qdm2.c

ffmpeg的完整源代码和作者自己写的文档。不但有在Linux的工程哦

    value = get_vlc2(gb, vlc->table, vlc->bits, depth);

    /* stage-2, 3 bits exponent escape sequence */
    if (value-- == 0)
        value = get_bits (gb, get_bits (gb, 3) + 1);

    /* stage-3, optional */
    if (flag) {
        int tmp = vlc_stage3_values[value];

        if ((value & ~3) > 0)
            tmp += get_bits (gb, (value >> 2));
        value = tmp;
    }

    return value;
}


static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
{
    int value = qdm2_get_vlc (gb, vlc, 0, depth);

    return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
}


/**
 * QDM2 checksum
 *
 * @param data      pointer to data to be checksum'ed
 * @param length    data length
 * @param value     checksum value
 *
 * @return          0 if checksum is OK
 */
static uint16_t qdm2_packet_checksum (uint8_t *data, int length, int value) {
    int i;

    for (i=0; i < length; i++)
        value -= data[i];

    return (uint16_t)(value & 0xffff);
}


/**
 * Fills a QDM2SubPacket structure with packet type, size, and data pointer.
 *
 * @param gb            bitreader context
 * @param sub_packet    packet under analysis
 */
static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
{
    sub_packet->type = get_bits (gb, 8);

    if (sub_packet->type == 0) {
        sub_packet->size = 0;
        sub_packet->data = NULL;
    } else {
        sub_packet->size = get_bits (gb, 8);

      if (sub_packet->type & 0x80) {
          sub_packet->size <<= 8;
          sub_packet->size  |= get_bits (gb, 8);
          sub_packet->type  &= 0x7f;
      }

      if (sub_packet->type == 0x7f)
          sub_packet->type |= (get_bits (gb, 8) << 8);

      sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
    }

    av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
        sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
}


/**
 * Return node pointer to first packet of requested type in list.
 *
 * @param list    list of subpackets to be scanned
 * @param type    type of searched subpacket
 * @return        node pointer for subpacket if found, else NULL
 */
static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
{
    while (list != NULL && list->packet != NULL) {
        if (list->packet->type == type)
            return list;
        list = list->next;
    }
    return NULL;
}


/**
 * Replaces 8 elements with their average value.
 * Called by qdm2_decode_superblock before starting subblock decoding.
 *
 * @param q       context
 */
static void average_quantized_coeffs (QDM2Context *q)
{
    int i, j, n, ch, sum;

    n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;

    for (ch = 0; ch < q->nb_channels; ch++)
        for (i = 0; i < n; i++) {
            sum = 0;

            for (j = 0; j < 8; j++)
                sum += q->quantized_coeffs[ch][i][j];

            sum /= 8;
            if (sum > 0)
                sum--;

            for (j=0; j < 8; j++)
                q->quantized_coeffs[ch][i][j] = sum;
        }
}


/**
 * Build subband samples with noise weighted by q->tone_level.
 * Called by synthfilt_build_sb_samples.
 *
 * @param q     context
 * @param sb    subband index
 */
static void build_sb_samples_from_noise (QDM2Context *q, int sb)
{
    int ch, j;

    FIX_NOISE_IDX(q->noise_idx);

    if (!q->nb_channels)
        return;

    for (ch = 0; ch < q->nb_channels; ch++)
        for (j = 0; j < 64; j++) {
            q->sb_samples[ch][j * 2][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
            q->sb_samples[ch][j * 2 + 1][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
        }
}


/**
 * Called while processing data from subpackets 11 and 12.
 * Used after making changes to coding_method array.
 *
 * @param sb               subband index
 * @param channels         number of channels
 * @param coding_method    q->coding_method[0][0][0]
 */
static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
{
    int j,k;
    int ch;
    int run, case_val;
    int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4};

    for (ch = 0; ch < channels; ch++) {
        for (j = 0; j < 64; ) {
            if((coding_method[ch][sb][j] - 8) > 22) {
                run = 1;
                case_val = 8;
            } else {
                switch (switchtable[coding_method[ch][sb][j]-8]) {
                    case 0: run = 10; case_val = 10; break;
                    case 1: run = 1; case_val = 16; break;
                    case 2: run = 5; case_val = 24; break;
                    case 3: run = 3; case_val = 30; break;
                    case 4: run = 1; case_val = 30; break;
                    case 5: run = 1; case_val = 8; break;
                    default: run = 1; case_val = 8; break;
                }
            }
            for (k = 0; k < run; k++)
                if (j + k < 128)
                    if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
                        if (k > 0) {
                           SAMPLES_NEEDED
                            //not debugged, almost never used
                            memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
                            memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
                        }
            j += run;
        }
    }
}


/**
 * Related to synthesis filter
 * Called by process_subpacket_10
 *
 * @param q       context
 * @param flag    1 if called after getting data from subpacket 10, 0 if no subpacket 10
 */
static void fill_tone_level_array (QDM2Context *q, int flag)
{
    int i, sb, ch, sb_used;
    int tmp, tab;

    // This should never happen
    if (q->nb_channels <= 0)
        return;

    for (ch = 0; ch < q->nb_channels; ch++)
        for (sb = 0; sb < 30; sb++)
            for (i = 0; i < 8; i++) {
                if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
                    tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
                          q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
                else
                    tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
                if(tmp < 0)
                    tmp += 0xff;
                q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
            }

    sb_used = QDM2_SB_USED(q->sub_sampling);

    if ((q->superblocktype_2_3 != 0) && !flag) {
        for (sb = 0; sb < sb_used; sb++)
            for (ch = 0; ch < q->nb_channels; ch++)
                for (i = 0; i < 64; i++) {
                    q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
                    if (q->tone_level_idx[ch][sb][i] < 0)
                        q->tone_level[ch][sb][i] = 0;
                    else
                        q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
                }
    } else {
        tab = q->superblocktype_2_3 ? 0 : 1;
        for (sb = 0; sb < sb_used; sb++) {
            if ((sb >= 4) && (sb <= 23)) {
                for (ch = 0; ch < q->nb_channels; ch++)
                    for (i = 0; i < 64; i++) {
                        tmp = q->tone_level_idx_base[ch][sb][i / 8] -
                              q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
                              q->tone_level_idx_mid[ch][sb - 4][i / 8] -
                              q->tone_level_idx_hi2[ch][sb - 4];
                        q->tone_level_idx[ch][sb][i] = tmp & 0xff;
                        if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
                            q->tone_level[ch][sb][i] = 0;
                        else
                            q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
                }
            } else {
                if (sb > 4) {
                    for (ch = 0; ch < q->nb_channels; ch++)
                        for (i = 0; i < 64; i++) {
                            tmp = q->tone_level_idx_base[ch][sb][i / 8] -
                                  q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
                                  q->tone_level_idx_hi2[ch][sb - 4];
                            q->tone_level_idx[ch][sb][i] = tmp & 0xff;
                            if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
                                q->tone_level[ch][sb][i] = 0;
                            else
                                q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
                    }
                } else {
                    for (ch = 0; ch < q->nb_channels; ch++)
                        for (i = 0; i < 64; i++) {
                            tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
                            if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
                                q->tone_level[ch][sb][i] = 0;
                            else
                                q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
                        }
                }
            }
        }
    }

    return;
}


/**
 * Related to synthesis filter
 * Called by process_subpacket_11
 * c is built with data from subpacket 11
 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
 *
 * @param tone_level_idx
 * @param tone_level_idx_temp
 * @param coding_method        q->coding_method[0][0][0]
 * @param nb_channels          number of channels
 * @param c                    coming from subpacket 11, passed as 8*c
 * @param superblocktype_2_3   flag based on superblock packet type
 * @param cm_table_select      q->cm_table_select
 */
static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
                sb_int8_array coding_method, int nb_channels,
                int c, int superblocktype_2_3, int cm_table_select)
{
    int ch, sb, j;
    int tmp, acc, esp_40, comp;
    int add1, add2, add3, add4;
    int64_t multres;

    // This should never happen
    if (nb_channels <= 0)
        return;

    if (!superblocktype_2_3) {
        /* This case is untested, no samples available */
        SAMPLES_NEEDED
        for (ch = 0; ch < nb_channels; ch++)
            for (sb = 0; sb < 30; sb++) {
                for (j = 1; j < 64; j++) {
                    add1 = tone_level_idx[ch][sb][j] - 10;
                    if (add1 < 0)
                        add1 = 0;
                    add2 = add3 = add4 = 0;
                    if (sb > 1) {
                        add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
                        if (add2 < 0)
                            add2 = 0;
                    }
                    if (sb > 0) {
                        add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
                        if (add3 < 0)
                            add3 = 0;
                    }
                    if (sb < 29) {
                        add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
                        if (add4 < 0)
                            add4 = 0;
                    }
                    tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
                    if (tmp < 0)
                        tmp = 0;
                    tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
                }
                tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
            }
            acc = 0;
            for (ch = 0; ch < nb_channels; ch++)
                for (sb = 0; sb < 30; sb++)
                    for (j = 0; j < 64; j++)
                        acc += tone_level_idx_temp[ch][sb][j];
            if (acc)
                tmp = c * 256 / (acc & 0xffff);
            multres = 0x66666667 * (acc * 10);
            esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
            for (ch = 0;  ch < nb_channels; ch++)
                for (sb = 0; sb < 30; sb++)
                    for (j = 0; j < 64; j++) {
                        comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
                        if (comp < 0)
                            comp += 0xff;
                        comp /= 256; // signed shift
                        switch(sb) {
                            case 0:
                                if (comp < 30)
                                    comp = 30;
                                comp += 15;
                                break;
                            case 1:
                                if (comp < 24)
                                    comp = 24;
                                comp += 10;
                                break;