Index: configure
===================================================================
--- configure	(revision 11448)
+++ configure	(working copy)
@@ -781,6 +781,7 @@
 h263_decoder_deps="h263_parser mpeg4video_parser"
 h263i_decoder_deps="h263_decoder"
 h264_decoder_deps="h264_parser"
+mlp_decoder_deps="mlp_parser"
 mpeg_xvmc_decoder_deps="xvmc"
 mpeg4_decoder_deps="h263_decoder"
 msmpeg4v1_decoder_deps="h263_decoder"
Index: libavcodec/mlpdec.c
===================================================================
--- libavcodec/mlpdec.c	(revision 0)
+++ libavcodec/mlpdec.c	(revision 0)
@@ -0,0 +1,1105 @@
+/*
+ * 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 mlpdec.c
+ * MLP decoder
+ */
+
+#include "avcodec.h"
+#include "intreadwrite.h"
+#include "bitstream.h"
+#include "crc.h"
+#include "parser.h"
+#include "mlp_parser.h"
+
+/** Maximum number of channels that can be decoded. */
+#define MAX_CHANNELS        16
+
+/** Maximum number of matrices used in decoding. Most streams have one matrix
+ *  per output channel, but some rematrix a channel (usually 0) more than once.
+ */
+
+#define MAX_MATRICES        15
+
+/** Maximum number of substreams that can be decoded. This could also be set
+ *  higher, but again I haven't seen any examples with more than two. */
+#define MAX_SUBSTREAMS      2
+
+/** Maximum sample frequency supported. */
+#define MAX_SAMPLERATE      192000
+
+/** The maximum number of audio samples within one access unit. */
+#define MAX_BLOCKSIZE       (40 * (MAX_SAMPLERATE / 48000))
+/** The next power of two greater than MAX_BLOCKSIZE. */
+#define MAX_BLOCKSIZE_POW2  (64 * (MAX_SAMPLERATE / 48000))
+
+/** The maximum number of taps in either the IIR or FIR filter.
+ *  I believe MLP actually specifies the maximum order for IIR filters is four,
+ *  and that the sum of the orders of both filters must be <= 8. */
+#define MAX_FILTER_ORDER    8
+
+/** 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.mplayerhq.hu/bugreports.html and include "
+    "a sample of this file.";
+
+typedef struct MLPDecodeContext {
+    AVCodecContext *avctx;
+
+    //! Do we have valid stream data read from a major sync block?
+    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;
+
+    //! For each substream, whether a restart header has been read
+    uint8_t     restart_seen[MAX_SUBSTREAMS];
+
+    //@{
+    /** Restart header data */
+    //! The sync word used at the start of the last restart header
+    uint16_t    restart_sync_word[MAX_SUBSTREAMS];
+
+    //! The index of the first channel coded in this substream
+    uint8_t     min_channel[MAX_SUBSTREAMS];
+    //! The index of the last channel coded in this substream
+    uint8_t     max_channel[MAX_SUBSTREAMS];
+    //! The number of channels input into the rematrix stage
+    uint8_t     max_matrix_channel[MAX_SUBSTREAMS];
+
+    //| The left shift applied to random noise in 0x31ea substreams
+    uint8_t     noise_shift[MAX_SUBSTREAMS];
+    //! The current seed value for the pseudorandom noise generator(s)
+    uint32_t    noisegen_seed[MAX_SUBSTREAMS];
+
+    //! Does this substream contain extra info to check the size of VLC blocks?
+    uint8_t     data_check_present[MAX_SUBSTREAMS];
+
+    //! Bitmask of which parameter sets are conveyed in a decoding parameter block
+    uint8_t     param_presence_flags[MAX_SUBSTREAMS];
+    //@}
+
+    //@{
+    /** Matrix data */
+
+    //! Number of matrices to be applied
+    uint8_t     num_primitive_matrices[MAX_SUBSTREAMS];
+
+    //! Output channel of matrix
+    uint8_t     matrix_ch[MAX_SUBSTREAMS][MAX_MATRICES];
+
+    //! Whether the LSBs of the matrix output are encoded in the bitstream
+    uint8_t     lsb_bypass[MAX_SUBSTREAMS][MAX_MATRICES];
+    //! Matrix coefficients, stored as 2.14 fixed point
+    int32_t     matrix_coeff[MAX_SUBSTREAMS][MAX_MATRICES][MAX_CHANNELS+2];
+    //! Left shift to apply to noise values in 0x31eb substreams
+    uint8_t     matrix_noise_shift[MAX_SUBSTREAMS][MAX_MATRICES];
+    //@}
+
+    //! Left shift to apply to huffman-decoded residuals
+    uint8_t     quant_step_size[MAX_SUBSTREAMS][MAX_CHANNELS];
+
+    //! Number of PCM samples in current audio block
+    uint16_t    blocksize[MAX_SUBSTREAMS];
+    //! Number of PCM samples decoded so far in this frame
+    uint16_t    blockpos[MAX_SUBSTREAMS];
+
+    //! Left shift to apply to decoded PCM values to get final 24-bit output
+    int8_t      output_shift[MAX_SUBSTREAMS][MAX_CHANNELS];
+
+    //@{
+    /* Filter data. Filter 0 is an FIR filter, filter 1 IIR. */
+    //! Number of taps in filter
+    uint8_t     filter_order[MAX_CHANNELS][2];
+    //! Right shift to apply to output of filter
+    uint8_t     filter_coeff_q[MAX_CHANNELS][2];
+
+    int32_t     filter_coeff[MAX_CHANNELS][2][MAX_FILTER_ORDER];
+    int32_t     filter_state[MAX_CHANNELS][2][MAX_FILTER_ORDER];
+    //@}
+
+    //@{
+    /** Sample data coding infomation */
+    //! Offset to apply to residual values
+    int16_t     huff_offset[MAX_CHANNELS];
+    //! Sign/rounding corrected version of huff_offset
+    int32_t     sign_huff_offset[MAX_CHANNELS];
+    //! Which VLC codebook to use to read residuals
+    uint8_t     codebook[MAX_CHANNELS];
+    //! Size of residual suffix not encoded using VLC
+    uint8_t     huff_lsbs[MAX_CHANNELS];
+    //@}
+
+    //! Running XOR of all output samples
+    int32_t     lossless_check_data[MAX_SUBSTREAMS];
+
+    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;
+
+/** Tables defining the huffman codes.
+ *  There are three entropy coding methods used in MLP (four if you count "none"
+ *  as a method). These use the same sequences for codes starting 00... or 01...
+ *  but have different codes starting 1....
+ */
+
+static const uint8_t huffman_tables[3][18][2] = {
+    { /* Huffman table 0, -7 - +10 */
+        {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},
+        {0x04, 3}, {0x05, 3}, {0x06, 3}, {0x07, 3},
+        {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},
+    }, { /* Huffman table 1, -7 - +8 */
+        {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},
+        {0x02, 2}, {0x03, 2},
+        {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},
+    }, { /* Huffman table 2, -7 - +7 */
+        {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},
+        {0x01, 1},
+        {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},
+    }
+};
+
+static VLC huff_vlc[3];
+
+static AVCRC crc_63[1024];
+static AVCRC crc_1D[1024];
+
+
+/** Initialize static data, constant between all invocations of the codec. */
+
+static void init_static()
+{
+    if (!huff_vlc[0].bits) {
+        init_vlc(&huff_vlc[0], VLC_BITS, 18,
+                 &huffman_tables[0][0][1], 2, 1,
+                 &huffman_tables[0][0][0], 2, 1, 1);
+        init_vlc(&huff_vlc[1], VLC_BITS, 16,
+                 &huffman_tables[1][0][1], 2, 1,
+                 &huffman_tables[1][0][0], 2, 1, 1);
+        init_vlc(&huff_vlc[2], VLC_BITS, 15,
+                 &huffman_tables[2][0][1], 2, 1,
+                 &huffman_tables[2][0][0], 2, 1, 1);
+
+        av_crc_init(crc_63, 0,  8,   0x63, sizeof(crc_63));
+        av_crc_init(crc_1D, 0,  8,   0x1D, sizeof(crc_1D));
+    }
+}
+
+
+/** MLP uses checksums that seem to be based on the standard CRC algorithm,
+ *  but not (in implementation terms, the table lookup and XOR are reversed).
+ *  We can implement this behaviour using a standard av_crc on all but the
+ *  last element, then XOR that with the last element.
+ */
+
+static uint8_t mlp_checksum8(const uint8_t *buf, unsigned int buf_size)
+{
+    uint8_t checksum = av_crc(crc_63, 0x3c, buf, buf_size - 1); // crc_63[0xa2] == 0x3c
+    checksum ^= buf[buf_size-1];
+    return checksum;
+}
+
+/** Calculate an 8-bit checksum over a restart header -- a non-multiple-of-8
+ *  number of bits, starting two bits into the first byte of buf.
+ */
+
+static uint8_t mlp_restart_checksum(const uint8_t *buf, unsigned int bit_size)
+{
+    int i;
+    int num_bytes = (bit_size + 2) / 8;
+
+    int crc = crc_1D[buf[0] & 0x3f];
+    crc = av_crc(crc_1D, crc, buf + 1, num_bytes - 2);
+    crc ^= buf[num_bytes - 1];
+
+    for (i = 0; i < ((bit_size + 2) & 7); i++) {
+        crc <<= 1;
+        if (crc & 0x100)
+            crc ^= 0x11D;
+        crc ^= (buf[num_bytes] >> (7 - i)) & 1;
+    }
+
+    return crc;
+}
+
+static inline void calculate_sign_huff(MLPDecodeContext *m, unsigned int substr,
+                                       unsigned int ch)
+{
+    int lsb_bits = m->huff_lsbs[ch] - m->quant_step_size[substr][ch];
+    int sign_shift = lsb_bits + (m->codebook[ch] ? 2 - m->codebook[ch] : -1);
+
+    m->sign_huff_offset[ch] = m->huff_offset[ch];
+
+    if (m->codebook[ch] > 0)
+        m->sign_huff_offset[ch] -= 7 << lsb_bits;
+
+    if (sign_shift >= 0)
+        m->sign_huff_offset[ch] -= 1 << sign_shift;
+}
+
+/** Read a sample, consisting of either, both or neither of entropy-coded MSBs
+ *  and plain LSBs.
+ */
+
+static inline int read_huff(MLPDecodeContext *m, GetBitContext *gbp,
+                            unsigned int substr, unsigned int channel)
+{
+    int codebook = m->codebook[channel];
+    int quant_step_size = m->quant_step_size[substr][channel];
+    int lsb_bits = m->huff_lsbs[channel] - 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 (lsb_bits > 0)
+        result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
+
+    result += m->sign_huff_offset[channel];
+
+    return result << quant_step_size;
+}
+
+/** Initialize the decoder. */
+
+static int mlp_decode_init(AVCodecContext *avctx)
+{
+    MLPDecodeContext *m = avctx->priv_data;
+
+    init_static();
+    m->avctx = avctx;
+    memset(m->lossless_check_data, 0xff, sizeof(m->lossless_check_data));
+    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, const uint8_t *buf,
+                           unsigned int buf_size)
+{
+    MLPHeaderInfo mh;
+
+    if (ff_mlp_read_major_sync(m->avctx, &mh, buf, buf_size) != 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 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 maximum supported (%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 maximum supported (%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 maximum supported (%d)\n",
+               mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
+        return -1;
+    }
+
+    if (mh.num_substreams == 0)
+        return -1;
+    if (mh.num_substreams > MAX_SUBSTREAMS) {
+        av_log(m->avctx, AV_LOG_ERROR,
+               "Number of substreams %d is more than maximum supported by "
+               "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;
+
+#ifdef CONFIG_AUDIO_NONSHORT
+    m->avctx->bits_per_sample = mh.group1_bits;
+    if (mh.group1_bits > 16) {
+        m->avctx->sample_fmt = SAMPLE_FMT_S32;
+    }
+#endif
+
+    m->params_valid = 1;
+    memset(m->restart_seen, 0, sizeof(m->restart_seen));
+
+    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)
+{
+    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, 14);
+
+    if ((sync_word & 0x3ffe) != 0x31ea) {
+        av_log(m->avctx, AV_LOG_ERROR,
+               "Restart header sync incorrect (got 0x%04x)\n", sync_word);
+        return -1;
+    }
+    m->restart_sync_word[substr] = sync_word;
+
+    skip_bits(gbp, 16); /* Output timestamp */
+
+    m->min_channel[substr]        = get_bits(gbp, 4);
+    m->max_channel[substr]        = get_bits(gbp, 4);
+    m->max_matrix_channel[substr] = get_bits(gbp, 4);
+
+    if (m->min_channel[substr] > m->max_channel[substr]) {
+        av_log(m->avctx, AV_LOG_ERROR,
+               "Substream min channel cannot be greater than max channel.\n");
+        m->min_channel[substr] = m->max_channel[substr]
+            = m->max_matrix_channel[substr] = 0;
+        return -1;
+    }
+
+    if (m->avctx->request_channels > 0
+        && m->max_channel[substr] + 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",
+               m->max_channel[substr] + 1, substr);
+        m->max_decoded_substream = substr;
+    }
+
+    m->noise_shift[substr] = get_bits(gbp, 4);
+    m->noisegen_seed[substr] = get_bits(gbp, 23);
+
+    skip_bits(gbp, 19);
+
+    m->data_check_present[substr] = get_bits1(gbp);
+    lossless_check = get_bits(gbp, 8);
+    if (substr == m->max_decoded_substream
+        && m->lossless_check_data[substr] != 0xffffffff) {
+        tmp = m->lossless_check_data[substr];
+        tmp ^= tmp >> 16;
+        tmp ^= tmp >> 8;
+        tmp &= 0xff;
+        if (tmp != lossless_check)
+            av_log(m->avctx, AV_LOG_WARNING,
+                   "Lossless check failed - expected %x, calculated %x\n",
+                   lossless_check, tmp);
+        else
+            dprintf(m->avctx, "Lossless check passed for substream %d (%x)\n",
+                    substr, tmp);
+    }
+
+    skip_bits(gbp, 16);
+
+    for (ch = 0; ch <= m->max_matrix_channel[substr]; ch++) {
+        int ch_assign = get_bits(gbp, 6);
+        dprintf(m->avctx, "ch_assign[%d][%d] = %d\n", substr, ch,
+                ch_assign);
+        if (ch_assign != ch) {
+            av_log(m->avctx, AV_LOG_ERROR,
+                   "Non 1:1 channel assignments are used in this stream. %s\n",
+                   sample_message);
+            return -1;
+        }
+    }
+
+    checksum = 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 */
+    m->param_presence_flags[substr] = 0xff;
+    m->num_primitive_matrices[substr] = 0;
+    m->blocksize[substr] = 8;
+    m->lossless_check_data[substr] = 0;
+
+    memset(m->output_shift[substr],    0, sizeof(m->output_shift[substr]));
+    memset(m->quant_step_size[substr], 0, sizeof(m->quant_step_size[substr]));
+
+    for (ch = m->min_channel[substr]; ch <= m->max_channel[substr]; ch++) {
+        m->filter_order  [ch][0] = 0;
+        m->filter_order  [ch][1] = 0;
+        m->filter_coeff_q[ch][0] = 0;
+        m->filter_coeff_q[ch][1] = 0;
+
+        memset(m->filter_coeff[ch], 0, sizeof(m->filter_coeff[ch]));
+        memset(m->filter_state[ch], 0, sizeof(m->filter_state[ch]));
+
+        /* Default audio coding is 24-bit raw PCM */
+        m->huff_offset[ch]      = 0;
+        m->sign_huff_offset[ch] = (-1) << 23;
+        m->codebook[ch]         = 0;
+        m->huff_lsbs[ch]        = 24;
+    }
+
+    if (substr == m->max_decoded_substream) {
+        m->avctx->channels = m->max_channel[substr] + 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)
+{
+    int i, order;
+
+    // filter is 0 for FIR, 1 for IIR
+    assert(filter < 2);
+
+    order = get_bits(gbp, 4);
+    if (order > MAX_FILTER_ORDER) {
+        av_log(m->avctx, AV_LOG_ERROR,
+               "%cIR filter order %d is greater than maximum %d\n",
+               filter ? 'I' : 'F', order, MAX_FILTER_ORDER);
+        return -1;
+    }
+    m->filter_order[channel][filter] = order;
+
+    if (order > 0) {
+        int coeff_bits, coeff_shift;
+
+        m->filter_coeff_q[channel][filter] = 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",
+                   filter ? 'I' : 'F');
+            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",
+                   filter ? 'I' : 'F');
+            return -1;
+        }
+
+        for (i = 0; i < order; i++)
+            m->filter_coeff[channel][filter][i] =
+                    get_sbits(gbp, coeff_bits) << coeff_shift;
+
+        if (get_bits1(gbp)) {
+            int state_bits, state_shift;
+
+            if (filter == 0) {
+                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++)
+                m->filter_state[channel][filter][i] =
+                    get_sbits(gbp, state_bits) << state_shift;
+        }
+    }
+
+    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)
+{
+    unsigned int mat, ch;
+
+    if (get_bits1(gbp))
+        m->param_presence_flags[substr] = get_bits(gbp, 8);
+
+    if (m->param_presence_flags[substr] & 0x80)
+        if (get_bits1(gbp)) {
+            m->blocksize[substr] = get_bits(gbp, 9);
+            if (m->blocksize[substr] > MAX_BLOCKSIZE) {
+                av_log(m->avctx, AV_LOG_ERROR, "Block size too large\n");
+                m->blocksize[substr] = 0;
+                return -1;
+            }
+        }
+
+    if (m->param_presence_flags[substr] & 0x40)
+        if (get_bits1(gbp)) {
+            m->num_primitive_matrices[substr] = get_bits(gbp, 4);
+
+            for (mat = 0; mat < m->num_primitive_matrices[substr]; mat++) {
+                int frac_bits, max_chan;
+                m->matrix_ch[substr][mat] = get_bits(gbp, 4);
+                frac_bits = get_bits(gbp, 4);
+                m->lsb_bypass[substr][mat] = get_bits1(gbp);
+
+                if (m->matrix_ch[substr][mat] > m->max_channel[substr]) {
+                    av_log(m->avctx, AV_LOG_ERROR,
+                           "Invalid channel %d specified as output from matrix\n",
+                           m->matrix_ch[substr][mat]);
+                    m->matrix_ch[substr][mat] = 0;
+                    return -1;
+                }
+                if (frac_bits > 14) {
+                    av_log(m->avctx, AV_LOG_ERROR,
+                           "Too many fractional bits specified\n");
+                    return -1;
+                }
+
+                max_chan = m->max_matrix_channel[substr];
+                if (m->restart_sync_word[substr] == 0x31ea)
+                    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);
+
+                    m->matrix_coeff[substr][mat][ch] = coeff_val << (14 - frac_bits);
+                }
+
+                if (m->restart_sync_word[substr] == 0x31eb)
+                    m->matrix_noise_shift[substr][mat] = get_bits(gbp, 4);
+                else
+                    m->matrix_noise_shift[substr][mat] = 0;
+            }
+        }
+
+    if (m->param_presence_flags[substr] & 0x20)
+        if (get_bits1(gbp)) {
+            for (ch = 0; ch <= m->max_matrix_channel[substr]; ch++) {
+                m->output_shift[substr][ch] = get_bits(gbp, 4);
+                dprintf(m->avctx, "output shift[%d] = %d\n",
+                        ch, m->output_shift[substr][ch]);
+                /* TODO: validate */
+            }
+        }
+
+    if (m->param_presence_flags[substr] & 0x10)
+        if (get_bits1(gbp))
+            for (ch = 0; ch <= m->max_channel[substr]; ch++) {
+                m->quant_step_size[substr][ch] = get_bits(gbp, 4);
+                /* TODO: validate */
+
+                calculate_sign_huff(m, substr, ch);
+            }
+
+    for (ch = m->min_channel[substr]; ch <= m->max_channel[substr]; ch++)
+        if (get_bits1(gbp)) {
+            if (m->param_presence_flags[substr] & 0x08)
+                if (get_bits1(gbp))
+                    if (read_filter_params(m, gbp, ch, 0) < 0)
+                        return -1;
+
+            if (m->param_presence_flags[substr] & 0x04)
+                if (get_bits1(gbp))
+                    if (read_filter_params(m, gbp, ch, 1) < 0)
+                        return -1;
+
+            if (m->filter_order[ch][0] > 0 && m->filter_order[ch][1] > 0
+                && m->filter_coeff_q[ch][0] != m->filter_coeff_q[ch][1]) {
+                av_log(m->avctx, AV_LOG_ERROR,
+                       "FIR and IIR filters must use same precision\n");
+                return -1;
+            }
+
+            if (m->param_presence_flags[substr] & 0x02)
+                if (get_bits1(gbp))
+                    m->huff_offset[ch] = get_sbits(gbp, 15);
+
+            m->codebook[ch] = get_bits(gbp, 2);
+            m->huff_lsbs[ch] = get_bits(gbp, 5);
+
+            calculate_sign_huff(m, substr, ch);
+
+            /* TODO: validate */
+        }
+
+    return 0;
+}
+
+/** Generate a PCM sample using the prediction filters and a residual value
+ *  read from the data stream, and update the filter state.
+ */
+
+static int filter_sample(MLPDecodeContext *m, unsigned int substr,
+                         unsigned int channel, int32_t residual)
+{
+    unsigned int i, j;
+    int64_t accum = 0;
+    int32_t result;
+
+    /* TODO: Move this code to DSPContext? */
+
+    for (j = 0; j < 2; j++)
+        for (i = 0; i < m->filter_order[channel][j]; i++)
+            accum += (int64_t)m->filter_state[channel][j][i] *
+                     m->filter_coeff[channel][j][i];
+
+    accum = accum >> m->filter_coeff_q[channel][0];
+    result = (accum + residual)
+                & ~((1 << m->quant_step_size[substr][channel]) - 1);
+
+    memmove(&m->filter_state[channel][0][1], &m->filter_state[channel][0][0],
+            sizeof(m->filter_state[channel][0][0]) * (MAX_FILTER_ORDER * 2 - 1));
+
+    m->filter_state[channel][0][0] = result;
+    m->filter_state[channel][1][0] = result - accum;
+
+    return result;
+}
+
+/** Read a block of PCM residual (or actual if no filtering active) data.
+ */
+
+static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
+                           unsigned int substr)
+{
+    unsigned int i, mat, ch, expected_stream_pos = 0;
+
+    if (m->data_check_present[substr])
+        expected_stream_pos = get_bits_count(gbp) + get_bits(gbp, 16);
+        /* UNTESTED - find an example stream */
+
+    if (m->blockpos[substr] + m->blocksize[substr] > m->access_unit_size) {
+        av_log(m->avctx, AV_LOG_ERROR, "Too many audio samples in frame\n");
+        return -1;
+    }
+
+    memset(&m->bypassed_lsbs[m->blockpos[substr]][0], 0,
+           m->blocksize[substr] * MAX_CHANNELS);
+
+    for (i = 0; i < m->blocksize[substr]; i++) {
+        for (mat = 0; mat < m->num_primitive_matrices[substr]; mat++)
+            if (m->lsb_bypass[substr][mat])
+                m->bypassed_lsbs[i + m->blockpos[substr]][mat] = get_bits1(gbp);
+
+        for (ch = m->min_channel[substr]; ch <= m->max_channel[substr]; ch++) {
+            int32_t sample = read_huff(m, gbp, substr, ch);
+            int32_t filtered = filter_sample(m, substr, ch, sample);
+
+            m->sample_buffer[i + m->blockpos[substr]][ch] = filtered;
+        }
+    }
+
+    m->blockpos[substr] += m->blocksize[substr];
+
+    if (m->data_check_present[substr]) {
+        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 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_noise_1(MLPDecodeContext *m, unsigned int substr)
+{
+    unsigned int i;
+    uint32_t seed = m->noisegen_seed[substr];
+    unsigned int maxchan = m->max_matrix_channel[substr];
+
+    for (i = 0; i < m->blockpos[substr]; i++) {
+        uint16_t seed_shr7 = seed >> 7;
+        m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << m->noise_shift[substr];
+        m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7)   << m->noise_shift[substr];
+
+        seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
+    }
+
+    m->noisegen_seed[substr] = seed;
+}
+
+/** Generate a block of noise, used when restart_sync_word == 0x31eb. */
+
+static void generate_noise_2(MLPDecodeContext *m, unsigned int substr)
+{
+    unsigned int i;
+    uint32_t seed = m->noisegen_seed[substr];
+
+    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);
+    }
+
+    m->noisegen_seed[substr] = seed;
+}
+
+
+/** Apply the channel matrices in turn to reconstruct the original audio samples.
+ */
+
+static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
+{
+    unsigned int mat, dest_ch, src_ch, i;
+    unsigned int maxchan;
+
+    maxchan = m->max_matrix_channel[substr];
+    if (m->restart_sync_word[substr] == 0x31ea) {
+        generate_noise_1(m, substr);
+        maxchan += 2;
+    } else {
+        generate_noise_2(m, substr);
+    }
+
+    for (mat = 0; mat < m->num_primitive_matrices[substr]; mat++) {
+        dest_ch = m->matrix_ch[substr][mat];
+
+        /* TODO: DSPContext? */
+
+        for (i = 0; i < m->blockpos[substr]; i++) {
+            int64_t accum = 0;
+            for (src_ch = 0; src_ch <= maxchan; src_ch++) {
+                accum += (int64_t)m->sample_buffer[i][src_ch]
+                                  * m->matrix_coeff[substr][mat][src_ch];
+            }
+            if (m->matrix_noise_shift[substr][mat]) {
+                uint32_t index = m->num_primitive_matrices[substr] - mat;
+                index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1);
+                accum += m->noise_buffer[index] << (m->matrix_noise_shift[substr][mat] + 7);
+            }
+            m->sample_buffer[i][dest_ch] = ((accum >> 14) & ~((1 << m->quant_step_size[substr][dest_ch]) - 1))
+                                             + 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)
+{
+    unsigned int i, ch = 0;
+    int32_t *data_32 = (int32_t*) data;
+    int16_t *data_16 = (int16_t*) data;
+
+    if (*data_size < m->max_channel[substr] * m->blockpos[substr]
+                      * (is32 ? 4 : 2))
+        return -1;
+
+    for (i = 0; i < m->blockpos[substr]; i++) {
+        for (ch = 0; ch <= m->max_channel[substr]; ch++) {
+            int32_t sample = m->sample_buffer[i][ch] << m->output_shift[substr][ch];
+            m->lossless_check_data[substr] ^= (sample & 0xffffff) << ch;
+            if (is32) *data_32++ = sample << 8;
+            else      *data_16++ = sample >> 8;
+        }
+    }
+
+    *data_size = i * 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);
+}
+
+
+/** XOR together all the bytes of a buffer.
+ *  Does this belong in dspcontext?
+ */
+
+static uint8_t calculate_parity(const uint8_t *buf, unsigned int buf_size)
+{
+    uint32_t scratch = 0;
+    const uint8_t *buf_end = buf + buf_size;
+
+    for (; buf < buf_end - 3; buf += 4)
+        scratch ^= *((const uint32_t*)buf);
+
+    scratch ^= scratch >> 16;
+    scratch ^= scratch >> 8;
+    scratch &= 0xff;
+
+    for (; buf < buf_end; buf++)
+        scratch ^= *buf;
+
+    return scratch;
+}
+
+/**
+ * Read an access unit from the stream.
+ * Returns -1 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,
+                            uint8_t *buf, int buf_size)
+{
+    MLPDecodeContext *m = avctx->priv_data;
+    GetBitContext gb;
+    unsigned int length, substr;
+    unsigned int substream_start;
+    unsigned int header_size;
+    uint8_t substream_parity_present[MAX_SUBSTREAMS];
+    uint16_t substream_data_len[MAX_SUBSTREAMS];
+
+    if (buf_size < 2)
+        return 0;
+
+    length = AV_RB16(buf) & 0xfff;
+
+    if (length * 2 > buf_size)
+        return -1;
+
+    init_get_bits(&gb, buf, length * 16);
+    skip_bits_long(&gb, 32);
+
+    if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
+        dprintf(m->avctx, "Found major sync\n");
+        if (read_major_sync(m, buf + 4, buf_size - 4) < 0)
+            goto error;
+        skip_bits_long(&gb, 28 * 8);
+    }
+
+    if (!m->params_valid) {
+        av_log(m->avctx, AV_LOG_WARNING,
+               "Stream parameters not seen; skipping frame");
+        return length * 2;
+    }
+
+    header_size = get_bits_count(&gb) >> 4;
+    substream_start = 0;
+
+    for (substr = 0; substr < m->num_substreams; substr++) {
+        int extraword_present, checkdata_present, end;
+
+        extraword_present = get_bits1(&gb);
+        skip_bits1(&gb);
+        checkdata_present = get_bits1(&gb);
+        skip_bits1(&gb);
+
+        end = get_bits(&gb, 12);
+
+        if (extraword_present)
+            skip_bits(&gb, 16);
+
+        if (end + header_size > length) {
+            av_log(m->avctx, AV_LOG_ERROR,
+                   "Substream %d data indicated length goes off end of packet.",
+                   substr);
+            end = length - header_size;
+        }
+
+        if (substr > m->max_decoded_substream)
+            continue;
+
+        substream_parity_present[substr] = checkdata_present;
+        substream_data_len[substr] = end - substream_start;
+        substream_start = end;
+    }
+
+    buf += get_bits_count(&gb) >> 3;
+    buf_size -= get_bits_count(&gb) >> 3;
+
+    for (substr = 0; substr <= m->max_decoded_substream; substr++) {
+        init_get_bits(&gb, buf, substream_data_len[substr] * 16);
+
+        m->blockpos[substr] = 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 error;
+                    m->restart_seen[substr] = 1;
+                }
+
+                if (!m->restart_seen[substr]) {
+                    av_log(m->avctx, AV_LOG_ERROR,
+                           "No restart header present in substream %d.\n",
+                           substr);
+                    goto error;
+                }
+
+                if (read_decoding_params(m, &gb, substr) < 0)
+                    goto error;
+            }
+
+            if (!m->restart_seen[substr]) {
+                av_log(m->avctx, AV_LOG_ERROR,
+                       "No restart header present in substream %d.\n",
+                       substr);
+                goto error;
+            }
+
+            if (read_block_data(m, &gb, substr) < 0)
+                return -1;
+
+        } while ((get_bits_count(&gb) < substream_data_len[substr] * 16)
+                 && get_bits1(&gb) == 0);
+
+        skip_bits(&gb, (-get_bits_count(&gb)) & 15);
+        if ((substream_data_len[substr] * 16) - get_bits_count(&gb) >= 48 &&
+            (show_bits_long(&gb, 32) == 0xd234d234 ||
+             show_bits_long(&gb, 20) == 0xd234e)) {
+            skip_bits(&gb, 18);
+            if (substr == m->max_decoded_substream)
+                av_log(m->avctx, AV_LOG_INFO, "End of stream indicated\n");
+
+            if (get_bits1(&gb)) {
+                int shorten_by = get_bits(&gb, 13);
+                shorten_by = FFMIN(shorten_by, m->blockpos[substr]);
+                m->blockpos[substr] -= shorten_by;
+            } else
+                skip_bits(&gb, 13);
+        }
+        if (substream_parity_present[substr]) {
+            uint8_t parity, checksum;
+
+            parity = calculate_parity(buf, substream_data_len[substr] * 2 - 2);
+            if ((parity ^ get_bits(&gb, 8)) != 0xa9)
+                av_log(m->avctx, AV_LOG_ERROR,
+                       "Substream %d parity check failed\n", substr);
+
+            checksum = mlp_checksum8(buf, substream_data_len[substr] * 2 - 2);
+            if (checksum != get_bits(&gb, 8))
+                av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed\n",
+                       substr);
+        }
+        if (substream_data_len[substr] * 16 != get_bits_count(&gb)) {
+            av_log(m->avctx, AV_LOG_ERROR, "Substream %d length mismatch.\n",
+                   substr);
+            return -1;
+        }
+
+        buf += substream_data_len[substr] * 2;
+        buf_size -= substream_data_len[substr] * 2;
+    }
+
+    rematrix_channels(m, substr - 1);
+
+    if (output_data(m, substr - 1, data, data_size) < 0)
+        return -1;
+
+    return length * 2;
+
+error:
+    m->params_valid = 0;
+    return -1;
+}
+
+AVCodec mlp_decoder = {
+    "mlp",
+    CODEC_TYPE_AUDIO,
+    CODEC_ID_MLP,
+    sizeof(MLPDecodeContext),
+    mlp_decode_init,
+    NULL,
+    NULL,
+    read_access_unit,
+};
+
Index: libavcodec/allcodecs.c
===================================================================
--- libavcodec/allcodecs.c	(revision 11448)
+++ libavcodec/allcodecs.c	(working copy)
@@ -184,6 +184,7 @@
     REGISTER_DECODER (IMC, imc);
     REGISTER_DECODER (MACE3, mace3);
     REGISTER_DECODER (MACE6, mace6);
+    REGISTER_DECODER (MLP, mlp);
     REGISTER_ENCDEC  (MP2, mp2);
     REGISTER_DECODER (MP3, mp3);
     REGISTER_DECODER (MP3ADU, mp3adu);
Index: libavcodec/Makefile
===================================================================
--- libavcodec/Makefile	(revision 11448)
+++ libavcodec/Makefile	(working copy)
@@ -109,6 +109,7 @@
 OBJS-$(CONFIG_MJPEG_DECODER)           += mjpegdec.o mjpeg.o
 OBJS-$(CONFIG_MJPEG_ENCODER)           += mjpegenc.o mjpeg.o mpegvideo_enc.o motion_est.o ratecontrol.o
 OBJS-$(CONFIG_MJPEGB_DECODER)          += mjpegbdec.o mjpegdec.o mjpeg.o
+OBJS-$(CONFIG_MLP_DECODER)             += mlpdec.o
 OBJS-$(CONFIG_MMVIDEO_DECODER)         += mmvideo.o
 OBJS-$(CONFIG_MP2_DECODER)             += mpegaudiodec.o mpegaudiodecheader.o mpegaudio.o mpegaudiodata.o
 OBJS-$(CONFIG_MP2_ENCODER)             += mpegaudioenc.o mpegaudio.o mpegaudiodata.o
