mlpdec.c

ffmpeg移植到symbian的全部源代码

/*
 * MLP decoder
 * Copyright (c) 2007-2008 Ian Caulfield
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

/**
 * @file mlpdec.c
 * MLP decoder
 */

#include <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;

typedef struct MLPDecodeContext {
    AVCodecContext *avctx;

    //! Set if a valid major sync block has been read. Otherwise no decoding is possible.
    uint8_t     params_valid;

    //! Number of substreams contained within this stream.
    uint8_t     num_substreams;

    //! Index of the last substream to decode - further substreams are skipped.
    uint8_t     max_decoded_substream;

    //! number of PCM samples contained in each frame
    int         access_unit_size;
    //! next power of two above the number of samples in each frame
    int         access_unit_size_pow2;

    SubStream   substream[MAX_SUBSTREAMS];

    //@{
    /** filter data */
#define FIR 0
#define IIR 1
    //! number of taps in filter
    uint8_t     filter_order[MAX_CHANNELS][NUM_FILTERS];
    //! Right shift to apply to output of filter.
    uint8_t     filter_shift[MAX_CHANNELS][NUM_FILTERS];

    int32_t     filter_coeff[MAX_CHANNELS][NUM_FILTERS][MAX_FILTER_ORDER];
    int32_t     filter_state[MAX_CHANNELS][NUM_FILTERS][MAX_FILTER_ORDER];
    //@}

    //@{
    /** sample data coding information */
    //! Offset to apply to residual values.
    int16_t     huff_offset[MAX_CHANNELS];
    //! sign/rounding-corrected version of huff_offset
    int32_t     sign_huff_offset[MAX_CHANNELS];
    //! Which VLC codebook to use to read residuals.
    uint8_t     codebook[MAX_CHANNELS];
    //! Size of residual suffix not encoded using VLC.
    uint8_t     huff_lsbs[MAX_CHANNELS];
    //@}

    int8_t      noise_buffer[MAX_BLOCKSIZE_POW2];
    int8_t      bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
    int32_t     sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS+2];
} MLPDecodeContext;

/** Tables defining the Huffman codes.
 *  There are three entropy coding methods used in MLP (four if you count
 *  "none" as a method). These use the same sequences for codes starting with
 *  00 or 01, but have different codes starting with 1. */

static const uint8_t huffman_tables[3][18][2] = {
    {    /* Huffman table 0, -7 - +10 */
        {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},
        {0x04, 3}, {0x05, 3}, {0x06, 3}, {0x07, 3},
        {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},
    }, { /* Huffman table 1, -7 - +8 */
        {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},
        {0x02, 2}, {0x03, 2},
        {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},
    }, { /* Huffman table 2, -7 - +7 */
        {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},
        {0x01, 1},
        {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},
    }
};

static VLC huff_vlc[3];

static int crc_init = 0;
static AVCRC crc_63[1024];
static AVCRC crc_1D[1024];


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

static av_cold void init_static()
{
    INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
                &huffman_tables[0][0][1], 2, 1,
                &huffman_tables[0][0][0], 2, 1, 512);
    INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
                &huffman_tables[1][0][1], 2, 1,
                &huffman_tables[1][0][0], 2, 1, 512);
    INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
                &huffman_tables[2][0][1], 2, 1,
                &huffman_tables[2][0][0], 2, 1, 512);

    if (!crc_init) {
        av_crc_init(crc_63, 0,  8,   0x63, sizeof(crc_63));
        av_crc_init(crc_1D, 0,  8,   0x1D, sizeof(crc_1D));
        crc_init = 1;
    }
}


/** MLP uses checksums that seem to be based on the standard CRC algorithm, but
 *  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;
    }

    return crc;
}

static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
                                          unsigned int substr, unsigned int ch)
{
    SubStream *s = &m->substream[substr];
    int lsb_bits = m->huff_lsbs[ch] - s->quant_step_size[ch];
    int sign_shift = lsb_bits + (m->codebook[ch] ? 2 - m->codebook[ch] : -1);
    int32_t sign_huff_offset = m->huff_offset[ch];

    if (m->codebook[ch] > 0)
        sign_huff_offset -= 7 << lsb_bits;

    if (sign_shift >= 0)
        sign_huff_offset -= 1 << sign_shift;

    return sign_huff_offset;
}

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

static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
                                     unsigned int substr, unsigned int pos)
{
    SubStream *s = &m->substream[substr];
    unsigned int mat, channel;

    for (mat = 0; mat < s->num_primitive_matrices; mat++)
        if (s->lsb_bypass[mat])
            m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);

    for (channel = s->min_channel; channel <= s->max_channel; channel++) {
        int codebook = m->codebook[channel];
        int quant_step_size = s->quant_step_size[channel];
        int lsb_bits = m->huff_lsbs[channel] - quant_step_size;
        int result = 0;

        if (codebook > 0)
            result = get_vlc2(gbp, huff_vlc[codebook-1].table,
                            VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);

        if (result < 0)
            return -1;

        if (lsb_bits > 0)
            result = (result << lsb_bits) + get_bits(gbp, lsb_bits);

        result  += m->sign_huff_offset[channel];
        result <<= quant_step_size;

        m->sample_buffer[pos + s->blockpos][channel] = result;
    }

    return 0;
}

static av_cold int mlp_decode_init(AVCodecContext *avctx)
{
    MLPDecodeContext *m = avctx->priv_data;
    int substr;

    init_static();
    m->avctx = avctx;
    for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
        m->substream[substr].lossless_check_data = 0xffffffff;
    return 0;
}

/** Read a major sync info header - contains high level information about
 *  the stream - sample rate, channel arrangement etc. Most of this
 *  information is not actually necessary for decoding, only for playback.
 */

static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
{
    MLPHeaderInfo mh;
    int substr;

    if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)
        return -1;

    if (mh.group1_bits == 0) {
        av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
        return -1;
    }
    if (mh.group2_bits > mh.group1_bits) {
        av_log(m->avctx, AV_LOG_ERROR,
               "Channel group 2 cannot have more bits per sample than group 1.\n");
        return -1;
    }

    if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
        av_log(m->avctx, AV_LOG_ERROR,
               "Channel groups with differing sample rates 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);