mlpdec.c

ffmpeg移植到symbian的全部源代码

        return -1;
    }

    m->access_unit_size      = mh.access_unit_size;
    m->access_unit_size_pow2 = mh.access_unit_size_pow2;

    m->num_substreams        = mh.num_substreams;
    m->max_decoded_substream = m->num_substreams - 1;

    m->avctx->sample_rate    = mh.group1_samplerate;
    m->avctx->frame_size     = mh.access_unit_size;

#ifdef CONFIG_AUDIO_NONSHORT
    m->avctx->bits_per_sample = mh.group1_bits;
    if (mh.group1_bits > 16) {
        m->avctx->sample_fmt = SAMPLE_FMT_S32;
    }
#endif

    m->params_valid = 1;
    for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
        m->substream[substr].restart_seen = 0;

    return 0;
}

/** Read a restart header from a block in a substream. This contains parameters
 *  required to decode the audio that do not change very often. Generally
 *  (always) present only in blocks following a major sync. */

static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
                               const uint8_t *buf, unsigned int substr)
{
    SubStream *s = &m->substream[substr];
    unsigned int ch;
    int sync_word, tmp;
    uint8_t checksum;
    uint8_t lossless_check;
    int start_count = get_bits_count(gbp);

    sync_word = get_bits(gbp, 13);

    if (sync_word != 0x31ea >> 1) {
        av_log(m->avctx, AV_LOG_ERROR,
               "restart header sync incorrect (got 0x%04x)\n", sync_word);
        return -1;
    }
    s->noise_type = get_bits1(gbp);

    skip_bits(gbp, 16); /* Output timestamp */

    s->min_channel        = get_bits(gbp, 4);
    s->max_channel        = get_bits(gbp, 4);
    s->max_matrix_channel = get_bits(gbp, 4);

    if (s->min_channel > s->max_channel) {
        av_log(m->avctx, AV_LOG_ERROR,
               "Substream min channel cannot be greater than max channel.\n");
        return -1;
    }

    if (m->avctx->request_channels > 0
        && s->max_channel + 1 >= m->avctx->request_channels
        && substr < m->max_decoded_substream) {
        av_log(m->avctx, AV_LOG_INFO,
               "Extracting %d channel downmix from substream %d. "
               "Further substreams will be skipped.\n",
               s->max_channel + 1, substr);
        m->max_decoded_substream = substr;
    }

    s->noise_shift   = get_bits(gbp,  4);
    s->noisegen_seed = get_bits(gbp, 23);

    skip_bits(gbp, 19);

    s->data_check_present = get_bits1(gbp);
    lossless_check = get_bits(gbp, 8);
    if (substr == m->max_decoded_substream
        && s->lossless_check_data != 0xffffffff) {
        tmp = s->lossless_check_data;
        tmp ^= tmp >> 16;
        tmp ^= tmp >> 8;
        tmp &= 0xff;
        if (tmp != lossless_check)
            av_log(m->avctx, AV_LOG_WARNING,
                   "Lossless check failed - expected %02x, calculated %02x.\n",
                   lossless_check, tmp);
        else
            dprintf(m->avctx, "Lossless check passed for substream %d (%x).\n",
                    substr, tmp);
    }

    skip_bits(gbp, 16);

    for (ch = 0; ch <= s->max_matrix_channel; ch++) {
        int ch_assign = get_bits(gbp, 6);
        dprintf(m->avctx, "ch_assign[%d][%d] = %d\n", substr, ch,
                ch_assign);
        if (ch_assign != ch) {
            av_log(m->avctx, AV_LOG_ERROR,
                   "Non-1:1 channel assignments are used in this stream. %s\n",
                   sample_message);
            return -1;
        }
    }

    checksum = mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);

    if (checksum != get_bits(gbp, 8))
        av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");

    /* Set default decoding parameters. */
    s->param_presence_flags   = 0xff;
    s->num_primitive_matrices = 0;
    s->blocksize              = 8;
    s->lossless_check_data    = 0;

    memset(s->output_shift   , 0, sizeof(s->output_shift   ));
    memset(s->quant_step_size, 0, sizeof(s->quant_step_size));

    for (ch = s->min_channel; ch <= s->max_channel; ch++) {
        m->filter_order[ch][FIR] = 0;
        m->filter_order[ch][IIR] = 0;
        m->filter_shift[ch][FIR] = 0;
        m->filter_shift[ch][IIR] = 0;

        /* Default audio coding is 24-bit raw PCM. */
        m->huff_offset     [ch] = 0;
        m->sign_huff_offset[ch] = (-1) << 23;
        m->codebook        [ch] = 0;
        m->huff_lsbs       [ch] = 24;
    }

    if (substr == m->max_decoded_substream) {
        m->avctx->channels = s->max_channel + 1;
    }

    return 0;
}

/** Read parameters for one of the prediction filters. */

static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
                              unsigned int channel, unsigned int filter)
{
    const char fchar = filter ? 'I' : 'F';
    int i, order;

    // Filter is 0 for FIR, 1 for IIR.
    assert(filter < 2);

    order = get_bits(gbp, 4);
    if (order > MAX_FILTER_ORDER) {
        av_log(m->avctx, AV_LOG_ERROR,
               "%cIR filter order %d is greater than maximum %d.\n",
               fchar, order, MAX_FILTER_ORDER);
        return -1;
    }
    m->filter_order[channel][filter] = order;

    if (order > 0) {
        int coeff_bits, coeff_shift;

        m->filter_shift[channel][filter] = get_bits(gbp, 4);

        coeff_bits  = get_bits(gbp, 5);
        coeff_shift = get_bits(gbp, 3);
        if (coeff_bits < 1 || coeff_bits > 16) {
            av_log(m->avctx, AV_LOG_ERROR,
                   "%cIR filter coeff_bits must be between 1 and 16.\n",
                   fchar);
            return -1;
        }
        if (coeff_bits + coeff_shift > 16) {
            av_log(m->avctx, AV_LOG_ERROR,
                   "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
                   fchar);
            return -1;
        }

        for (i = 0; i < order; i++)
            m->filter_coeff[channel][filter][i] =
                    get_sbits(gbp, coeff_bits) << coeff_shift;

        if (get_bits1(gbp)) {
            int state_bits, state_shift;

            if (filter == FIR) {
                av_log(m->avctx, AV_LOG_ERROR,
                       "FIR filter has state data specified.\n");
                return -1;
            }

            state_bits  = get_bits(gbp, 4);
            state_shift = get_bits(gbp, 4);

            /* TODO: Check validity of state data. */

            for (i = 0; i < order; i++)
                m->filter_state[channel][filter][i] =
                    get_sbits(gbp, state_bits) << state_shift;
        }
    }

    return 0;
}

/** Read decoding parameters that change more often than those in the restart
 *  header. */

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

    if (get_bits1(gbp))
        s->param_presence_flags = get_bits(gbp, 8);

    if (s->param_presence_flags & PARAM_BLOCKSIZE)
        if (get_bits1(gbp)) {
            s->blocksize = get_bits(gbp, 9);
            if (s->blocksize > MAX_BLOCKSIZE) {
                av_log(m->avctx, AV_LOG_ERROR, "block size too large\n");
                s->blocksize = 0;
                return -1;
            }
        }

    if (s->param_presence_flags & PARAM_MATRIX)
        if (get_bits1(gbp)) {
            s->num_primitive_matrices = get_bits(gbp, 4);

            for (mat = 0; mat < s->num_primitive_matrices; mat++) {
                int frac_bits, max_chan;
                s->matrix_out_ch[mat] = get_bits(gbp, 4);
                frac_bits             = get_bits(gbp, 4);
                s->lsb_bypass   [mat] = get_bits1(gbp);

                if (s->matrix_out_ch[mat] > s->max_channel) {
                    av_log(m->avctx, AV_LOG_ERROR,
                           "Invalid channel %d specified as output from matrix.\n",
                           s->matrix_out_ch[mat]);
                    return -1;
                }
                if (frac_bits > 14) {
                    av_log(m->avctx, AV_LOG_ERROR,
                           "Too many fractional bits specified.\n");
                    return -1;
                }

                max_chan = s->max_matrix_channel;
                if (!s->noise_type)
                    max_chan+=2;

                for (ch = 0; ch <= max_chan; ch++) {
                    int coeff_val = 0;
                    if (get_bits1(gbp))
                        coeff_val = get_sbits(gbp, frac_bits + 2);

                    s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
                }

                if (s->noise_type)
                    s->matrix_noise_shift[mat] = get_bits(gbp, 4);
                else
                    s->matrix_noise_shift[mat] = 0;
            }
        }

    if (s->param_presence_flags & PARAM_OUTSHIFT)
        if (get_bits1(gbp))
            for (ch = 0; ch <= s->max_matrix_channel; ch++) {
                s->output_shift[ch] = get_bits(gbp, 4);
                dprintf(m->avctx, "output shift[%d] = %d\n",
                        ch, s->output_shift[ch]);
                /* TODO: validate */
            }

    if (s->param_presence_flags & PARAM_QUANTSTEP)
        if (get_bits1(gbp))
            for (ch = 0; ch <= s->max_channel; ch++) {
                s->quant_step_size[ch] = get_bits(gbp, 4);
                /* TODO: validate */

                m->sign_huff_offset[ch] = calculate_sign_huff(m, substr, ch);
            }

    for (ch = s->min_channel; ch <= s->max_channel; ch++)
        if (get_bits1(gbp)) {
            if (s->param_presence_flags & PARAM_FIR)
                if (get_bits1(gbp))
                    if (read_filter_params(m, gbp, ch, FIR) < 0)
                        return -1;

            if (s->param_presence_flags & PARAM_IIR)
                if (get_bits1(gbp))
                    if (read_filter_params(m, gbp, ch, IIR) < 0)
                        return -1;

            if (m->filter_order[ch][FIR] && m->filter_order[ch][IIR] &&
                m->filter_shift[ch][FIR] != m->filter_shift[ch][IIR]) {
                av_log(m->avctx, AV_LOG_ERROR,
                       "FIR and IIR filters must use 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 (!m->filter_order[ch][FIR] && m->filter_order[ch][IIR])
                m->filter_shift[ch][FIR] = m->filter_shift[ch][IIR];

            if (s->param_presence_flags & PARAM_HUFFOFFSET)
                if (get_bits1(gbp))
                    m->huff_offset[ch] = get_sbits(gbp, 15);

            m->codebook [ch] = get_bits(gbp, 2);
            m->huff_lsbs[ch] = get_bits(gbp, 5);

            m->sign_huff_offset[ch] = calculate_sign_huff(m, substr, ch);

            /* TODO: validate */
        }

    return 0;
}

#define MSB_MASK(bits)  (-1u << bits)

/** Generate PCM samples using the prediction filters and residual values
 *  read from the data stream, and update the filter state. */

static void filter_channel(MLPDecodeContext *m, unsigned int substr,
                           unsigned int channel)
{
    SubStream *s = &m->substream[substr];
    int32_t filter_state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FILTER_ORDER];
    unsigned int filter_shift = m->filter_shift[channel][FIR];
    int32_t mask = MSB_MASK(s->quant_step_size[channel]);
    int index = MAX_BLOCKSIZE;
    int j, i;

    for (j = 0; j < NUM_FILTERS; j++) {
        memcpy(&   filter_state_buffer  [j][MAX_BLOCKSIZE],
               &m->filter_state[channel][j][0],
               MAX_FILTER_ORDER * sizeof(int32_t));
    }

    for (i = 0; i < s->blocksize; i++) {
        int32_t residual = m->sample_buffer[i + s->blockpos][channel];
        unsigned int order;
        int64_t accum = 0;
        int32_t result;

        /* TODO: Move this code to DSPContext? */

        for (j = 0; j < NUM_FILTERS; j++)
            for (order = 0; order < m->filter_order[channel][j]; order++)
                accum += (int64_t)filter_state_buffer[j][index + order] *
                        m->filter_coeff[channel][j][order];

        accum  = accum >> filter_shift;
        result = (accum + residual) & mask;

        --index;

        filter_state_buffer[FIR][index] = result;
        filter_state_buffer[IIR][index] = result - accum;

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

    for (j = 0; j < NUM_FILTERS; j++) {
        memcpy(&m->filter_state[channel][j][0],
               &   filter_state_buffer  [j][index],
               MAX_FILTER_ORDER * sizeof(int32_t));
    }
}

/** Read a block of PCM residual data (or actual if no filtering active). */

static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
                           unsigned int substr)
{
    SubStream *s = &m->substream[substr];
    unsigned int i, ch, expected_stream_pos = 0;

    if (s->data_check_present) {
        expected_stream_pos  = get_bits_count(gbp);
        expected_stream_pos += get_bits(gbp, 16);
        av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
               "we have not tested yet. %s\n", sample_message);
    }

    if (s->blockpos + s->blocksize > m->access_unit_size) {
        av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");