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

                            case 2:
                            case 3:
                            case 4:
                                if (comp < 16)
                                    comp = 16;
                        }
                        if (comp <= 5)
                            tmp = 0;
                        else if (comp <= 10)
                            tmp = 10;
                        else if (comp <= 16)
                            tmp = 16;
                        else if (comp <= 24)
                            tmp = -1;
                        else
                            tmp = 0;
                        coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
                    }
            for (sb = 0; sb < 30; sb++)
                fix_coding_method_array(sb, nb_channels, coding_method);
            for (ch = 0; ch < nb_channels; ch++)
                for (sb = 0; sb < 30; sb++)
                    for (j = 0; j < 64; j++)
                        if (sb >= 10) {
                            if (coding_method[ch][sb][j] < 10)
                                coding_method[ch][sb][j] = 10;
                        } else {
                            if (sb >= 2) {
                                if (coding_method[ch][sb][j] < 16)
                                    coding_method[ch][sb][j] = 16;
                            } else {
                                if (coding_method[ch][sb][j] < 30)
                                    coding_method[ch][sb][j] = 30;
                            }
                        }
    } else { // superblocktype_2_3 != 0
        for (ch = 0; ch < nb_channels; ch++)
            for (sb = 0; sb < 30; sb++)
                for (j = 0; j < 64; j++)
                    coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
    }

    return;
}


/**
 *
 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
 *
 * @param q         context
 * @param gb        bitreader context
 * @param length    packet length in bits
 * @param sb_min    lower subband processed (sb_min included)
 * @param sb_max    higher subband processed (sb_max excluded)
 */
static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
{
    int sb, j, k, n, ch, run, channels;
    int joined_stereo, zero_encoding, chs;
    int type34_first;
    float type34_div = 0;
    float type34_predictor;
    float samples[10], sign_bits[16];

    if (length == 0) {
        // If no data use noise
        for (sb=sb_min; sb < sb_max; sb++)
            build_sb_samples_from_noise (q, sb);

        return;
    }

    for (sb = sb_min; sb < sb_max; sb++) {
        FIX_NOISE_IDX(q->noise_idx);

        channels = q->nb_channels;

        if (q->nb_channels <= 1 || sb < 12)
            joined_stereo = 0;
        else if (sb >= 24)
            joined_stereo = 1;
        else
            joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;

        if (joined_stereo) {
            if (BITS_LEFT(length,gb) >= 16)
                for (j = 0; j < 16; j++)
                    sign_bits[j] = get_bits1 (gb);

            for (j = 0; j < 64; j++)
                if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
                    q->coding_method[0][sb][j] = q->coding_method[1][sb][j];

            fix_coding_method_array(sb, q->nb_channels, q->coding_method);
            channels = 1;
        }

        for (ch = 0; ch < channels; ch++) {
            zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
            type34_predictor = 0.0;
            type34_first = 1;

            for (j = 0; j < 128; ) {
                switch (q->coding_method[ch][sb][j / 2]) {
                    case 8:
                        if (BITS_LEFT(length,gb) >= 10) {
                            if (zero_encoding) {
                                for (k = 0; k < 5; k++) {
                                    if ((j + 2 * k) >= 128)
                                        break;
                                    samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
                                }
                            } else {
                                n = get_bits(gb, 8);
                                for (k = 0; k < 5; k++)
                                    samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
                            }
                            for (k = 0; k < 5; k++)
                                samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
                        } else {
                            for (k = 0; k < 10; k++)
                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
                        }
                        run = 10;
                        break;

                    case 10:
                        if (BITS_LEFT(length,gb) >= 1) {
                            float f = 0.81;

                            if (get_bits1(gb))
                                f = -f;
                            f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
                            samples[0] = f;
                        } else {
                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
                        }
                        run = 1;
                        break;

                    case 16:
                        if (BITS_LEFT(length,gb) >= 10) {
                            if (zero_encoding) {
                                for (k = 0; k < 5; k++) {
                                    if ((j + k) >= 128)
                                        break;
                                    samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
                                }
                            } else {
                                n = get_bits (gb, 8);
                                for (k = 0; k < 5; k++)
                                    samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
                            }
                        } else {
                            for (k = 0; k < 5; k++)
                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
                        }
                        run = 5;
                        break;

                    case 24:
                        if (BITS_LEFT(length,gb) >= 7) {
                            n = get_bits(gb, 7);
                            for (k = 0; k < 3; k++)
                                samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
                        } else {
                            for (k = 0; k < 3; k++)
                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
                        }
                        run = 3;
                        break;

                    case 30:
                        if (BITS_LEFT(length,gb) >= 4)
                            samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
                        else
                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);

                        run = 1;
                        break;

                    case 34:
                        if (BITS_LEFT(length,gb) >= 7) {
                            if (type34_first) {
                                type34_div = (float)(1 << get_bits(gb, 2));
                                samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
                                type34_predictor = samples[0];
                                type34_first = 0;
                            } else {
                                samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
                                type34_predictor = samples[0];
                            }
                        } else {
                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
                        }
                        run = 1;
                        break;

                    default:
                        samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
                        run = 1;
                        break;
                }

                if (joined_stereo) {
                    float tmp[10][MPA_MAX_CHANNELS];

                    for (k = 0; k < run; k++) {
                        tmp[k][0] = samples[k];
                        tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
                    }
                    for (chs = 0; chs < q->nb_channels; chs++)
                        for (k = 0; k < run; k++)
                            if ((j + k) < 128)
                                q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
                } else {
                    for (k = 0; k < run; k++)
                        if ((j + k) < 128)
                            q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
                }

                j += run;
            } // j loop
        } // channel loop
    } // subband loop
}


/**
 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
 * This is similar to process_subpacket_9, but for a single channel and for element [0]
 * same VLC tables as process_subpacket_9 are used.
 *
 * @param q         context
 * @param quantized_coeffs    pointer to quantized_coeffs[ch][0]
 * @param gb        bitreader context
 * @param length    packet length in bits
 */
static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
{
    int i, k, run, level, diff;

    if (BITS_LEFT(length,gb) < 16)
        return;
    level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);

    quantized_coeffs[0] = level;

    for (i = 0; i < 7; ) {
        if (BITS_LEFT(length,gb) < 16)
            break;
        run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;

        if (BITS_LEFT(length,gb) < 16)
            break;
        diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);

        for (k = 1; k <= run; k++)
            quantized_coeffs[i + k] = (level + ((k * diff) / run));

        level += diff;
        i += run;
    }
}


/**
 * Related to synthesis filter, process data from packet 10
 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
 *
 * @param q         context
 * @param gb        bitreader context
 * @param length    packet length in bits
 */
static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
{
    int sb, j, k, n, ch;

    for (ch = 0; ch < q->nb_channels; ch++) {
        init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);

        if (BITS_LEFT(length,gb) < 16) {
            memset(q->quantized_coeffs[ch][0], 0, 8);
            break;
        }
    }

    n = q->sub_sampling + 1;

    for (sb = 0; sb < n; sb++)
        for (ch = 0; ch < q->nb_channels; ch++)
            for (j = 0; j < 8; j++) {
                if (BITS_LEFT(length,gb) < 1)
                    break;
                if (get_bits1(gb)) {
                    for (k=0; k < 8; k++) {
                        if (BITS_LEFT(length,gb) < 16)
                            break;
                        q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
                    }
                } else {
                    for (k=0; k < 8; k++)
                        q->tone_level_idx_hi1[ch][sb][j][k] = 0;
                }
            }

    n = QDM2_SB_USED(q->sub_sampling) - 4;

    for (sb = 0; sb < n; sb++)
        for (ch = 0; ch < q->nb_channels; ch++) {
            if (BITS_LEFT(length,gb) < 16)
                break;
            q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
            if (sb > 19)
                q->tone_level_idx_hi2[ch][sb] -= 16;
            else
                for (j = 0; j < 8; j++)
                    q->tone_level_idx_mid[ch][sb][j] = -16;
        }

    n = QDM2_SB_USED(q->sub_sampling) - 5;

    for (sb = 0; sb < n; sb++)
        for (ch = 0; ch < q->nb_channels; ch++)
            for (j = 0; j < 8; j++) {
                if (BITS_LEFT(length,gb) < 16)
                    break;
                q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
            }
}

/**
 * Process subpacket 9, init quantized_coeffs with data from it
 *
 * @param q       context
 * @param node    pointer to node with packet
 */
static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
{
    GetBitContext gb;
    int i, j, k, n, ch, run, level, diff;

    init_get_bits(&gb, node->packet->data, node->packet->size*8);

    n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function

    for (i = 1; i < n; i++)
        for (ch=0; ch < q->nb_channels; ch++) {
            level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
            q->quantized_coeffs[ch][i][0] = level;

            for (j = 0; j < (8 - 1); ) {
                run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
                diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);

                for (k = 1; k <= run; k++)
                    q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));

                level += diff;
                j += run;
            }
        }

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