comparison dca.c @ 11912:6db8c8adf407 libavcodec

Support DTS-ES extension (XCh) in dca: Cosmetic cleanup Patch by Nick Brereton, nick at nbrereton dot net

11911:180c6aca1f6e

        init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
                 tmode_bits[i], 1, 1,
                 tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
    }

    for(i = 0; i < 10; i++)
        for(j = 0; j < 7; j++){
            if(!bitalloc_codes[i][j]) break;
            dca_smpl_bitalloc[i+1].offset = bitalloc_offsets[i];
            dca_smpl_bitalloc[i+1].wrap = 1 + (j > 4);
            dca_smpl_bitalloc[i+1].vlc[j].table = &dca_table[dca_vlc_offs[c]];
            dca_smpl_bitalloc[i+1].vlc[j].table_allocated = dca_vlc_offs[c + 1] - dca_vlc_offs[c];
            init_vlc(&dca_smpl_bitalloc[i+1].vlc[j], bitalloc_maxbits[i][j],

    s->surround_sum      = get_bits(&s->gb, 1);
    s->dialog_norm       = get_bits(&s->gb, 4);

    /* FIXME: channels mixing levels */
    s->output = s->amode;
    if(s->lfe) s->output |= DCA_LFE;

#ifdef TRACE
    av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type);
    av_log(s->avctx, AV_LOG_DEBUG, "samples deficit: %i\n", s->samples_deficit);
    av_log(s->avctx, AV_LOG_DEBUG, "crc present: %i\n", s->crc_present);

static inline int get_scale(GetBitContext *gb, int level, int value)
{
   if (level < 5) {
       /* huffman encoded */
       value += get_bitalloc(gb, &dca_scalefactor, level);
   } else if(level < 8)
       value = get_bits(gb, level + 1);
   return value;
}

static int dca_subframe_header(DCAContext * s, int base_channel, int block_index)

        }
    }

    /* Stereo downmix coefficients */
    if (!base_channel && s->prim_channels > 2) {
        if(s->downmix) {
            for (j = base_channel; 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 {

    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 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) {

            int sel = s->quant_index_huffman[k][abits];

            /*
             * Extract bits from the bit stream
             */
            if(!abits){
                memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));
            } else {
                /* Deal with transients */
                int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];
                float rscale = quant_step_size * s->scale_factor[k][l][sfi] * s->scalefactor_adj[k][sel];

                if(abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){
                    if(abits <= 7){
                        /* Block code */
                        int block_code1, block_code2, size, levels;

                        size = abits_sizes[abits-1];
                        levels = abits_levels[abits-1];

     * Unpack optional information
     */

    /* presumably optional information only appears in the core? */
    if (!base_channel) {
    if (s->timestamp)
        get_bits(&s->gb, 32);

    if (s->aux_data)
        aux_data_count = get_bits(&s->gb, 6);

    for (i = 0; i < aux_data_count; i++)
        get_bits(&s->gb, 8);

    if (s->crc_present && (s->downmix || s->dynrange))
        get_bits(&s->gb, 16);
    }

    return 0;
}

    int i, tmp;
    const uint16_t *ssrc = (const uint16_t *) src;
    uint16_t *sdst = (uint16_t *) dst;
    PutBitContext pb;

    if((unsigned)src_size > (unsigned)max_size) {
//        av_log(NULL, AV_LOG_ERROR, "Input frame size larger then DCA_MAX_FRAME_SIZE!\n");
//        return -1;
        src_size = max_size;
    }

        if (s->prim_channels > 0 &&
            s->channel_order_tab[s->prim_channels - 1] < 0)
            return -1;

        if(avctx->request_channels == 2 && s->prim_channels > 2) {
            channels = 2;
            s->output = DCA_STEREO;
            avctx->channel_layout = CH_LAYOUT_STEREO;
        }
    } else {

       and only set avctx->channels when the crc is ok. Right now the decoder could
       set the channels based on a broken first frame.*/
    if (!avctx->channels)
        avctx->channels = channels;

    if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
        return -1;
    *data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels;

    /* filter to get final output */
    for (i = 0; i < (s->sample_blocks / 8); i++) {

    for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)
        s->samples_chanptr[i] = s->samples + i * 256;
    avctx->sample_fmt = SAMPLE_FMT_S16;

    if(s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
        s->add_bias = 385.0f;
        s->scale_bias = 1.0 / 32768.0;
    } else {
        s->add_bias = 0.0f;
        s->scale_bias = 1.0;

11912:6db8c8adf407

        init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
                 tmode_bits[i], 1, 1,
                 tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
    }

    for (i = 0; i < 10; i++)
        for (j = 0; j < 7; j++){
            if (!bitalloc_codes[i][j]) break;
            dca_smpl_bitalloc[i+1].offset = bitalloc_offsets[i];
            dca_smpl_bitalloc[i+1].wrap = 1 + (j > 4);
            dca_smpl_bitalloc[i+1].vlc[j].table = &dca_table[dca_vlc_offs[c]];
            dca_smpl_bitalloc[i+1].vlc[j].table_allocated = dca_vlc_offs[c + 1] - dca_vlc_offs[c];
            init_vlc(&dca_smpl_bitalloc[i+1].vlc[j], bitalloc_maxbits[i][j],

    s->surround_sum      = get_bits(&s->gb, 1);
    s->dialog_norm       = get_bits(&s->gb, 4);

    /* FIXME: channels mixing levels */
    s->output = s->amode;
    if (s->lfe) s->output |= DCA_LFE;

#ifdef TRACE
    av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type);
    av_log(s->avctx, AV_LOG_DEBUG, "samples deficit: %i\n", s->samples_deficit);
    av_log(s->avctx, AV_LOG_DEBUG, "crc present: %i\n", s->crc_present);

static inline int get_scale(GetBitContext *gb, int level, int value)
{
   if (level < 5) {
       /* huffman encoded */
       value += get_bitalloc(gb, &dca_scalefactor, level);
   } else if (level < 8)
       value = get_bits(gb, level + 1);
   return value;
}

static int dca_subframe_header(DCAContext * s, int base_channel, int block_index)

        }
    }

    /* Stereo downmix coefficients */
    if (!base_channel && s->prim_channels > 2) {
        if (s->downmix) {
            for (j = base_channel; 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 {

    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 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) {

            int sel = s->quant_index_huffman[k][abits];

            /*
             * Extract bits from the bit stream
             */
            if (!abits){
                memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));
            } else {
                /* Deal with transients */
                int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];
                float rscale = quant_step_size * s->scale_factor[k][l][sfi] * s->scalefactor_adj[k][sel];

                if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){
                    if (abits <= 7){
                        /* Block code */
                        int block_code1, block_code2, size, levels;

                        size = abits_sizes[abits-1];
                        levels = abits_levels[abits-1];

     * Unpack optional information
     */

    /* presumably optional information only appears in the core? */
    if (!base_channel) {
        if (s->timestamp)
            get_bits(&s->gb, 32);

        if (s->aux_data)
            aux_data_count = get_bits(&s->gb, 6);

        for (i = 0; i < aux_data_count; i++)
            get_bits(&s->gb, 8);

        if (s->crc_present && (s->downmix || s->dynrange))
            get_bits(&s->gb, 16);
    }

    return 0;
}

    int i, tmp;
    const uint16_t *ssrc = (const uint16_t *) src;
    uint16_t *sdst = (uint16_t *) dst;
    PutBitContext pb;

    if ((unsigned)src_size > (unsigned)max_size) {
//        av_log(NULL, AV_LOG_ERROR, "Input frame size larger then DCA_MAX_FRAME_SIZE!\n");
//        return -1;
        src_size = max_size;
    }

        if (s->prim_channels > 0 &&
            s->channel_order_tab[s->prim_channels - 1] < 0)
            return -1;

        if (avctx->request_channels == 2 && s->prim_channels > 2) {
            channels = 2;
            s->output = DCA_STEREO;
            avctx->channel_layout = CH_LAYOUT_STEREO;
        }
    } else {

       and only set avctx->channels when the crc is ok. Right now the decoder could
       set the channels based on a broken first frame.*/
    if (!avctx->channels)
        avctx->channels = channels;

    if (*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
        return -1;
    *data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels;

    /* filter to get final output */
    for (i = 0; i < (s->sample_blocks / 8); i++) {

    for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)
        s->samples_chanptr[i] = s->samples + i * 256;
    avctx->sample_fmt = SAMPLE_FMT_S16;

    if (s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
        s->add_bias = 385.0f;
        s->scale_bias = 1.0 / 32768.0;
    } else {
        s->add_bias = 0.0f;
        s->scale_bias = 1.0;