_sample(m->avctx,
+                              "%d substreams (more than the "
+                              "maximum supported by the decoder)",
+                              mh.num_substreams);
+        return AVERROR_PATCHWELCOME;
+    }
+
+    m->access_unit_size      = mh.access_unit_size;
+    m->access_unit_size_pow2 = mh.access_unit_size_pow2;
+
+    m->num_substreams        = mh.num_substreams;
+    m->max_decoded_substream = m->num_substreams - 1;
+
+    m->avctx->sample_rate    = mh.group1_samplerate;
+    m->avctx->frame_size     = mh.access_unit_size;
+
+    m->avctx->bits_per_raw_sample = mh.group1_bits;
+    if (mh.group1_bits > 16)
+        m->avctx->sample_fmt = AV_SAMPLE_FMT_S32;
+    else
+        m->avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+
+    m->params_valid = 1;
+    for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
+        m->substream[substr].restart_seen = 0;
+
+    /* Set the layout for each substream. When there's more than one, the first
+     * substream is Stereo. Subsequent substreams' layouts are indicated in the
+     * major sync. */
+    if (m->avctx->codec_id == AV_CODEC_ID_MLP) {
+        if ((substr = (mh.num_substreams > 1)))
+            m->substream[0].ch_layout = AV_CH_LAYOUT_STEREO;
+        m->substream[substr].ch_layout = mh.channel_layout_mlp;
+    } else {
+        if ((substr = (mh.num_substreams > 1)))
+            m->substream[0].ch_layout = AV_CH_LAYOUT_STEREO;
+        if (mh.num_substreams > 2)
+            if (mh.channel_layout_thd_stream2)
+                m->substream[2].ch_layout = mh.channel_layout_thd_stream2;
+            else
+                m->substream[2].ch_layout = mh.channel_layout_thd_stream1;
+        m->substream[substr].ch_layout = mh.channel_layout_thd_stream1;
+
+        if (m->avctx->channels<=2 && m->substream[substr].ch_layout == AV_CH_LAYOUT_MONO && m->max_decoded_substream == 1) {
+            av_log(m->avctx, AV_LOG_DEBUG, "Mono stream with 2 substreams, ignoring 2nd\n");
+            m->max_decoded_substream = 0;
+            if (m->avctx->channels==2)
+                m->avctx->channel_layout = AV_CH_LAYOUT_STEREO;
+        }
+    }
+
+    m->needs_reordering = mh.channel_arrangement >= 18 && mh.channel_arrangement <= 20;
+
+    return 0;
+}
+
+/** Read a restart header from a block in a substream. This contains parameters
+ *  required to decode the audio that do not change very often. Generally
+ *  (always) present only in blocks following a major sync. */
+
+static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
+                               const uint8_t *buf, unsigned int substr)
+{
+    SubStream *s = &m->substream[substr];
+    unsigned int ch;
+    int sync_word, tmp;
+    uint8_t checksum;
+    uint8_t lossless_check;
+    int start_count = get_bits_count(gbp);
+    const int max_matrix_channel = m->avctx->codec_id == AV_CODEC_ID_MLP
+                                 ? MAX_MATRIX_CHANNEL_MLP
+                                 : MAX_MATRIX_CHANNEL_TRUEHD;
+    int max_channel, min_channel, matrix_channel;
+
+    sync_word = get_bits(gbp, 13);
+
+    if (sync_word != 0x31ea >> 1) {
+        av_log(m->avctx, AV_LOG_ERROR,
+               "restart header sync incorrect (got 0x%04x)\n", sync_word);
+        return AVERROR_INVALIDDATA;
+    }
+
+    s->noise_type = get_bits1(gbp);
+
+    if (m->avctx->codec_id == AV_CODEC_ID_MLP && s->noise_type) {
+        av_log(m->avctx, AV_LOG_ERROR, "MLP must have 0x31ea sync word.\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    skip_bits(gbp, 16); /* Output timestamp */
+
+    min_channel    = get_bits(gbp, 4);
+    max_channel    = get_bits(gbp, 4);
+    matrix_channel = get_bits(gbp, 4);
+
+    if (matrix_channel > max_matrix_channel) {
+        av_log(m->avctx, AV_LOG_ERROR,
+               "Max matrix channel cannot be greater than %d.\n",
+               max_matrix_channel);
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (max_channel != matrix_channel) {
+        av_log(m->avctx, AV_LOG_ERROR,
+               "Max channel must be equal max matrix channel.\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    /* This should happen for TrueHD streams with >6 channels and MLP's noise
+     * type. It is not yet known if this is allowed. */
+    if (max_channel > MAX_MATRIX_CHANNEL_MLP && !s->noise_type) {
+        avpriv_request_sample(m->avctx,
+                              "%d channels (more than the "
+                              "maximum supported by the decoder)",
+                              max_channel + 2);
+        return AVERROR_PATCHWELCOME;
+    }
+
+    if (min_channel > max_channel) {
+        av_log(m->avctx, AV_LOG_ERROR,
+               "Substream min channel cannot be greater than max channel.\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    s->min_channel = min_channel;
+    s->max_channel = max_channel;
+    s->max_matrix_channel = matrix_channel;
+
+#if FF_API_REQUEST_CHANNELS
+    if (m->avctx->request_channels > 0 &&
+        m->avctx->request_channels <= s->max_channel + 1 &&
+        m->max_decoded_substream > substr) {
+        av_log(m->avctx, AV_LOG_DEBUG,
+               "Extracting %d-channel downmix from substream %d. "
+               "Further substreams will be skipped.\n",
+               s->max_channel + 1, substr);
+        m->max_decoded_substream = substr;
+    } else
+#endif
+    if (m->avctx->request_channel_layout == s->ch_layout &&
+        m->max_decoded_substream > substr) {
+        av_log(m->avctx, AV_LOG_DEBUG,
+               "Extracting %d-channel downmix (0x%"PRIx64") from substream %d. "
+               "Further substreams will be skipped.\n",
+               s->max_channel + 1, s->ch_layout, substr);
+        m->max_decoded_substream = substr;
+    }
+
+    s->noise_shift   = get_bits(gbp,  4);
+    s->noisegen_seed = get_bits(gbp, 23);
+
+    skip_bits(gbp, 19);
+
+    s->data_check_present = get_bits1(gbp);
+    lossless_check = get_bits(gbp, 8);
+    if (substr == m->max_decoded_substream
+        && s->lossless_check_data != 0xffffffff) {
+        tmp = xor_32_to_8(s->lossless_check_data);
+        if (tmp != lossless_check)
+            av_log(m->avctx, AV_LOG_WARNING,
+                   "Lossless check failed - expected %02x, calculated %02x.\n",
+                   lossless_check, tmp);
+    }
+
+    skip_bits(gbp, 16);
+
+    memset(s->ch_assign, 0, sizeof(s->ch_assign));
+
+    for (ch = 0; ch <= s->max_matrix_channel; ch++) {
+        int ch_assign = get_bits(gbp, 6);
+        if (m->avctx->codec_id == AV_CODEC_ID_TRUEHD) {
+            uint64_t channel = thd_channel_layout_extract_channel(s->ch_layout,
+                                                                  ch_assign);
+            ch_assign = av_get_channel_layout_channel_index(s->ch_layout,
+                                                            channel);
+        }
+        if ((unsigned)ch_assign > s->max_matrix_channel) {
+            avpriv_request_sample(m->avctx,
+                                  "Assignment of matrix channel %d to invalid output channel %d",
+                                  ch, ch_assign);
+            return AVERROR_PATCHWELCOME;
+        }
+        s->ch_assign[ch_assign] = ch;
+    }
+
+    checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
+
+    if (checksum != get_bits(gbp, 8))
+        av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
+
+    /* Set default decoding parameters. */
+    s->param_presence_flags   = 0xff;
+    s->num_primitive_matrices = 0;
+    s->blocksize              = 8;
+    s->lossless_check_data    = 0;
+
+    memset(s->output_shift   , 0, sizeof(s->output_shift   ));
+    memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
+
+    for (ch = s->min_channel; ch <= s->max_channel; ch++) {
+        ChannelParams *cp = &s->channel_params[ch];
+        cp->filter_params[FIR].order = 0;
+        cp->filter_params[IIR].order = 0;
+        cp->filter_params[FIR].shift = 0;
+        cp->filter_params[IIR].shift = 0;
+
+        /* Default audio coding is 24-bit raw PCM. */
+        cp->huff_offset      = 0;
+        cp->sign_huff_offset = (-1) << 23;
+        cp->codebook         = 0;
+        cp->huff_lsbs        = 24;
+    }
+
+    if (substr == m->max_decoded_substream) {
+        m->avctx->channels       = s->max_matrix_channel + 1;
+        m->avctx->channel_layout = s->ch_layout;
+
+        if (m->avctx->codec_id == AV_CODEC_ID_MLP && m->needs_reordering) {
+            if (m->avctx->channel_layout == (AV_CH_LAYOUT_QUAD|AV_CH_LOW_FREQUENCY) ||
+                m->avctx->channel_layout == AV_CH_LAYOUT_5POINT0_BACK) {
+                int i = s->ch_assign[4];
+                s->ch_assign[4] = s->ch_assign[3];
+                s->ch_assign[3] = s->ch_assign[2];
+                s->ch_assign[2] = i;
+            } else if (m->avctx->channel_layout == AV_CH_LAYOUT_5POINT1_BACK) {
+                FFSWAP(int, s->ch_assign[2], s->ch_assign[4]);
+                FFSWAP(int, s->ch_assign[3], s->ch_assign[5]);
+            }
+        }
+
+    }
+
+    return 0;
+}
+
+/** Read parameters for one of the prediction filters. */
+
+static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
+                              unsigned int substr, unsigned int channel,
+                              unsigned int filter)
+{
+    SubStream *s = &m->substream[substr];
+    FilterParams *fp = &s->channel_params[channel].filter_params[filter];
+    const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;
+    const char fchar = filter ? 'I' : 'F';
+    int i, order;
+
+    // Filter is 0 for FIR, 1 for IIR.
+    av_assert0(filter < 2);
+
+    if (m->filter_changed[channel][filter]++ > 1) {
+        av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    order = get_bits(gbp, 4);
+    if (order > max_order) {
+        av_log(m->avctx, AV_LOG_ERROR,
+               "%cIR filter order %d is greater than maximum %d.\n",
+               fchar, order, max_order);
+        return AVERROR_INVALIDDATA;
+    }
+    fp->order = order;
+
+    if (order > 0) {
+        int32_t *fcoeff = s->channel_params[channel].coeff[filter];
+        int coeff_bits, coeff_shift;
+
+        fp->shift = get_bits(gbp, 4);
+
+        coeff_bits  = get_bits(gbp, 5);
+        coeff_shift = get_bits(gbp, 3);
+        if (coeff_bits < 1 || coeff_bits > 16) {
+            av_log(m->avctx, AV_LOG_ERROR,
+                   "%cIR filter coeff_bits must be between 1 and 16.\n",
+                   fchar);
+            return AVERROR_INVALIDDATA;
+        }
+        if (coeff_bits + coeff_shift > 16) {
+            av_log(m->avctx, AV_LOG_ERROR,
+                   "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
+                   fchar);
+            return AVERROR_INVALIDDATA;
+        }
+
+        for (i = 0; i < order; i++)
+            fcoeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
+
+        if (get_bits1(gbp)) {
+            int state_bits, state_shift;
+
+            if (filter == FIR) {
+                av_log(m->avctx, AV_LOG_ERROR,
+                       "FIR filter has state data specified.\n");
+                return AVERROR_INVALIDDATA;
+            }
+
+            state_bits  = get_bits(gbp, 4);
+            state_shift = get_bits(gbp, 4);
+
+            /* TODO: Check validity of state data. */
+
+            for (i = 0; i < order; i++)
+                fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
+        }
+    }
+
+    return 0;
+}
+
+/** Read parameters for primitive matrices. */
+
+static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp)
+{
+    SubStream *s = &m->substream[substr];
+    unsigned int mat, ch;
+    const int max_primitive_matrices = m->avctx->codec_id == AV_CODEC_ID_MLP
+                                     ? MAX_MATRICES_MLP
+                                     : MAX_MATRICES_TRUEHD;
+
+    if (m->matrix_changed++ > 1) {
+        av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    s->num_primitive_matrices = get_bits(gbp, 4);
+
+    if (s->num_primitive_matrices > max_primitive_matrices) {
+        av_log(m->avctx, AV_LOG_ERROR,
+               "Number of primitive matrices cannot be greater than %d.\n",
+               max_primitive_matrices);
+        return AVERROR_INVALIDDATA;
+    }
+
+    for (mat = 0; mat < s->num_primitive_matrices; mat++) {
+        int frac_bits, max_chan;
+        s->matrix_out_ch[mat] = get_bits(gbp, 4);
+        frac_bits             = get_bits(gbp, 4);
+        s->lsb_bypass   [mat] = get_bits1(gbp);
+
+        if (s->matrix_out_ch[mat] > s->max_matrix_channel) {
+            av_log(m->avctx, AV_LOG_ERROR,
+                    "Invalid channel %d specified as output from matrix.\n",
+                    s->matrix_out_ch[mat]);
+            return AVERROR_INVALIDDATA;
+        }
+        if (frac_bits > 14) {
+            av_log(m->avctx, AV_LOG_ERROR,
+                    "Too many fractional bits specified.\n");
+            return AVERROR_INVALIDDATA;
+        }
+
+        max_chan = s->max_matrix_channel;
+        if (!s->noise_type)
+            max_chan+=2;
+
+        for (ch = 0; ch <= max_chan; ch++) {
+            int coeff_val = 0;
+            if (get_bits1(gbp))
+                coeff_val = get_sbits(gbp, frac_bits + 2);
+
+            s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
+        }
+
+        if (s->noise_type)
+            s->matrix_noise_shift[mat] = get_bits(gbp, 4);
+        else
+            s->matrix_noise_shift[mat] = 0;
+    }
+
+    return 0;
+}
+
+/** Read channel parameters. */
+
+static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
+                               GetBitContext *gbp, unsigned int ch)
+{
+    SubStream *s = &m->substream[substr];
+    ChannelParams *cp = &s->channel_params[ch];
+    FilterParams *fir = &cp->filter_params[FIR];
+    FilterParams *iir = &cp->filter_params[IIR];
+    int ret;
+
+    if (s->param_presence_flags & PARAM_FIR)
+        if (get_bits1(gbp))
+            if ((ret = read_filter_params(m, gbp, substr, ch, FIR)) < 0)
+                return ret;
+
+    if (s->param_presence_flags & PARAM_IIR)
+        if (get_bits1(gbp))
+            if ((ret = read_filter_params(m, gbp, substr, ch, IIR)) < 0)
+                return ret;
+
+    if (fir->order + iir->order > 8) {
+        av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (fir->order && iir->order &&
+        fir->shift != iir->shift) {
+        av_log(m->avctx, AV_LOG_ERROR,
+                "FIR and IIR filters must use the same precision.\n");
+        return AVERROR_INVALIDDATA;
+    }
+    /* The FIR and IIR filters must have the same precision.
+     * To simplify the filtering code, only the precision of the
+     * FIR filter is considered. If only the IIR filter is employed,
+     * the FIR filter precision is set to that of the IIR filter, so
+     * that the filtering code can use it. */
+    if (!fir->order && iir->order)
+        fir->shift = iir->shift;
+
+    if (s->param_presence_flags & PARAM_HUFFOFFSET)
+        if (get_bits1(gbp))
+            cp->huff_offset = get_sbits(gbp, 15);
+
+    cp->codebook  = get_bits(gbp, 2);
+    cp->huff_lsbs = get_bits(gbp, 5);
+
+    if (cp->huff_lsbs > 24) {
+        av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
+        cp->huff_lsbs = 0;
+        return AVERROR_INVALIDDATA;
+    }
+
+    cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
+
+    return 0;
+}
+
+/** Read decoding parameters that change more often than those in the restart
+ *  header. */
+
+static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
+                                unsigned int substr)
+{
+    SubStream *s = &m->substream[substr];
+    unsigned int ch;
+    int ret;
+
+    if (s->param_presence_flags & PARAM_PRESENCE)
+        if (get_bits1(gbp))
+            s->param_presence_flags = get_bits(gbp, 8);
+
+    if (s->param_presence_flags & PARAM_BLOCKSIZE)
+        if (get_bits1(gbp)) {
+            s->blocksize = get_bits(gbp, 9);
+            if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
+                av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.\n");
+                s->blocksize = 0;
+                return AVERROR_INVALIDDATA;
+            }
+        }
+
+    if (s->param_presence_flags & PARAM_MATRIX)
+        if (get_bits1(gbp))
+            if ((ret = read_matrix_params(m, substr, gbp)) < 0)
+                return ret;
+
+    if (s->param_presence_flags & PARAM_OUTSHIFT)
+        if (get_bits1(gbp))
+            for (ch = 0; ch <= s->max_matrix_channel; ch++)
+                s->output_shift[ch] = get_sbits(gbp, 4);
+
+    if (s->param_presence_flags & PARAM_QUANTSTEP)
+        if (get_bits1(gbp))
+            for (ch = 0; ch <= s->max_channel; ch++) {
+                ChannelParams *cp = &s->channel_params[ch];
+
+                s->quant_step_size[ch] = get_bits(gbp, 4);
+
+                cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
+            }
+
+    for (ch = s->min_channel; ch <= s->max_channel; ch++)
+        if (get_bits1(gbp))
+            if ((ret = read_channel_params(m, substr, gbp, ch)) < 0)
+                return ret;
+
+    return 0;
+}
+
+#define MSB_MASK(bits)  (-1u << bits)
+
+/** Generate PCM samples using the prediction filters and residual values
+ *  read from the data stream, and update the filter state. */
+
+static void filter_channel(MLPDecodeContext *m, unsigned int substr,
+                           unsigned int channel)
+{
+    SubStream *s = &m->substream[substr];
+    const int32_t *fircoeff = s->channel_params[channel].coeff[FIR];
+    int32_t state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FIR_ORDER];
+    int32_t *firbuf = state_buffer[FIR] + MAX_BLOCKSIZE;
+    int32_t *iirbuf = state_buffer[IIR] + MAX_BLOCKSIZE;
+    FilterParams *fir = &s->channel_params[channel].filter_params[FIR];
+    FilterParams *iir = &s->channel_params[channel].filter_params[IIR];
+    unsigned int filter_shift = fir->shift;
+    int32_t mask = MSB_MASK(s->quant_step_size[channel]);
+
+    memcpy(firbuf, fir->state, MAX_FIR_ORDER * sizeof(int32_t));
+    memcpy(iirbuf, iir->state, MAX_IIR_ORDER * sizeof(int32_t));
+
+    m->dsp.mlp_filter_channel(firbuf, fircoeff,
+                              fir->order, iir->order,
+                              filter_shift, mask, s->blocksize,
+                              &m->sample_buffer[s->blockpos][channel]);
+
+    memcpy(fir->state, firbuf - s->blocksize, MAX_FIR_ORDER * sizeof(int32_t));
+    memcpy(iir->state, iirbuf - s->blocksize, MAX_IIR_ORDER * sizeof(int32_t));
+}
+
+/** Read a block of PCM residual data (or actual if no filtering active). */
+
+static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
+                           unsigned int substr)
+{
+    SubStream *s = &m->substream[substr];
+    unsigned int i, ch, expected_stream_pos = 0;
+    int ret;
+
+    if (s->data_check_present) {
+        expected_stream_pos  = get_bits_count(gbp);
+        expected_stream_pos += get_bits(gbp, 16);
+        avpriv_request_sample(m->avctx,
+                              "Substreams with VLC block size check info");
+    }
+
+    if (s->blockpos + s->blocksize > m->access_unit_size) {
+        av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    memset(&m->bypassed_lsbs[s->blockpos][0], 0,
+           s->blocksize * sizeof(m->bypassed_lsbs[0]));
+
+    for (i = 0; i < s->blocksize; i++)
+        if ((ret = read_huff_channels(m, gbp, substr, i)) < 0)
+            return ret;
+
+    for (ch = s->min_channel; ch <= s->max_channel; ch++)
+        filter_channel(m, substr, ch);
+
+    s->blockpos += s->blocksize;
+
+    if (s->data_check_present) {
+        if (get_bits_count(gbp) != expected_stream_pos)
+            av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
+        skip_bits(gbp, 8);
+    }
+
+    return 0;
+}
+
+/** Data table used for TrueHD noise generation function. */
+
+static const int8_t noise_table[256] = {
+     30,  51,  22,  54,   3,   7,  -4,  38,  14,  55,  46,  81,  22,  58,  -3,   2,
+     52,  31,  -7,  51,  15,  44,  74,  30,  85, -17,  10,  33,  18,  80,  28,  62,
+     10,  32,  23,  69,  72,  26,  35,  17,  73,  60,   8,  56,   2,   6,  -2,  -5,
+     51,   4,  11,  50,  66,  76,  21,  44,  33,  47,   1,  26,  64,  48,  57,  40,
+     38,  16, -10, -28,  92,  22, -18,  29, -10,   5, -13,  49,  19,  24,  70,  34,
+     61,  48,  30,  14,  -6,  25,  58,  33,  42,  60,  67,  17,  54,  17,  22,  30,
+     67,  44,  -9,  50, -11,  43,  40,  32,  59,  82,  13,  49, -14,  55,  60,  36,
+     48,  49,  31,  47,  15,  12,   4,  65,   1,  23,  29,  39,  45,  -2,  84,  69,
+      0,  72,  37,  57,  27,  41, -15, -16,  35,  31,  14,  61,  24,   0,  27,  24,
+     16,  41,  55,  34,  53,   9,  56,  12,  25,  29,  53,   5,  20, -20,  -8,  20,
+     13,  28,  -3,  78,  38,  16,  11,  62,  46,  29,  21,  24,  46,  65,  43, -23,
+     89,  18,  74,  21,  38, -12,  19,  12, -19,   8,  15,  33,   4,  57,   9,  -8,
+     36,  35,  26,  28,   7,  83,  63,  79,  75,  11,   3,  87,  37,  47,  34,  40,
+     39,  19,  20,  42,  27,  34,  39,  77,  13,  42,  59,  64,  45,  -1,  32,  37,
+     45,  -5,  53,  -6,   7,  36,  50,  23,   6,  32,   9, -21,  18,  71,  27,  52,
+    -25,  31,  35,  42,  -1,  68,  63,  52,  26,  43,  66,  37,  41,  25,  40,  70,
+};
+
+/** Noise generation functions.
+ *  I'm not sure what these are for - they seem to be some kind of pseudorandom
+ *  sequence generators, used to generate noise data which is used when the
+ *  channels are rematrixed. I'm not sure if they provide a practical benefit
+ *  to compression, or just obfuscate the decoder. Are they for some kind of
+ *  dithering? */
+
+/** Generate two channels of noise, used in the matrix when
+ *  restart sync word == 0x31ea. */
+
+static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
+{
+    SubStream *s = &m->substream[substr];
+    unsigned int i;
+    uint32_t seed = s->noisegen_seed;
+    unsigned int maxchan = s->max_matrix_channel;
+
+    for (i = 0; i < s->blockpos; i++) {
+        uint16_t seed_shr7 = seed >> 7;
+        m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
+        m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7)   << s->noise_shift;
+
+        seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
+    }
+
+    s->noisegen_seed = seed;
+}
+
+/** Generate a block of noise, used when restart sync word == 0x31eb. */
+
+static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
+{
+    SubStream *s = &m->substream[substr];
+    unsigned int i;
+    uint32_t seed = s->noisegen_seed;
+
+    for (i = 0; i < m->access_unit_size_pow2; i++) {
+        uint8_t seed_shr15 = seed >> 15;
+        m->noise_buffer[i] = noise_table[seed_shr15];
+        seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
+    }
+
+    s->noisegen_seed = seed;
+}
+
+
+/** Apply the channel matrices in turn to reconstruct the original audio
+ *  samples. */
+
+static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
+{
+    SubStream *s = &m->substream[substr];
+    unsigned int mat, src_ch, i;
+    unsigned int maxchan;
+
+    maxchan = s->max_matrix_channel;
+    if (!s->noise_type) {
+        generate_2_noise_channels(m, substr);
+        maxchan += 2;
+    } else {
+        fill_noise_buffer(m, substr);
+    }
+
+    for (mat = 0; mat < s->num_primitive_matrices; mat++) {
+        int matrix_noise_shift = s->matrix_noise_shift[mat];
+        unsigned int dest_ch = s->matrix_out_ch[mat];
+        int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
+        int32_t *coeffs = s->matrix_coeff[mat];
+        int index  = s->num_primitive_matrices - mat;
+        int index2 = 2 * index + 1;
+
+        /* TODO: DSPContext? */
+
+        for (i = 0; i < s->blockpos; i++) {
+            int32_t bypassed_lsb = m->bypassed_lsbs[i][mat];
+            int32_t *samples = m->sample_buffer[i];
+            int64_t accum = 0;
+
+            for (src_ch = 0; src_ch <= maxchan; src_ch++)
+                accum += (int64_t) samples[src_ch] * coeffs[src_ch];
+
+            if (matrix_noise_shift) {
+                index &= m->access_unit_size_pow2 - 1;
+                accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
+                index += index2;
+            }
+
+            samples[dest_ch] = ((accum >> 14) & mask) + bypassed_lsb;
+        }
+    }
+}
+
+/** Write the audio data into the output buffer. */
+
+static int output_data(MLPDecodeContext *m, unsigned int substr,
+                       AVFrame *frame, int *got_frame_ptr)
+{
+    AVCodecContext *avctx = m->avctx;
+    SubStream *s = &m->substream[substr];
+    unsigned int i, out_ch = 0;
+    int32_t *data_32;
+    int16_t *data_16;
+    int ret;
+    int is32 = (m->avctx->sample_fmt == AV_SAMPLE_FMT_S32);
+
+    if (m->avctx->channels != s->max_matrix_channel + 1) {
+        av_log(m->avctx, AV_LOG_ERROR, "channel count mismatch\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (!s->blockpos) {
+        av_log(avctx, AV_LOG_ERROR, "No samples to output.\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    /* get output buffer */
+    frame->nb_samples = s->blockpos;
+    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
+        return ret;
+    data_32 = (int32_t *)frame->data[0];
+    data_16 = (int16_t *)frame->data[0];
+
+    for (i = 0; i < s->blockpos; i++) {
+        for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
+            int mat_ch = s->ch_assign[out_ch];
+            int32_t sample = m->sample_buffer[i][mat_ch]
+                          << s->output_shift[mat_ch];
+            s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
+            if (is32) *data_32++ = sample << 8;
+            else      *data_16++ = sample >> 8;
+        }
+    }
+
+    *got_frame_ptr = 1;
+
+    return 0;
+}
+
+/** Read an access unit from the stream.
+ *  @return negative on error, 0 if not enough data is present in the input stream,
+ *  otherwise the number of bytes consumed. */
+
+static int read_access_unit(AVCodecContext *avctx, void* data,
+                            int *got_frame_ptr, AVPacket *avpkt)
+{
+    const uint8_t *buf = avpkt->data;
+    int buf_size = avpkt->size;
+    MLPDecodeContext *m = avctx->priv_data;
+    GetBitContext gb;
+    unsigned int length, substr;
+    unsigned int substream_start;
+    unsigned int header_size = 4;
+    unsigned int substr_header_size = 0;
+    uint8_t substream_parity_present[MAX_SUBSTREAMS];
+    uint16_t substream_data_len[MAX_SUBSTREAMS];
+    uint8_t parity_bits;
+    int ret;
+
+    if (buf_size < 4)
+        return 0;
+
+    length = (AV_RB16(buf) & 0xfff) * 2;
+
+    if (length < 4 || length > buf_size)
+        return AVERROR_INVALIDDATA;
+
+    init_get_bits(&gb, (buf + 4), (length - 4) * 8);
+
+    m->is_major_sync_unit = 0;
+    if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
+        if (read_major_sync(m, &gb) < 0)
+            goto error;
+        m->is_major_sync_unit = 1;
+        header_size += 28;
+    }
+
+    if (!m->params_valid) {
+        av_log(m->avctx, AV_LOG_WARNING,
+               "Stream parameters not seen; skipping frame.\n");
+        *got_frame_ptr = 0;
+        return length;
+    }
+
+    substream_start = 0;
+
+    for (substr = 0; substr < m->num_substreams; substr++) {
+        int extraword_present, checkdata_present, end, nonrestart_substr;
+
+        extraword_present = get_bits1(&gb);
+        nonrestart_substr = get_bits1(&gb);
+        checkdata_present = get_bits1(&gb);
+        skip_bits1(&gb);
+
+        end = get_bits(&gb, 12) * 2;
+
+        substr_header_size += 2;
+
+        if (extraword_present) {
+            if (m->avctx->codec_id == AV_CODEC_ID_MLP) {
+                av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n");
+                goto error;
+            }
+            skip_bits(&gb, 16);
+            substr_header_size += 2;
+        }
+
+        if (!(nonrestart_substr ^ m->is_major_sync_unit)) {
+            av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n");
+            goto error;
+        }
+
+        if (end + header_size + substr_header_size > length) {
+            av_log(m->avctx, AV_LOG_ERROR,
+                   "Indicated length of substream %d data goes off end of "
+                   "packet.\n", substr);
+            end = length - header_size - substr_header_size;
+        }
+
+        if (end < substream_start) {
+            av_log(avctx, AV_LOG_ERROR,
+                   "Indicated end offset of substream %d data "
+                   "is smaller than calculated start offset.\n",
+                   substr);
+            goto error;
+        }
+
+        if (substr > m->max_decoded_substream)
+            continue;
+
+        substream_parity_present[substr] = checkdata_present;
+        substream_data_len[substr] = end - substream_start;
+        substream_start = end;
+    }
+
+    parity_bits  = ff_mlp_calculate_parity(buf, 4);
+    parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
+
+    if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
+        av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
+        goto error;
+    }
+
+    buf += header_size + substr_header_size;
+
+    for (substr = 0; substr <= m->max_decoded_substream; substr++) {
+        SubStream *s = &m->substream[substr];
+        init_get_bits(&gb, buf, substream_data_len[substr] * 8);
+
+        m->matrix_changed = 0;
+        memset(m->filter_changed, 0, sizeof(m->filter_changed));
+
+        s->blockpos = 0;
+        do {
+            if (get_bits1(&gb)) {
+                if (get_bits1(&gb)) {
+                    /* A restart header should be present. */
+                    if (read_restart_header(m, &gb, buf, substr) < 0)
+                        goto next_substr;
+                    s->restart_seen = 1;
+                }
+
+                if (!s->restart_seen)
+                    goto next_substr;
+                if (read_decoding_params(m, &gb, substr) < 0)
+                    goto next_substr;
+            }
+
+            if (!s->restart_seen)
+                goto next_substr;
+
+            if ((ret = read_block_data(m, &gb, substr)) < 0)
+                return ret;
+
+            if (get_bits_count(&gb) >= substream_data_len[substr] * 8)
+                goto substream_length_mismatch;
+
+        } while (!get_bits1(&gb));
+
+        skip_bits(&gb, (-get_bits_count(&gb)) & 15);
+
+        if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) {
+            int shorten_by;
+
+            if (get_bits(&gb, 16) != 0xD234)
+                return AVERROR_INVALIDDATA;
+
+            shorten_by = get_bits(&gb, 16);
+            if      (m->avctx->codec_id == AV_CODEC_ID_TRUEHD && shorten_by  & 0x2000)
+                s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
+            else if (m->avctx->codec_id == AV_CODEC_ID_MLP    && shorten_by != 0xD234)
+                return AVERROR_INVALIDDATA;
+
+            if (substr == m->max_decoded_substream)
+                av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
+        }
+
+        if (substream_parity_present[substr]) {
+            uint8_t parity, checksum;
+
+            if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16)
+                goto substream_length_mismatch;
+
+            parity   = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
+            checksum = ff_mlp_checksum8       (buf, substream_data_len[substr] - 2);
+
+            if ((get_bits(&gb, 8) ^ parity) != 0xa9    )
+                av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr);
+            if ( get_bits(&gb, 8)           != checksum)
+                av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n"    , substr);
+        }
+
+        if (substream_data_len[substr] * 8 != get_bits_count(&gb))
+            goto substream_length_mismatch;
+
+next_substr:
+        if (!s->restart_seen)
+            av_log(m->avctx, AV_LOG_ERROR,
+                   "No restart header present in substream %d.\n", substr);
+
+        buf += substream_data_len[substr];
+    }
+
+    rematrix_channels(m, m->max_decoded_substream);
+
+    if ((ret = output_data(m, m->max_decoded_substream, data, got_frame_ptr)) < 0)
+        return ret;
+
+    return length;
+
+substream_length_mismatch:
+    av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);
+    return AVERROR_INVALIDDATA;
+
+error:
+    m->params_valid = 0;
+    return AVERROR_INVALIDDATA;
+}
+
+#if CONFIG_MLP_DECODER
+AVCodec ff_mlp_decoder = {
+    .name           = "mlp",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_MLP,
+    .priv_data_size = sizeof(MLPDecodeContext),
+    .init           = mlp_decode_init,
+    .decode         = read_access_unit,
+    .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
+};
+#endif
+#if CONFIG_TRUEHD_DECODER
+AVCodec ff_truehd_decoder = {
+    .name           = "truehd",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_TRUEHD,
+    .priv_data_size = sizeof(MLPDecodeContext),
+    .init           = mlp_decode_init,
+    .decode         = read_access_unit,
+    .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("TrueHD"),
+};
+#endif /* CONFIG_TRUEHD_DECODER */