[FFmpeg-cvslog] r14733 - in trunk/libavcodec: Makefile mlp.c mlp.h mlp_parser.c mlpdec.c

ramiro subversion
Wed Aug 13 20:47:03 CEST 2008


Author: ramiro
Date: Wed Aug 13 20:47:03 2008
New Revision: 14733

Log:
mlp: Split common code from parser and decoder to be used by encoder.

Added:
   trunk/libavcodec/mlp.c
      - copied, changed from r14728, /trunk/libavcodec/mlpdec.c
   trunk/libavcodec/mlp.h
      - copied, changed from r14728, /trunk/libavcodec/mlpdec.c
Modified:
   trunk/libavcodec/Makefile
   trunk/libavcodec/mlp_parser.c
   trunk/libavcodec/mlpdec.c

Modified: trunk/libavcodec/Makefile
==============================================================================
--- trunk/libavcodec/Makefile	(original)
+++ trunk/libavcodec/Makefile	Wed Aug 13 20:47:03 2008
@@ -109,7 +109,7 @@ OBJS-$(CONFIG_MIMIC_DECODER)           +
 OBJS-$(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_MLP_DECODER)             += mlp.o mlpdec.o
 OBJS-$(CONFIG_MMVIDEO_DECODER)         += mmvideo.o
 OBJS-$(CONFIG_MOTIONPIXELS_DECODER)    += motionpixels.o
 OBJS-$(CONFIG_MP2_DECODER)             += mpegaudiodec.o mpegaudiodecheader.o mpegaudio.o mpegaudiodata.o
@@ -348,7 +348,7 @@ OBJS-$(CONFIG_H261_PARSER)             +
 OBJS-$(CONFIG_H263_PARSER)             += h263_parser.o
 OBJS-$(CONFIG_H264_PARSER)             += h264_parser.o
 OBJS-$(CONFIG_MJPEG_PARSER)            += mjpeg_parser.o
-OBJS-$(CONFIG_MLP_PARSER)              += mlp_parser.o
+OBJS-$(CONFIG_MLP_PARSER)              += mlp.o mlp_parser.o
 OBJS-$(CONFIG_MPEG4VIDEO_PARSER)       += mpeg4video_parser.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
 OBJS-$(CONFIG_MPEGAUDIO_PARSER)        += mpegaudio_parser.o mpegaudiodecheader.o mpegaudiodata.o
 OBJS-$(CONFIG_MPEGVIDEO_PARSER)        += mpegvideo_parser.o mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o

Copied: trunk/libavcodec/mlp.c (from r14728, /trunk/libavcodec/mlpdec.c)
==============================================================================
--- /trunk/libavcodec/mlpdec.c	(original)
+++ trunk/libavcodec/mlp.c	Wed Aug 13 20:47:03 2008
@@ -1,5 +1,5 @@
 /*
- * MLP decoder
+ * MLP codec common code
  * Copyright (c) 2007-2008 Ian Caulfield
  *
  * This file is part of FFmpeg.
@@ -19,180 +19,12 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-/**
- * @file mlpdec.c
- * MLP decoder
- */
-
 #include <stdint.h>
 
-#include "avcodec.h"
-#include "libavutil/intreadwrite.h"
-#include "bitstream.h"
 #include "libavutil/crc.h"
-#include "parser.h"
-#include "mlp_parser.h"
-
-/** 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 I haven't seen any examples with more than two. */
-#define MAX_SUBSTREAMS      2
-
-/** maximum sample frequency seen in files */
-#define MAX_SAMPLERATE      192000
-
-/** maximum number of audio samples within one access unit */
-#define MAX_BLOCKSIZE       (40 * (MAX_SAMPLERATE / 48000))
-/** 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 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;
-
-    //! 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;
-
-#define FIR 0
-#define IIR 1
-
-/** filter data */
-typedef struct {
-    uint8_t     order; ///< number of taps in filter
-    uint8_t     shift; ///< Right shift to apply to output of filter.
-
-    int32_t     coeff[MAX_FILTER_ORDER];
-    int32_t     state[MAX_FILTER_ORDER];
-} FilterParams;
-
-/** sample data coding information */
-typedef struct {
-    FilterParams filter_params[NUM_FILTERS];
-
-    int16_t     huff_offset;      ///< Offset to apply to residual values.
-    int32_t     sign_huff_offset; ///< sign/rounding-corrected version of huff_offset
-    uint8_t     codebook;         ///< Which VLC codebook to use to read residuals.
-    uint8_t     huff_lsbs;        ///< Size of residual suffix not encoded using VLC.
-} ChannelParams;
-
-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];
-
-    ChannelParams channel_params[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. */
+#include "mlp.h"
 
-static const uint8_t huffman_tables[3][18][2] = {
+const uint8_t ff_mlp_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},
@@ -208,27 +40,26 @@ static const uint8_t huffman_tables[3][1
     }
 };
 
-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 int crc_init_2D = 0;
+static AVCRC crc_2D[1024];
 
-static av_cold void init_static()
+int av_cold ff_mlp_init_crc2D(AVCodecParserContext *s)
 {
-    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_2D) {
+        av_crc_init(crc_2D, 0, 16, 0x002D, sizeof(crc_2D));
+        crc_init_2D = 1;
+    }
 
+    return 0;
+}
+
+void av_cold ff_mlp_init_crc()
+{
     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));
@@ -236,23 +67,23 @@ static av_cold void init_static()
     }
 }
 
+uint16_t ff_mlp_checksum16(const uint8_t *buf, unsigned int buf_size)
+{
+    uint16_t crc;
 
-/** MLP uses checksums that seem to be based on the standard CRC algorithm, but
- *  are 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. */
+    crc = av_crc(crc_2D, 0, buf, buf_size - 2);
+    crc ^= AV_RL16(buf + buf_size - 2);
+    return crc;
+}
 
-static uint8_t mlp_checksum8(const uint8_t *buf, unsigned int buf_size)
+uint8_t ff_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)
+uint8_t ff_mlp_restart_checksum(const uint8_t *buf, unsigned int bit_size)
 {
     int i;
     int num_bytes = (bit_size + 2) / 8;
@@ -271,714 +102,7 @@ static uint8_t mlp_restart_checksum(cons
     return crc;
 }
 
-static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
-                                          unsigned int substr, unsigned int ch)
-{
-    ChannelParams *cp = &m->channel_params[ch];
-    SubStream *s = &m->substream[substr];
-    int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
-    int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
-    int32_t sign_huff_offset = cp->huff_offset;
-
-    if (cp->codebook > 0)
-        sign_huff_offset -= 7 << lsb_bits;
-
-    if (sign_shift >= 0)
-        sign_huff_offset -= 1 << sign_shift;
-
-    return sign_huff_offset;
-}
-
-/** Read a sample, consisting of either, both or neither of entropy-coded MSBs
- *  and plain LSBs. */
-
-static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
-                                     unsigned int substr, unsigned int pos)
-{
-    SubStream *s = &m->substream[substr];
-    unsigned int mat, channel;
-
-    for (mat = 0; mat < s->num_primitive_matrices; mat++)
-        if (s->lsb_bypass[mat])
-            m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
-
-    for (channel = s->min_channel; channel <= s->max_channel; channel++) {
-        ChannelParams *cp = &m->channel_params[channel];
-        int codebook = cp->codebook;
-        int quant_step_size = s->quant_step_size[channel];
-        int lsb_bits = cp->huff_lsbs - quant_step_size;
-        int result = 0;
-
-        if (codebook > 0)
-            result = get_vlc2(gbp, huff_vlc[codebook-1].table,
-                            VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
-
-        if (result < 0)
-            return -1;
-
-        if (lsb_bits > 0)
-            result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
-
-        result  += cp->sign_huff_offset;
-        result <<= quant_step_size;
-
-        m->sample_buffer[pos + s->blockpos][channel] = result;
-    }
-
-    return 0;
-}
-
-static av_cold int mlp_decode_init(AVCodecContext *avctx)
-{
-    MLPDecodeContext *m = avctx->priv_data;
-    int substr;
-
-    init_static();
-    m->avctx = avctx;
-    for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
-        m->substream[substr].lossless_check_data = 0xffffffff;
-    avctx->sample_fmt = SAMPLE_FMT_S16;
-    return 0;
-}
-
-/** Read a major sync info header - contains high level information about
- *  the stream - sample rate, channel arrangement etc. Most of this
- *  information is not actually necessary for decoding, only for playback.
- */
-
-static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
-{
-    MLPHeaderInfo mh;
-    int substr;
-
-    if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)
-        return -1;
-
-    if (mh.group1_bits == 0) {
-        av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
-        return -1;
-    }
-    if (mh.group2_bits > mh.group1_bits) {
-        av_log(m->avctx, AV_LOG_ERROR,
-               "Channel group 2 cannot have more bits per sample than group 1.\n");
-        return -1;
-    }
-
-    if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
-        av_log(m->avctx, AV_LOG_ERROR,
-               "Channel groups with differing sample rates are not currently supported.\n");
-        return -1;
-    }
-
-    if (mh.group1_samplerate == 0) {
-        av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
-        return -1;
-    }
-    if (mh.group1_samplerate > MAX_SAMPLERATE) {
-        av_log(m->avctx, AV_LOG_ERROR,
-               "Sampling rate %d is greater than the supported maximum (%d).\n",
-               mh.group1_samplerate, MAX_SAMPLERATE);
-        return -1;
-    }
-    if (mh.access_unit_size > MAX_BLOCKSIZE) {
-        av_log(m->avctx, AV_LOG_ERROR,
-               "Block size %d is greater than the supported maximum (%d).\n",
-               mh.access_unit_size, MAX_BLOCKSIZE);
-        return -1;
-    }
-    if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
-        av_log(m->avctx, AV_LOG_ERROR,
-               "Block size pow2 %d is greater than the supported maximum (%d).\n",
-               mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
-        return -1;
-    }
-
-    if (mh.num_substreams == 0)
-        return -1;
-    if (mh.num_substreams > MAX_SUBSTREAMS) {
-        av_log(m->avctx, AV_LOG_ERROR,
-               "Number of substreams %d is larger than the maximum supported "
-               "by the decoder. %s\n", mh.num_substreams, sample_message);
-        return -1;
-    }
-
-    m->access_unit_size      = mh.access_unit_size;
-    m->access_unit_size_pow2 = mh.access_unit_size_pow2;
-
-    m->num_substreams        = mh.num_substreams;
-    m->max_decoded_substream = m->num_substreams - 1;
-
-    m->avctx->sample_rate    = mh.group1_samplerate;
-    m->avctx->frame_size     = mh.access_unit_size;
-
-#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++) {
-        ChannelParams *cp = &m->channel_params[ch];
-        cp->filter_params[FIR].order = 0;
-        cp->filter_params[IIR].order = 0;
-        cp->filter_params[FIR].shift = 0;
-        cp->filter_params[IIR].shift = 0;
-
-        /* Default audio coding is 24-bit raw PCM. */
-        cp->huff_offset      = 0;
-        cp->sign_huff_offset = (-1) << 23;
-        cp->codebook         = 0;
-        cp->huff_lsbs        = 24;
-    }
-
-    if (substr == m->max_decoded_substream) {
-        m->avctx->channels = s->max_channel + 1;
-    }
-
-    return 0;
-}
-
-/** Read parameters for one of the prediction filters. */
-
-static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
-                              unsigned int channel, unsigned int filter)
-{
-    FilterParams *fp = &m->channel_params[channel].filter_params[filter];
-    const 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;
-    }
-    fp->order = order;
-
-    if (order > 0) {
-        int coeff_bits, coeff_shift;
-
-        fp->shift = get_bits(gbp, 4);
-
-        coeff_bits  = get_bits(gbp, 5);
-        coeff_shift = get_bits(gbp, 3);
-        if (coeff_bits < 1 || coeff_bits > 16) {
-            av_log(m->avctx, AV_LOG_ERROR,
-                   "%cIR filter coeff_bits must be between 1 and 16.\n",
-                   fchar);
-            return -1;
-        }
-        if (coeff_bits + coeff_shift > 16) {
-            av_log(m->avctx, AV_LOG_ERROR,
-                   "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
-                   fchar);
-            return -1;
-        }
-
-        for (i = 0; i < order; i++)
-            fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
-
-        if (get_bits1(gbp)) {
-            int state_bits, state_shift;
-
-            if (filter == FIR) {
-                av_log(m->avctx, AV_LOG_ERROR,
-                       "FIR filter has state data specified.\n");
-                return -1;
-            }
-
-            state_bits  = get_bits(gbp, 4);
-            state_shift = get_bits(gbp, 4);
-
-            /* TODO: Check validity of state data. */
-
-            for (i = 0; i < order; i++)
-                fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
-        }
-    }
-
-    return 0;
-}
-
-/** Read 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++) {
-                ChannelParams *cp = &m->channel_params[ch];
-
-                s->quant_step_size[ch] = get_bits(gbp, 4);
-                /* TODO: validate */
-
-                cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
-            }
-
-    for (ch = s->min_channel; ch <= s->max_channel; ch++)
-        if (get_bits1(gbp)) {
-            ChannelParams *cp = &m->channel_params[ch];
-            FilterParams *fir = &cp->filter_params[FIR];
-            FilterParams *iir = &cp->filter_params[IIR];
-
-            if (s->param_presence_flags & PARAM_FIR)
-                if (get_bits1(gbp))
-                    if (read_filter_params(m, gbp, ch, FIR) < 0)
-                        return -1;
-
-            if (s->param_presence_flags & PARAM_IIR)
-                if (get_bits1(gbp))
-                    if (read_filter_params(m, gbp, ch, IIR) < 0)
-                        return -1;
-
-            if (fir->order && iir->order &&
-                fir->shift != iir->shift) {
-                av_log(m->avctx, AV_LOG_ERROR,
-                       "FIR and IIR filters must use the same precision.\n");
-                return -1;
-            }
-            /* The FIR and IIR filters must have the same precision.
-             * To simplify the filtering code, only the precision of the
-             * FIR filter is considered. If only the IIR filter is employed,
-             * the FIR filter precision is set to that of the IIR filter, so
-             * that the filtering code can use it. */
-            if (!fir->order && iir->order)
-                fir->shift = iir->shift;
-
-            if (s->param_presence_flags & PARAM_HUFFOFFSET)
-                if (get_bits1(gbp))
-                    cp->huff_offset = get_sbits(gbp, 15);
-
-            cp->codebook  = get_bits(gbp, 2);
-            cp->huff_lsbs = get_bits(gbp, 5);
-
-            cp->sign_huff_offset = 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];
-    FilterParams *fp[NUM_FILTERS] = { &m->channel_params[channel].filter_params[FIR],
-                                      &m->channel_params[channel].filter_params[IIR], };
-    unsigned int filter_shift = fp[FIR]->shift;
-    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], &fp[j]->state[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 < fp[j]->order; order++)
-                accum += (int64_t)filter_state_buffer[j][index + order] *
-                                  fp[j]->coeff[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(&fp[j]->state[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)
+uint8_t ff_mlp_calculate_parity(const uint8_t *buf, unsigned int buf_size)
 {
     uint32_t scratch = 0;
     const uint8_t *buf_end = buf + buf_size;
@@ -994,198 +118,3 @@ static uint8_t calculate_parity(const ui
 
     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) >= 32 &&
-            (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_data_len[substr] * 8 - get_bits_count(&gb) >= 16 &&
-            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"),
-};
-

Copied: trunk/libavcodec/mlp.h (from r14728, /trunk/libavcodec/mlpdec.c)
==============================================================================
--- /trunk/libavcodec/mlpdec.c	(original)
+++ trunk/libavcodec/mlp.h	Wed Aug 13 20:47:03 2008
@@ -1,5 +1,5 @@
 /*
- * MLP decoder
+ * MLP codec common header file
  * Copyright (c) 2007-2008 Ian Caulfield
  *
  * This file is part of FFmpeg.
@@ -19,19 +19,12 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-/**
- * @file mlpdec.c
- * MLP decoder
- */
+#ifndef FFMPEG_MLP_H
+#define FFMPEG_MLP_H
 
 #include <stdint.h>
 
 #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
@@ -62,83 +55,6 @@
  *  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 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;
-
-    //! 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;
-
 #define FIR 0
 #define IIR 1
 
@@ -161,1031 +77,33 @@ typedef struct {
     uint8_t     huff_lsbs;        ///< Size of residual suffix not encoded using VLC.
 } ChannelParams;
 
-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];
-
-    ChannelParams channel_params[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;
-    }
-}
-
+extern const uint8_t ff_mlp_huffman_tables[3][18][2];
 
 /** MLP uses checksums that seem to be based on the standard CRC algorithm, but
  *  are 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;
-}
+uint8_t  ff_mlp_checksum8 (const uint8_t *buf, unsigned int buf_size);
+uint16_t ff_mlp_checksum16(const uint8_t *buf, unsigned int buf_size);
 
 /** 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)
-{
-    ChannelParams *cp = &m->channel_params[ch];
-    SubStream *s = &m->substream[substr];
-    int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
-    int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
-    int32_t sign_huff_offset = cp->huff_offset;
-
-    if (cp->codebook > 0)
-        sign_huff_offset -= 7 << lsb_bits;
-
-    if (sign_shift >= 0)
-        sign_huff_offset -= 1 << sign_shift;
-
-    return sign_huff_offset;
-}
-
-/** Read a sample, consisting of either, both or neither of entropy-coded MSBs
- *  and plain LSBs. */
-
-static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
-                                     unsigned int substr, unsigned int pos)
-{
-    SubStream *s = &m->substream[substr];
-    unsigned int mat, channel;
-
-    for (mat = 0; mat < s->num_primitive_matrices; mat++)
-        if (s->lsb_bypass[mat])
-            m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
-
-    for (channel = s->min_channel; channel <= s->max_channel; channel++) {
-        ChannelParams *cp = &m->channel_params[channel];
-        int codebook = cp->codebook;
-        int quant_step_size = s->quant_step_size[channel];
-        int lsb_bits = cp->huff_lsbs - quant_step_size;
-        int result = 0;
-
-        if (codebook > 0)
-            result = get_vlc2(gbp, huff_vlc[codebook-1].table,
-                            VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
-
-        if (result < 0)
-            return -1;
-
-        if (lsb_bits > 0)
-            result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
-
-        result  += cp->sign_huff_offset;
-        result <<= quant_step_size;
-
-        m->sample_buffer[pos + s->blockpos][channel] = result;
-    }
-
-    return 0;
-}
-
-static av_cold int mlp_decode_init(AVCodecContext *avctx)
-{
-    MLPDecodeContext *m = avctx->priv_data;
-    int substr;
-
-    init_static();
-    m->avctx = avctx;
-    for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
-        m->substream[substr].lossless_check_data = 0xffffffff;
-    avctx->sample_fmt = SAMPLE_FMT_S16;
-    return 0;
-}
-
-/** Read a major sync info header - contains high level information about
- *  the stream - sample rate, channel arrangement etc. Most of this
- *  information is not actually necessary for decoding, only for playback.
- */
-
-static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
-{
-    MLPHeaderInfo mh;
-    int substr;
-
-    if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)
-        return -1;
-
-    if (mh.group1_bits == 0) {
-        av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
-        return -1;
-    }
-    if (mh.group2_bits > mh.group1_bits) {
-        av_log(m->avctx, AV_LOG_ERROR,
-               "Channel group 2 cannot have more bits per sample than group 1.\n");
-        return -1;
-    }
-
-    if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
-        av_log(m->avctx, AV_LOG_ERROR,
-               "Channel groups with differing sample rates are not currently supported.\n");
-        return -1;
-    }
-
-    if (mh.group1_samplerate == 0) {
-        av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
-        return -1;
-    }
-    if (mh.group1_samplerate > MAX_SAMPLERATE) {
-        av_log(m->avctx, AV_LOG_ERROR,
-               "Sampling rate %d is greater than the supported maximum (%d).\n",
-               mh.group1_samplerate, MAX_SAMPLERATE);
-        return -1;
-    }
-    if (mh.access_unit_size > MAX_BLOCKSIZE) {
-        av_log(m->avctx, AV_LOG_ERROR,
-               "Block size %d is greater than the supported maximum (%d).\n",
-               mh.access_unit_size, MAX_BLOCKSIZE);
-        return -1;
-    }
-    if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
-        av_log(m->avctx, AV_LOG_ERROR,
-               "Block size pow2 %d is greater than the supported maximum (%d).\n",
-               mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
-        return -1;
-    }
-
-    if (mh.num_substreams == 0)
-        return -1;
-    if (mh.num_substreams > MAX_SUBSTREAMS) {
-        av_log(m->avctx, AV_LOG_ERROR,
-               "Number of substreams %d is larger than the maximum supported "
-               "by the decoder. %s\n", mh.num_substreams, sample_message);
-        return -1;
-    }
-
-    m->access_unit_size      = mh.access_unit_size;
-    m->access_unit_size_pow2 = mh.access_unit_size_pow2;
-
-    m->num_substreams        = mh.num_substreams;
-    m->max_decoded_substream = m->num_substreams - 1;
-
-    m->avctx->sample_rate    = mh.group1_samplerate;
-    m->avctx->frame_size     = mh.access_unit_size;
-
-#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++) {
-        ChannelParams *cp = &m->channel_params[ch];
-        cp->filter_params[FIR].order = 0;
-        cp->filter_params[IIR].order = 0;
-        cp->filter_params[FIR].shift = 0;
-        cp->filter_params[IIR].shift = 0;
-
-        /* Default audio coding is 24-bit raw PCM. */
-        cp->huff_offset      = 0;
-        cp->sign_huff_offset = (-1) << 23;
-        cp->codebook         = 0;
-        cp->huff_lsbs        = 24;
-    }
-
-    if (substr == m->max_decoded_substream) {
-        m->avctx->channels = s->max_channel + 1;
-    }
-
-    return 0;
-}
-
-/** Read parameters for one of the prediction filters. */
-
-static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
-                              unsigned int channel, unsigned int filter)
-{
-    FilterParams *fp = &m->channel_params[channel].filter_params[filter];
-    const 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;
-    }
-    fp->order = order;
-
-    if (order > 0) {
-        int coeff_bits, coeff_shift;
-
-        fp->shift = get_bits(gbp, 4);
-
-        coeff_bits  = get_bits(gbp, 5);
-        coeff_shift = get_bits(gbp, 3);
-        if (coeff_bits < 1 || coeff_bits > 16) {
-            av_log(m->avctx, AV_LOG_ERROR,
-                   "%cIR filter coeff_bits must be between 1 and 16.\n",
-                   fchar);
-            return -1;
-        }
-        if (coeff_bits + coeff_shift > 16) {
-            av_log(m->avctx, AV_LOG_ERROR,
-                   "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
-                   fchar);
-            return -1;
-        }
-
-        for (i = 0; i < order; i++)
-            fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
-
-        if (get_bits1(gbp)) {
-            int state_bits, state_shift;
-
-            if (filter == FIR) {
-                av_log(m->avctx, AV_LOG_ERROR,
-                       "FIR filter has state data specified.\n");
-                return -1;
-            }
-
-            state_bits  = get_bits(gbp, 4);
-            state_shift = get_bits(gbp, 4);
-
-            /* TODO: Check validity of state data. */
-
-            for (i = 0; i < order; i++)
-                fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
-        }
-    }
-
-    return 0;
-}
-
-/** Read 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++) {
-                ChannelParams *cp = &m->channel_params[ch];
-
-                s->quant_step_size[ch] = get_bits(gbp, 4);
-                /* TODO: validate */
-
-                cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
-            }
-
-    for (ch = s->min_channel; ch <= s->max_channel; ch++)
-        if (get_bits1(gbp)) {
-            ChannelParams *cp = &m->channel_params[ch];
-            FilterParams *fir = &cp->filter_params[FIR];
-            FilterParams *iir = &cp->filter_params[IIR];
-
-            if (s->param_presence_flags & PARAM_FIR)
-                if (get_bits1(gbp))
-                    if (read_filter_params(m, gbp, ch, FIR) < 0)
-                        return -1;
-
-            if (s->param_presence_flags & PARAM_IIR)
-                if (get_bits1(gbp))
-                    if (read_filter_params(m, gbp, ch, IIR) < 0)
-                        return -1;
-
-            if (fir->order && iir->order &&
-                fir->shift != iir->shift) {
-                av_log(m->avctx, AV_LOG_ERROR,
-                       "FIR and IIR filters must use the same precision.\n");
-                return -1;
-            }
-            /* The FIR and IIR filters must have the same precision.
-             * To simplify the filtering code, only the precision of the
-             * FIR filter is considered. If only the IIR filter is employed,
-             * the FIR filter precision is set to that of the IIR filter, so
-             * that the filtering code can use it. */
-            if (!fir->order && iir->order)
-                fir->shift = iir->shift;
-
-            if (s->param_presence_flags & PARAM_HUFFOFFSET)
-                if (get_bits1(gbp))
-                    cp->huff_offset = get_sbits(gbp, 15);
-
-            cp->codebook  = get_bits(gbp, 2);
-            cp->huff_lsbs = get_bits(gbp, 5);
-
-            cp->sign_huff_offset = 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];
-    FilterParams *fp[NUM_FILTERS] = { &m->channel_params[channel].filter_params[FIR],
-                                      &m->channel_params[channel].filter_params[IIR], };
-    unsigned int filter_shift = fp[FIR]->shift;
-    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], &fp[j]->state[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 < fp[j]->order; order++)
-                accum += (int64_t)filter_state_buffer[j][index + order] *
-                                  fp[j]->coeff[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(&fp[j]->state[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);
-}
-
+uint8_t ff_mlp_restart_checksum(const uint8_t *buf, unsigned int bit_size);
 
 /** 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) >= 32 &&
-            (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_data_len[substr] * 8 - get_bits_count(&gb) >= 16 &&
-            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;
+uint8_t ff_mlp_calculate_parity(const uint8_t *buf, unsigned int buf_size);
 
-error:
-    m->params_valid = 0;
-    return -1;
-}
+int ff_mlp_init_crc2D(AVCodecParserContext *s);
 
-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"),
-};
+void ff_mlp_init_crc();
 
+#endif /* FFMPEG_MLP_H */

Modified: trunk/libavcodec/mlp_parser.c
==============================================================================
--- trunk/libavcodec/mlp_parser.c	(original)
+++ trunk/libavcodec/mlp_parser.c	Wed Aug 13 20:47:03 2008
@@ -30,6 +30,7 @@
 #include "bitstream.h"
 #include "parser.h"
 #include "mlp_parser.h"
+#include "mlp.h"
 
 static const uint8_t mlp_quants[16] = {
     16, 20, 24, 0, 0, 0, 0, 0,
@@ -64,34 +65,6 @@ static int truehd_channels(int chanmap)
     return channels;
 }
 
-static int crc_init = 0;
-static AVCRC crc_2D[1024];
-
-/** MLP uses checksums that seem to be based on the standard CRC algorithm, but
- *  are 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 uint16_t mlp_checksum16(const uint8_t *buf, unsigned int buf_size)
-{
-    uint16_t crc;
-
-    crc = av_crc(crc_2D, 0, buf, buf_size - 2);
-    crc ^= AV_RL16(buf + buf_size - 2);
-    return crc;
-}
-
-static int av_cold mlp_parse_init(AVCodecParserContext *s)
-{
-    if (!crc_init) {
-        av_crc_init(crc_2D, 0, 16, 0x002D, sizeof(crc_2D));
-        crc_init = 1;
-    }
-
-    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.
@@ -110,7 +83,7 @@ int ff_mlp_read_major_sync(void *log, ML
         return -1;
     }
 
-    checksum = mlp_checksum16(gb->buffer, 26);
+    checksum = ff_mlp_checksum16(gb->buffer, 26);
     if (checksum != AV_RL16(gb->buffer+26)) {
         av_log(log, AV_LOG_ERROR, "major sync info header checksum error\n");
         return -1;
@@ -310,7 +283,7 @@ lost_sync:
 AVCodecParser mlp_parser = {
     { CODEC_ID_MLP },
     sizeof(MLPParseContext),
-    mlp_parse_init,
+    ff_mlp_init_crc2D,
     mlp_parse,
     NULL,
 };

Modified: trunk/libavcodec/mlpdec.c
==============================================================================
--- trunk/libavcodec/mlpdec.c	(original)
+++ trunk/libavcodec/mlpdec.c	Wed Aug 13 20:47:03 2008
@@ -32,35 +32,7 @@
 #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 I haven't seen any examples with more than two. */
-#define MAX_SUBSTREAMS      2
-
-/** maximum sample frequency seen in files */
-#define MAX_SAMPLERATE      192000
-
-/** maximum number of audio samples within one access unit */
-#define MAX_BLOCKSIZE       (40 * (MAX_SAMPLERATE / 48000))
-/** 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
+#include "mlp.h"
 
 /** number of bits used for VLC lookup - longest Huffman code is 9 */
 #define VLC_BITS            9
@@ -139,28 +111,6 @@ typedef struct SubStream {
 
 } SubStream;
 
-#define FIR 0
-#define IIR 1
-
-/** filter data */
-typedef struct {
-    uint8_t     order; ///< number of taps in filter
-    uint8_t     shift; ///< Right shift to apply to output of filter.
-
-    int32_t     coeff[MAX_FILTER_ORDER];
-    int32_t     state[MAX_FILTER_ORDER];
-} FilterParams;
-
-/** sample data coding information */
-typedef struct {
-    FilterParams filter_params[NUM_FILTERS];
-
-    int16_t     huff_offset;      ///< Offset to apply to residual values.
-    int32_t     sign_huff_offset; ///< sign/rounding-corrected version of huff_offset
-    uint8_t     codebook;         ///< Which VLC codebook to use to read residuals.
-    uint8_t     huff_lsbs;        ///< Size of residual suffix not encoded using VLC.
-} ChannelParams;
-
 typedef struct MLPDecodeContext {
     AVCodecContext *avctx;
 
@@ -187,88 +137,23 @@ typedef struct MLPDecodeContext {
     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);
+                &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,
-                &huffman_tables[1][0][1], 2, 1,
-                &huffman_tables[1][0][0], 2, 1, 512);
+                &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,
-                &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
- *  are 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;
-    }
+                &ff_mlp_huffman_tables[2][0][1], 2, 1,
+                &ff_mlp_huffman_tables[2][0][0], 2, 1, 512);
 
-    return crc;
+    ff_mlp_init_crc();
 }
 
 static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
@@ -506,7 +391,7 @@ static int read_restart_header(MLPDecode
         }
     }
 
-    checksum = mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
+    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");
@@ -975,26 +860,6 @@ static int output_data(MLPDecodeContext 
 }
 
 
-/** 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. */
@@ -1078,8 +943,8 @@ static int read_access_unit(AVCodecConte
         substream_start = end;
     }
 
-    parity_bits  = calculate_parity(buf, 4);
-    parity_bits ^= calculate_parity(buf + header_size, substr_header_size);
+    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");
@@ -1145,12 +1010,12 @@ static int read_access_unit(AVCodecConte
             substream_parity_present[substr]) {
             uint8_t parity, checksum;
 
-            parity = calculate_parity(buf, substream_data_len[substr] - 2);
+            parity = ff_mlp_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);
+            checksum = ff_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);




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