dca.c

ffmpeg移植到symbian的全部源代码

            if (s->bitalloc_huffman[j] == 6)
                s->bitalloc[j][k] = get_bits(&s->gb, 5);
            else if (s->bitalloc_huffman[j] == 5)
                s->bitalloc[j][k] = get_bits(&s->gb, 4);
            else if (s->bitalloc_huffman[j] == 7) {
                av_log(s->avctx, AV_LOG_ERROR,
                       "Invalid bit allocation index\n");
                return -1;
            } else {
                s->bitalloc[j][k] =
                    get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);
            }

            if (s->bitalloc[j][k] > 26) {
//                 av_log(s->avctx,AV_LOG_DEBUG,"bitalloc index [%i][%i] too big (%i)\n",
//                          j, k, s->bitalloc[j][k]);
                return -1;
            }
        }
    }

    /* Transition mode */
    for (j = 0; j < s->prim_channels; j++) {
        for (k = 0; k < s->subband_activity[j]; k++) {
            s->transition_mode[j][k] = 0;
            if (s->subsubframes > 1 &&
                k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) {
                s->transition_mode[j][k] =
                    get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]);
            }
        }
    }

    for (j = 0; j < s->prim_channels; j++) {
        const uint32_t *scale_table;
        int scale_sum;

        memset(s->scale_factor[j], 0, s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);

        if (s->scalefactor_huffman[j] == 6)
            scale_table = scale_factor_quant7;
        else
            scale_table = scale_factor_quant6;

        /* When huffman coded, only the difference is encoded */
        scale_sum = 0;

        for (k = 0; k < s->subband_activity[j]; k++) {
            if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) {
                scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum);
                s->scale_factor[j][k][0] = scale_table[scale_sum];
            }

            if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) {
                /* Get second scale factor */
                scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum);
                s->scale_factor[j][k][1] = scale_table[scale_sum];
            }
        }
    }

    /* Joint subband scale factor codebook select */
    for (j = 0; j < s->prim_channels; j++) {
        /* Transmitted only if joint subband coding enabled */
        if (s->joint_intensity[j] > 0)
            s->joint_huff[j] = get_bits(&s->gb, 3);
    }

    /* Scale factors for joint subband coding */
    for (j = 0; j < s->prim_channels; j++) {
        int source_channel;

        /* Transmitted only if joint subband coding enabled */
        if (s->joint_intensity[j] > 0) {
            int scale = 0;
            source_channel = s->joint_intensity[j] - 1;

            /* When huffman coded, only the difference is encoded
             * (is this valid as well for joint scales ???) */

            for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) {
                scale = get_scale(&s->gb, s->joint_huff[j], 0);
                scale += 64;    /* bias */
                s->joint_scale_factor[j][k] = scale;    /*joint_scale_table[scale]; */
            }

            if (!s->debug_flag & 0x02) {
                av_log(s->avctx, AV_LOG_DEBUG,
                       "Joint stereo coding not supported\n");
                s->debug_flag |= 0x02;
            }
        }
    }

    /* Stereo downmix coefficients */
    if (s->prim_channels > 2) {
        if(s->downmix) {
            for (j = 0; j < s->prim_channels; j++) {
                s->downmix_coef[j][0] = get_bits(&s->gb, 7);
                s->downmix_coef[j][1] = get_bits(&s->gb, 7);
            }
        } else {
            int am = s->amode & DCA_CHANNEL_MASK;
            for (j = 0; j < s->prim_channels; j++) {
                s->downmix_coef[j][0] = dca_default_coeffs[am][j][0];
                s->downmix_coef[j][1] = dca_default_coeffs[am][j][1];
            }
        }
    }

    /* Dynamic range coefficient */
    if (s->dynrange)
        s->dynrange_coef = get_bits(&s->gb, 8);

    /* Side information CRC check word */
    if (s->crc_present) {
        get_bits(&s->gb, 16);
    }

    /*
     * Primary audio data arrays
     */

    /* VQ encoded high frequency subbands */
    for (j = 0; j < s->prim_channels; j++)
        for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
            /* 1 vector -> 32 samples */
            s->high_freq_vq[j][k] = get_bits(&s->gb, 10);

    /* Low frequency effect data */
    if (s->lfe) {
        /* LFE samples */
        int lfe_samples = 2 * s->lfe * s->subsubframes;
        float lfe_scale;

        for (j = lfe_samples; j < lfe_samples * 2; j++) {
            /* Signed 8 bits int */
            s->lfe_data[j] = get_sbits(&s->gb, 8);
        }

        /* Scale factor index */
        s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 8)];

        /* Quantization step size * scale factor */
        lfe_scale = 0.035 * s->lfe_scale_factor;

        for (j = lfe_samples; j < lfe_samples * 2; j++)
            s->lfe_data[j] *= lfe_scale;
    }

#ifdef TRACE
    av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n", s->subsubframes);
    av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n",
           s->partial_samples);
    for (j = 0; j < s->prim_channels; j++) {
        av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:");
        for (k = 0; k < s->subband_activity[j]; k++)
            av_log(s->avctx, AV_LOG_DEBUG, " %i", s->prediction_mode[j][k]);
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
    }
    for (j = 0; j < s->prim_channels; j++) {
        for (k = 0; k < s->subband_activity[j]; k++)
                av_log(s->avctx, AV_LOG_DEBUG,
                       "prediction coefs: %f, %f, %f, %f\n",
                       (float) adpcm_vb[s->prediction_vq[j][k]][0] / 8192,
                       (float) adpcm_vb[s->prediction_vq[j][k]][1] / 8192,
                       (float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192,
                       (float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192);
    }
    for (j = 0; j < s->prim_channels; j++) {
        av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: ");
        for (k = 0; k < s->vq_start_subband[j]; k++)
            av_log(s->avctx, AV_LOG_DEBUG, "%2.2i ", s->bitalloc[j][k]);
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
    }
    for (j = 0; j < s->prim_channels; j++) {
        av_log(s->avctx, AV_LOG_DEBUG, "Transition mode:");
        for (k = 0; k < s->subband_activity[j]; k++)
            av_log(s->avctx, AV_LOG_DEBUG, " %i", s->transition_mode[j][k]);
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
    }
    for (j = 0; j < s->prim_channels; j++) {
        av_log(s->avctx, AV_LOG_DEBUG, "Scale factor:");
        for (k = 0; k < s->subband_activity[j]; k++) {
            if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0)
                av_log(s->avctx, AV_LOG_DEBUG, " %i", s->scale_factor[j][k][0]);
            if (k < s->vq_start_subband[j] && s->transition_mode[j][k])
                av_log(s->avctx, AV_LOG_DEBUG, " %i(t)", s->scale_factor[j][k][1]);
        }
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
    }
    for (j = 0; j < s->prim_channels; j++) {
        if (s->joint_intensity[j] > 0) {
            int source_channel = s->joint_intensity[j] - 1;
            av_log(s->avctx, AV_LOG_DEBUG, "Joint scale factor index:\n");
            for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++)
                av_log(s->avctx, AV_LOG_DEBUG, " %i", s->joint_scale_factor[j][k]);
            av_log(s->avctx, AV_LOG_DEBUG, "\n");
        }
    }
    if (s->prim_channels > 2 && s->downmix) {
        av_log(s->avctx, AV_LOG_DEBUG, "Downmix coeffs:\n");
        for (j = 0; j < s->prim_channels; j++) {
            av_log(s->avctx, AV_LOG_DEBUG, "Channel 0,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][0]]);
            av_log(s->avctx, AV_LOG_DEBUG, "Channel 1,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][1]]);
        }
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
    }
    for (j = 0; j < s->prim_channels; j++)
        for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
            av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]);
    if(s->lfe){
        int lfe_samples = 2 * s->lfe * s->subsubframes;
        av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n");
        for (j = lfe_samples; j < lfe_samples * 2; j++)
            av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]);
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
    }
#endif

    return 0;
}

static void qmf_32_subbands(DCAContext * s, int chans,
                            float samples_in[32][8], float *samples_out,
                            float scale, float bias)
{
    const float *prCoeff;
    int i, j, k;
    float praXin[33], *raXin = &praXin[1];

    float *subband_fir_hist = s->subband_fir_hist[chans];
    float *subband_fir_hist2 = s->subband_fir_noidea[chans];

    int chindex = 0, subindex;

    praXin[0] = 0.0;

    /* Select filter */
    if (!s->multirate_inter)    /* Non-perfect reconstruction */
        prCoeff = fir_32bands_nonperfect;
    else                        /* Perfect reconstruction */
        prCoeff = fir_32bands_perfect;

    /* Reconstructed channel sample index */
    for (subindex = 0; subindex < 8; subindex++) {
        float t1, t2, sum[16], diff[16];

        /* Load in one sample from each subband and clear inactive subbands */
        for (i = 0; i < s->subband_activity[chans]; i++)
            raXin[i] = samples_in[i][subindex];
        for (; i < 32; i++)
            raXin[i] = 0.0;

        /* Multiply by cosine modulation coefficients and
         * create temporary arrays SUM and DIFF */
        for (j = 0, k = 0; k < 16; k++) {
            t1 = 0.0;
            t2 = 0.0;
            for (i = 0; i < 16; i++, j++){
                t1 += (raXin[2 * i] + raXin[2 * i + 1]) * cos_mod[j];
                t2 += (raXin[2 * i] + raXin[2 * i - 1]) * cos_mod[j + 256];
            }
            sum[k] = t1 + t2;
            diff[k] = t1 - t2;
        }

        j = 512;
        /* Store history */
        for (k = 0; k < 16; k++)
            subband_fir_hist[k] = cos_mod[j++] * sum[k];
        for (k = 0; k < 16; k++)
            subband_fir_hist[32-k-1] = cos_mod[j++] * diff[k];

        /* Multiply by filter coefficients */
        for (k = 31, i = 0; i < 32; i++, k--)
            for (j = 0; j < 512; j += 64){
                subband_fir_hist2[i]    += prCoeff[i+j]  * ( subband_fir_hist[i+j] - subband_fir_hist[j+k]);
                subband_fir_hist2[i+32] += prCoeff[i+j+32]*(-subband_fir_hist[i+j] - subband_fir_hist[j+k]);
            }

        /* Create 32 PCM output samples */
        for (i = 0; i < 32; i++)
            samples_out[chindex++] = subband_fir_hist2[i] * scale + bias;

        /* Update working arrays */
        memmove(&subband_fir_hist[32], &subband_fir_hist[0], (512 - 32) * sizeof(float));
        memmove(&subband_fir_hist2[0], &subband_fir_hist2[32], 32 * sizeof(float));
        memset(&subband_fir_hist2[32], 0, 32 * sizeof(float));
    }
}

static void lfe_interpolation_fir(int decimation_select,
                                  int num_deci_sample, float *samples_in,
                                  float *samples_out, float scale,
                                  float bias)
{
    /* samples_in: An array holding decimated samples.
     *   Samples in current subframe starts from samples_in[0],
     *   while samples_in[-1], samples_in[-2], ..., stores samples
     *   from last subframe as history.
     *
     * samples_out: An array holding interpolated samples
     */

    int decifactor, k, j;
    const float *prCoeff;

    int interp_index = 0;       /* Index to the interpolated samples */
    int deciindex;

    /* Select decimation filter */
    if (decimation_select == 1) {
        decifactor = 128;
        prCoeff = lfe_fir_128;
    } else {
        decifactor = 64;
        prCoeff = lfe_fir_64;
    }
    /* Interpolation */
    for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
        /* One decimated sample generates decifactor interpolated ones */
        for (k = 0; k < decifactor; k++) {
            float rTmp = 0.0;
            //FIXME the coeffs are symetric, fix that
            for (j = 0; j < 512 / decifactor; j++)
                rTmp += samples_in[deciindex - j] * prCoeff[k + j * decifactor];
            samples_out[interp_index++] = rTmp / scale + bias;
        }
    }
}

/* downmixing routines */
#define MIX_REAR1(samples, si1, rs, coef) \
     samples[i]     += samples[si1] * coef[rs][0]; \
     samples[i+256] += samples[si1] * coef[rs][1];

#define MIX_REAR2(samples, si1, si2, rs, coef) \
     samples[i]     += samples[si1] * coef[rs][0] + samples[si2] * coef[rs+1][0]; \
     samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs+1][1];

#define MIX_FRONT3(samples, coef) \
    t = samples[i]; \
    samples[i]     = t * coef[0][0] + samples[i+256] * coef[1][0] + samples[i+512] * coef[2][0]; \
    samples[i+256] = t * coef[0][1] + samples[i+256] * coef[1][1] + samples[i+512] * coef[2][1];

#define DOWNMIX_TO_STEREO(op1, op2) \
    for(i = 0; i < 256; i++){ \
        op1 \
        op2 \
    }

static void dca_downmix(float *samples, int srcfmt,
                        int downmix_coef[DCA_PRIM_CHANNELS_MAX][2])
{
    int i;
    float t;
    float coef[DCA_PRIM_CHANNELS_MAX][2];

    for(i=0; i<DCA_PRIM_CHANNELS_MAX; i++) {
        coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]];
        coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]];
    }

    switch (srcfmt) {
    case DCA_MONO:
    case DCA_CHANNEL:
    case DCA_STEREO_TOTAL:
    case DCA_STEREO_SUMDIFF:
    case DCA_4F2R:
        av_log(NULL, 0, "Not implemented!\n");
        break;
    case DCA_STEREO:
        break;
    case DCA_3F:
        DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),);
        break;
    case DCA_2F1R:
        DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + 512, 2, coef),);
        break;
    case DCA_3F1R:
        DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
                          MIX_REAR1(samples, i + 768, 3, coef));
        break;
    case DCA_2F2R:
        DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + 512, i + 768, 2, coef),);
        break;
    case DCA_3F2R:
        DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
                          MIX_REAR2(samples, i + 768, i + 1024, 3, coef));
        break;
    }
}


/* Very compact version of the block code decoder that does not use table
 * look-up but is slightly slower */
static int decode_blockcode(int code, int levels, int *values)
{
    int i;
    int offset = (levels - 1) >> 1;

    for (i = 0; i < 4; i++) {
        values[i] = (code % levels) - offset;
        code /= levels;
    }

    if (code == 0)
        return 0;
    else {
        av_log(NULL, AV_LOG_ERROR, "ERROR: block code look-up failed\n");
        return -1;
    }
}

static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };

static int dca_subsubframe(DCAContext * s)
{
    int k, l;
    int subsubframe = s->current_subsubframe;

    const float *quant_step_table;

    /* FIXME */
    float subband_samples[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];

    /*