changeset 7194:8427f12555a6 libavcodec

MLP/TrueHD decoder.
author ramiro
date Fri, 04 Jul 2008 15:44:13 +0000
parents 3ab9c2bc0413
children 5e5c1178ae9a
files Makefile allcodecs.c avcodec.h mlpdec.c
diffstat 4 files changed, 1184 insertions(+), 2 deletions(-) [+]
line wrap: on
line diff
--- a/Makefile	Fri Jul 04 15:37:52 2008 +0000
+++ b/Makefile	Fri Jul 04 15:44:13 2008 +0000
@@ -107,6 +107,7 @@
 OBJS-$(CONFIG_MJPEG_DECODER)           += mjpegdec.o mjpeg.o
 OBJS-$(CONFIG_MJPEG_ENCODER)           += mjpegenc.o mjpeg.o mpegvideo_enc.o motion_est.o ratecontrol.o mpeg12data.o mpegvideo.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
--- a/allcodecs.c	Fri Jul 04 15:37:52 2008 +0000
+++ b/allcodecs.c	Fri Jul 04 15:44:13 2008 +0000
@@ -189,6 +189,7 @@
     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);
--- a/avcodec.h	Fri Jul 04 15:37:52 2008 +0000
+++ b/avcodec.h	Fri Jul 04 15:44:13 2008 +0000
@@ -30,8 +30,8 @@
 #include "libavutil/avutil.h"
 
 #define LIBAVCODEC_VERSION_MAJOR 51
-#define LIBAVCODEC_VERSION_MINOR 57
-#define LIBAVCODEC_VERSION_MICRO  2
+#define LIBAVCODEC_VERSION_MINOR 58
+#define LIBAVCODEC_VERSION_MICRO  0
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
                                                LIBAVCODEC_VERSION_MINOR, \
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/mlpdec.c	Fri Jul 04 15:44:13 2008 +0000
@@ -0,0 +1,1180 @@
+/*
+ * 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 "libavutil/intreadwrite.h"
+#include "bitstream.h"
+#include "libavutil/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 seen in files. */
+#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))
+
+/** Number of allowed filters. */
+#define NUM_FILTERS         2
+
+/** The maximum number of taps in either the IIR or FIR filter.
+ *  I believe MLP actually specifies the maximum order for IIR filters as 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 SubStream {
+    //! Set if a valid restart header has been read. Otherwise the substream can not 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;
+
+    //! 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)
+    //@}
+
+    //@{
+    /** matrix data */
+
+    //! Number of matrices to be applied.
+    uint8_t     num_primitive_matrices;
+
+    //! matrix output channel
+    uint8_t     matrix_out_ch[MAX_MATRICES];
+
+    //! Whether the LSBs of the matrix output are encoded in the bitstream.
+    uint8_t     lsb_bypass[MAX_MATRICES];
+    //! Matrix coefficients, stored as 2.14 fixed point.
+    int32_t     matrix_coeff[MAX_MATRICES][MAX_CHANNELS+2];
+    //! Left shift to apply to noise values in 0x31eb substreams.
+    uint8_t     matrix_noise_shift[MAX_MATRICES];
+    //@}
+
+    //! Left shift to apply to huffman-decoded residuals.
+    uint8_t     quant_step_size[MAX_CHANNELS];
+
+    //! Number of PCM samples in current audio block.
+    uint16_t    blocksize;
+    //! Number of PCM samples decoded so far in this frame.
+    uint16_t    blockpos;
+
+    //! Left shift to apply to decoded PCM values to get final 24-bit output.
+    int8_t      output_shift[MAX_CHANNELS];
+
+    //! Running XOR of all output samples.
+    int32_t     lossless_check_data;
+
+} SubStream;
+
+typedef struct MLPDecodeContext {
+    AVCodecContext *avctx;
+
+    //! 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];
+
+    //@{
+    /** filter data */
+#define FIR 0
+#define IIR 1
+    //! Number of taps in filter.
+    uint8_t     filter_order[MAX_CHANNELS][NUM_FILTERS];
+    //! Right shift to apply to output of filter.
+    uint8_t     filter_shift[MAX_CHANNELS][NUM_FILTERS];
+
+    int32_t     filter_coeff[MAX_CHANNELS][NUM_FILTERS][MAX_FILTER_ORDER];
+    int32_t     filter_state[MAX_CHANNELS][NUM_FILTERS][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];
+    //@}
+
+    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 with
+ *  00 or 01, but have different codes starting with 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 int crc_init = 0;
+static AVCRC crc_63[1024];
+static AVCRC crc_1D[1024];
+
+
+/** Initialize static data, constant between all invocations of the codec. */
+
+static av_cold void init_static()
+{
+    INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
+                &huffman_tables[0][0][1], 2, 1,
+                &huffman_tables[0][0][0], 2, 1, 512);
+    INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
+                &huffman_tables[1][0][1], 2, 1,
+                &huffman_tables[1][0][0], 2, 1, 512);
+    INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
+                &huffman_tables[2][0][1], 2, 1,
+                &huffman_tables[2][0][0], 2, 1, 512);
+
+    if (!crc_init) {
+        av_crc_init(crc_63, 0,  8,   0x63, sizeof(crc_63));
+        av_crc_init(crc_1D, 0,  8,   0x1D, sizeof(crc_1D));
+        crc_init = 1;
+    }
+}
+
+
+/** 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 behavior 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 int32_t calculate_sign_huff(MLPDecodeContext *m,
+                                          unsigned int substr, unsigned int ch)
+{
+    SubStream *s = &m->substream[substr];
+    int lsb_bits = m->huff_lsbs[ch] - s->quant_step_size[ch];
+    int sign_shift = lsb_bits + (m->codebook[ch] ? 2 - m->codebook[ch] : -1);
+    int32_t sign_huff_offset = m->huff_offset[ch];
+
+    if (m->codebook[ch] > 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++) {
+        int codebook = m->codebook[channel];
+        int quant_step_size = s->quant_step_size[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 (result < 0)
+            return -1;
+
+        if (lsb_bits > 0)
+            result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
+
+        result  += m->sign_huff_offset[channel];
+        result <<= quant_step_size;
+
+        m->sample_buffer[pos + s->blockpos][channel] = result;
+    }
+
+    return 0;
+}
+
+static av_cold int mlp_decode_init(AVCodecContext *avctx)
+{
+    MLPDecodeContext *m = avctx->priv_data;
+    int substr;
+
+    init_static();
+    m->avctx = avctx;
+    for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
+        m->substream[substr].lossless_check_data = 0xffffffff;
+    return 0;
+}
+
+/** Read a major sync info header - contains high level information about
+ *  the stream - sample rate, channel arrangement etc. Most of this
+ *  information is not actually necessary for decoding, only for playback.
+ */
+
+static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
+{
+    MLPHeaderInfo mh;
+    int substr;
+
+    if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)
+        return -1;
+
+    if (mh.group1_bits == 0) {
+        av_log(m->avctx, AV_LOG_ERROR, "Invalid/unknown bits per sample\n");
+        return -1;
+    }
+    if (mh.group2_bits > mh.group1_bits) {
+        av_log(m->avctx, AV_LOG_ERROR,
+               "Channel group 2 cannot have more bits per sample than group 1\n");
+        return -1;
+    }
+
+    if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
+        av_log(m->avctx, AV_LOG_ERROR,
+               "Channel groups with differing sample rates 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;
+    for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
+        m->substream[substr].restart_seen = 0;
+
+    return 0;
+}
+
+/** Read a restart header from a block in a substream. This contains parameters
+ *  required to decode the audio that do not change very often. Generally
+ *  (always) present only in blocks following a major sync. */
+
+static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
+                               const uint8_t *buf, unsigned int substr)
+{
+    SubStream *s = &m->substream[substr];
+    unsigned int ch;
+    int sync_word, tmp;
+    uint8_t checksum;
+    uint8_t lossless_check;
+    int start_count = get_bits_count(gbp);
+
+    sync_word = get_bits(gbp, 13);
+
+    if (sync_word != 0x31ea >> 1) {
+        av_log(m->avctx, AV_LOG_ERROR,
+               "Restart header sync incorrect (got 0x%04x)\n", sync_word);
+        return -1;
+    }
+    s->noise_type = get_bits1(gbp);
+
+    skip_bits(gbp, 16); /* Output timestamp */
+
+    s->min_channel        = get_bits(gbp, 4);
+    s->max_channel        = get_bits(gbp, 4);
+    s->max_matrix_channel = get_bits(gbp, 4);
+
+    if (s->min_channel > s->max_channel) {
+        av_log(m->avctx, AV_LOG_ERROR,
+               "Substream min channel cannot be greater than max channel.\n");
+        return -1;
+    }
+
+    if (m->avctx->request_channels > 0
+        && s->max_channel + 1 >= m->avctx->request_channels
+        && substr < m->max_decoded_substream) {
+        av_log(m->avctx, AV_LOG_INFO,
+               "Extracting %d channel downmix from substream %d. "
+               "Further substreams will be skipped.\n",
+               s->max_channel + 1, substr);
+        m->max_decoded_substream = substr;
+    }
+
+    s->noise_shift   = get_bits(gbp,  4);
+    s->noisegen_seed = get_bits(gbp, 23);
+
+    skip_bits(gbp, 19);
+
+    s->data_check_present = get_bits1(gbp);
+    lossless_check = get_bits(gbp, 8);
+    if (substr == m->max_decoded_substream
+        && s->lossless_check_data != 0xffffffff) {
+        tmp = s->lossless_check_data;
+        tmp ^= tmp >> 16;
+        tmp ^= tmp >> 8;
+        tmp &= 0xff;
+        if (tmp != lossless_check)
+            av_log(m->avctx, AV_LOG_WARNING,
+                   "Lossless check failed - expected %02x, calculated %02x\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 <= s->max_matrix_channel; 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 */
+    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++) {
+        m->filter_order[ch][FIR] = 0;
+        m->filter_order[ch][IIR] = 0;
+        m->filter_shift[ch][FIR] = 0;
+        m->filter_shift[ch][IIR] = 0;
+
+        /* 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 = s->max_channel + 1;
+    }
+
+    return 0;
+}
+
+/** Read parameters for one of the prediction filters. */
+
+static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
+                              unsigned int channel, unsigned int filter)
+{
+    const char fchar = filter ? 'I' : 'F';
+    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",
+               fchar, order, MAX_FILTER_ORDER);
+        return -1;
+    }
+    m->filter_order[channel][filter] = order;
+
+    if (order > 0) {
+        int coeff_bits, coeff_shift;
+
+        m->filter_shift[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",
+                   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++)
+            m->filter_coeff[channel][filter][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++)
+                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)
+{
+    SubStream *s = &m->substream[substr];
+    unsigned int mat, ch;
+
+    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 > MAX_BLOCKSIZE) {
+                av_log(m->avctx, AV_LOG_ERROR, "Block size too large\n");
+                s->blocksize = 0;
+                return -1;
+            }
+        }
+
+    if (s->param_presence_flags & PARAM_MATRIX)
+        if (get_bits1(gbp)) {
+            s->num_primitive_matrices = get_bits(gbp, 4);
+
+            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_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;
+            }
+        }
+
+    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_bits(gbp, 4);
+                dprintf(m->avctx, "output shift[%d] = %d\n",
+                        ch, s->output_shift[ch]);
+                /* TODO: validate */
+            }
+
+    if (s->param_presence_flags & PARAM_QUANTSTEP)
+        if (get_bits1(gbp))
+            for (ch = 0; ch <= s->max_channel; ch++) {
+                s->quant_step_size[ch] = get_bits(gbp, 4);
+                /* TODO: validate */
+
+                m->sign_huff_offset[ch] = calculate_sign_huff(m, substr, ch);
+            }
+
+    for (ch = s->min_channel; ch <= s->max_channel; ch++)
+        if (get_bits1(gbp)) {
+            if (s->param_presence_flags & PARAM_FIR)
+                if (get_bits1(gbp))
+                    if (read_filter_params(m, gbp, ch, FIR) < 0)
+                        return -1;
+
+            if (s->param_presence_flags & PARAM_IIR)
+                if (get_bits1(gbp))
+                    if (read_filter_params(m, gbp, ch, IIR) < 0)
+                        return -1;
+
+            if (m->filter_order[ch][FIR] && m->filter_order[ch][IIR] &&
+                m->filter_shift[ch][FIR] != m->filter_shift[ch][IIR]) {
+                av_log(m->avctx, AV_LOG_ERROR,
+                       "FIR and IIR filters must use 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 (!m->filter_order[ch][FIR] && m->filter_order[ch][IIR])
+                m->filter_shift[ch][FIR] = m->filter_shift[ch][IIR];
+
+            if (s->param_presence_flags & PARAM_HUFFOFFSET)
+                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);
+
+            m->sign_huff_offset[ch] = calculate_sign_huff(m, substr, ch);
+
+            /* TODO: validate */
+        }
+
+    return 0;
+}
+
+#define MSB_MASK(bits)  (-1u << bits)
+
+/** Generate PCM samples using the prediction filters and residual values
+ *  read from the data stream, and update the filter state. */
+
+static void filter_channel(MLPDecodeContext *m, unsigned int substr,
+                           unsigned int channel)
+{
+    SubStream *s = &m->substream[substr];
+    int32_t filter_state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FILTER_ORDER];
+    unsigned int filter_shift = m->filter_shift[channel][FIR];
+    int32_t mask = MSB_MASK(s->quant_step_size[channel]);
+    int index = MAX_BLOCKSIZE;
+    int j, i;
+
+    for (j = 0; j < NUM_FILTERS; j++) {
+        memcpy(&   filter_state_buffer  [j][MAX_BLOCKSIZE],
+               &m->filter_state[channel][j][0],
+               MAX_FILTER_ORDER * sizeof(int32_t));
+    }
+
+    for (i = 0; i < s->blocksize; i++) {
+        int32_t residual = m->sample_buffer[i + s->blockpos][channel];
+        unsigned int order;
+        int64_t accum = 0;
+        int32_t result;
+
+        /* TODO: Move this code to DSPContext? */
+
+        for (j = 0; j < NUM_FILTERS; j++)
+            for (order = 0; order < m->filter_order[channel][j]; order++)
+                accum += (int64_t)filter_state_buffer[j][index + order] *
+                        m->filter_coeff[channel][j][order];
+
+        accum  = accum >> filter_shift;
+        result = (accum + residual) & mask;
+
+        --index;
+
+        filter_state_buffer[FIR][index] = result;
+        filter_state_buffer[IIR][index] = result - accum;
+
+        m->sample_buffer[i + s->blockpos][channel] = result;
+    }
+
+    for (j = 0; j < NUM_FILTERS; j++) {
+        memcpy(&m->filter_state[channel][j][0],
+               &   filter_state_buffer  [j][index],
+               MAX_FILTER_ORDER * sizeof(int32_t));
+    }
+}
+
+/** Read a block of PCM residual data (or actual if no filtering active). */
+
+static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
+                           unsigned int substr)
+{
+    SubStream *s = &m->substream[substr];
+    unsigned int i, ch, expected_stream_pos = 0;
+
+    if (s->data_check_present) {
+        expected_stream_pos  = get_bits_count(gbp);
+        expected_stream_pos += get_bits(gbp, 16);
+        av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
+               "we have not tested yet. %s\n", sample_message);
+    }
+
+    if (s->blockpos + s->blocksize > m->access_unit_size) {
+        av_log(m->avctx, AV_LOG_ERROR, "Too many audio samples in frame\n");
+        return -1;
+    }
+
+    memset(&m->bypassed_lsbs[s->blockpos][0], 0,
+           s->blocksize * sizeof(m->bypassed_lsbs[0]));
+
+    for (i = 0; i < s->blocksize; i++) {
+        if (read_huff_channels(m, gbp, substr, i) < 0)
+            return -1;
+    }
+
+    for (ch = s->min_channel; ch <= s->max_channel; ch++) {
+        filter_channel(m, substr, ch);
+    }
+
+    s->blockpos += s->blocksize;
+
+    if (s->data_check_present) {
+        if (get_bits_count(gbp) != expected_stream_pos)
+            av_log(m->avctx, AV_LOG_ERROR, "Block data length mismatch\n");
+        skip_bits(gbp, 8);
+    }
+
+    return 0;
+}
+
+/** Data table used for TrueHD noise generation function */
+
+static const int8_t noise_table[256] = {
+     30,  51,  22,  54,   3,   7,  -4,  38,  14,  55,  46,  81,  22,  58,  -3,   2,
+     52,  31,  -7,  51,  15,  44,  74,  30,  85, -17,  10,  33,  18,  80,  28,  62,
+     10,  32,  23,  69,  72,  26,  35,  17,  73,  60,   8,  56,   2,   6,  -2,  -5,
+     51,   4,  11,  50,  66,  76,  21,  44,  33,  47,   1,  26,  64,  48,  57,  40,
+     38,  16, -10, -28,  92,  22, -18,  29, -10,   5, -13,  49,  19,  24,  70,  34,
+     61,  48,  30,  14,  -6,  25,  58,  33,  42,  60,  67,  17,  54,  17,  22,  30,
+     67,  44,  -9,  50, -11,  43,  40,  32,  59,  82,  13,  49, -14,  55,  60,  36,
+     48,  49,  31,  47,  15,  12,   4,  65,   1,  23,  29,  39,  45,  -2,  84,  69,
+      0,  72,  37,  57,  27,  41, -15, -16,  35,  31,  14,  61,  24,   0,  27,  24,
+     16,  41,  55,  34,  53,   9,  56,  12,  25,  29,  53,   5,  20, -20,  -8,  20,
+     13,  28,  -3,  78,  38,  16,  11,  62,  46,  29,  21,  24,  46,  65,  43, -23,
+     89,  18,  74,  21,  38, -12,  19,  12, -19,   8,  15,  33,   4,  57,   9,  -8,
+     36,  35,  26,  28,   7,  83,  63,  79,  75,  11,   3,  87,  37,  47,  34,  40,
+     39,  19,  20,  42,  27,  34,  39,  77,  13,  42,  59,  64,  45,  -1,  32,  37,
+     45,  -5,  53,  -6,   7,  36,  50,  23,   6,  32,   9, -21,  18,  71,  27,  52,
+    -25,  31,  35,  42,  -1,  68,  63,  52,  26,  43,  66,  37,  41,  25,  40,  70,
+};
+
+/** Noise generation functions.
+ *  I'm not sure what these are for - they seem to be some kind of pseudorandom
+ *  sequence generators, used to generate noise data which is used when the
+ *  channels are rematrixed. I'm not sure if they provide a practical benefit
+ *  to compression, or just obfuscate the decoder. Are they for some kind of
+ *  dithering? */
+
+/** Generate two channels of noise, used in the matrix when
+ *  restart sync word == 0x31ea. */
+
+static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
+{
+    SubStream *s = &m->substream[substr];
+    unsigned int i;
+    uint32_t seed = s->noisegen_seed;
+    unsigned int maxchan = s->max_matrix_channel;
+
+    for (i = 0; i < s->blockpos; i++) {
+        uint16_t seed_shr7 = seed >> 7;
+        m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
+        m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7)   << s->noise_shift;
+
+        seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
+    }
+
+    s->noisegen_seed = seed;
+}
+
+/** Generate a block of noise, used when restart sync word == 0x31eb. */
+
+static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
+{
+    SubStream *s = &m->substream[substr];
+    unsigned int i;
+    uint32_t seed = s->noisegen_seed;
+
+    for (i = 0; i < m->access_unit_size_pow2; i++) {
+        uint8_t seed_shr15 = seed >> 15;
+        m->noise_buffer[i] = noise_table[seed_shr15];
+        seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
+    }
+
+    s->noisegen_seed = seed;
+}
+
+
+/** Apply the channel matrices in turn to reconstruct the original audio
+ *  samples. */
+
+static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
+{
+    SubStream *s = &m->substream[substr];
+    unsigned int mat, src_ch, i;
+    unsigned int maxchan;
+
+    maxchan = s->max_matrix_channel;
+    if (!s->noise_type) {
+        generate_2_noise_channels(m, substr);
+        maxchan += 2;
+    } else {
+        fill_noise_buffer(m, substr);
+    }
+
+    for (mat = 0; mat < s->num_primitive_matrices; mat++) {
+        int matrix_noise_shift = s->matrix_noise_shift[mat];
+        unsigned int dest_ch = s->matrix_out_ch[mat];
+        int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
+
+        /* TODO: DSPContext? */
+
+        for (i = 0; i < s->blockpos; i++) {
+            int64_t accum = 0;
+            for (src_ch = 0; src_ch <= maxchan; src_ch++) {
+                accum += (int64_t)m->sample_buffer[i][src_ch]
+                                  * s->matrix_coeff[mat][src_ch];
+            }
+            if (matrix_noise_shift) {
+                uint32_t index = s->num_primitive_matrices - mat;
+                index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1);
+                accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
+            }
+            m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask)
+                                             + m->bypassed_lsbs[i][mat];
+        }
+    }
+}
+
+/** Write the audio data into the output buffer. */
+
+static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
+                                uint8_t *data, unsigned int *data_size, int is32)
+{
+    SubStream *s = &m->substream[substr];
+    unsigned int i, 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 (ch = 0; ch <= s->max_channel; ch++) {
+            int32_t sample = m->sample_buffer[i][ch] << s->output_shift[ch];
+            s->lossless_check_data ^= (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;
+
+    for (; buf < buf_end; buf++)
+        scratch ^= *buf;
+
+    return scratch;
+}
+
+/** Read an access unit from the stream.
+ *  Returns < 0 on error, 0 if not enough data is present in the input stream
+ *  otherwise returns the number of bytes consumed. */
+
+static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
+                            const 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 = 4;
+    unsigned int substr_header_size = 0;
+    uint8_t substream_parity_present[MAX_SUBSTREAMS];
+    uint16_t substream_data_len[MAX_SUBSTREAMS];
+    uint8_t parity_bits;
+
+    if (buf_size < 4)
+        return 0;
+
+    length = (AV_RB16(buf) & 0xfff) * 2;
+
+    if (length > buf_size)
+        return -1;
+
+    init_get_bits(&gb, (buf + 4), (length - 4) * 8);
+
+    if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
+        dprintf(m->avctx, "Found major sync\n");
+        if (read_major_sync(m, &gb) < 0)
+            goto error;
+        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;
+
+        extraword_present = get_bits1(&gb);
+        skip_bits1(&gb);
+        checkdata_present = get_bits1(&gb);
+        skip_bits1(&gb);
+
+        end = get_bits(&gb, 12) * 2;
+
+        substr_header_size += 2;
+
+        if (extraword_present) {
+            skip_bits(&gb, 16);
+            substr_header_size += 2;
+        }
+
+        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  = calculate_parity(buf, 4);
+    parity_bits ^= 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);
+
+        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) {
+                    av_log(m->avctx, AV_LOG_ERROR,
+                           "No restart header present in substream %d.\n",
+                           substr);
+                    goto next_substr;
+                }
+
+                if (read_decoding_params(m, &gb, substr) < 0)
+                    goto next_substr;
+            }
+
+            if (!s->restart_seen) {
+                av_log(m->avctx, AV_LOG_ERROR,
+                       "No restart header present in substream %d.\n",
+                       substr);
+                goto next_substr;
+            }
+
+            if (read_block_data(m, &gb, substr) < 0)
+                return -1;
+
+        } while ((get_bits_count(&gb) < substream_data_len[substr] * 8)
+                 && get_bits1(&gb) == 0);
+
+        skip_bits(&gb, (-get_bits_count(&gb)) & 15);
+        if (substream_data_len[substr] * 8 - 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, s->blockpos);
+                s->blockpos -= 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);
+            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);
+            if (checksum != get_bits(&gb, 8))
+                av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed\n",
+                       substr);
+        }
+        if (substream_data_len[substr] * 8 != get_bits_count(&gb)) {
+            av_log(m->avctx, AV_LOG_ERROR, "Substream %d length mismatch.\n",
+                   substr);
+            return -1;
+        }
+
+next_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;
+
+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,
+    .long_name = NULL_IF_CONFIG_SMALL("Meridian Lossless Packing"),
+};
+