comparison dca.c @ 11909:3bfcb8fd3dc9 libavcodec

Support DTS-ES extension (XCh) in dca: move original code around to allow reused by DTS-ES code Patch by Nick Brereton, nick at nbrereton dot net

11908:9b1095b2616a

    int downmix_coef[DCA_PRIM_CHANNELS_MAX][2];                  ///< stereo downmix coefficients
    int dynrange_coef;                                           ///< dynamic range coefficient

    int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];       ///< VQ encoded high frequency subbands

    float lfe_data[2 * DCA_SUBSUBFRAMES_MAX * DCA_LFE_MAX *
                   2 /*history */ ];    ///< Low frequency effect data
    int lfe_scale_factor;

    /* Subband samples history (for ADPCM) */
    float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
    DECLARE_ALIGNED(16, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512];

{
    while(len--)
        *dst++ = get_bits(gb, bits);
}

static int dca_parse_frame_header(DCAContext * s)
{
    int i, j;
    static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
    static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
    static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };

    init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);

    /* Sync code */
    get_bits(&s->gb, 32);

    av_log(s->avctx, AV_LOG_DEBUG, "\n");
#endif

    /* Primary audio coding header */
    s->subframes         = get_bits(&s->gb, 4) + 1;
    s->total_channels    = get_bits(&s->gb, 3) + 1;
    s->prim_channels     = s->total_channels;
    if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)
        s->prim_channels = DCA_PRIM_CHANNELS_MAX;   /* We only support DTS core */


    for (i = 0; i < s->prim_channels; i++) {
        s->subband_activity[i] = get_bits(&s->gb, 5) + 2;
        if (s->subband_activity[i] > DCA_SUBBANDS)
            s->subband_activity[i] = DCA_SUBBANDS;
    }
    for (i = 0; i < s->prim_channels; i++) {
        s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
        if (s->vq_start_subband[i] > DCA_SUBBANDS)
            s->vq_start_subband[i] = DCA_SUBBANDS;
    }
    get_array(&s->gb, s->joint_intensity,     s->prim_channels, 3);
    get_array(&s->gb, s->transient_huffman,   s->prim_channels, 2);
    get_array(&s->gb, s->scalefactor_huffman, s->prim_channels, 3);
    get_array(&s->gb, s->bitalloc_huffman,    s->prim_channels, 3);

    /* Get codebooks quantization indexes */
    memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
    for (j = 1; j < 11; j++)
        for (i = 0; i < s->prim_channels; i++)
            s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);

    /* Get scale factor adjustment */
    for (j = 0; j < 11; j++)
        for (i = 0; i < s->prim_channels; i++)
            s->scalefactor_adj[i][j] = 1;

    for (j = 1; j < 11; j++)
        for (i = 0; i < s->prim_channels; i++)
            if (s->quant_index_huffman[i][j] < thr[j])
                s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];

    if (s->crc_present) {
        /* Audio header CRC check */
        get_bits(&s->gb, 16);
    }

    s->current_subframe = 0;
    s->current_subsubframe = 0;

#ifdef TRACE
    av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes);
    av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels);
    for(i = 0; i < s->prim_channels; i++){
        av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", s->subband_activity[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", s->vq_start_subband[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", s->joint_intensity[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", s->transient_huffman[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", s->scalefactor_huffman[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", s->bitalloc_huffman[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:");
        for (j = 0; j < 11; j++)
            av_log(s->avctx, AV_LOG_DEBUG, " %i",
                   s->quant_index_huffman[i][j]);
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
        av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");
        for (j = 0; j < 11; j++)
            av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]);
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
    }
#endif

    return 0;
}


static inline int get_scale(GetBitContext *gb, int level, int value)
{

   } else if(level < 8)
       value = get_bits(gb, level + 1);
   return value;
}

static int dca_subframe_header(DCAContext * s)
{
    /* Primary audio coding side information */
    int j, k;

    s->subsubframes = get_bits(&s->gb, 2) + 1;

            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);

    }
    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

    for (k = 0; k < s->prim_channels; k++)
        for (l = 0; l < s->vq_start_subband[k]; l++)
            memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4],
                        4 * sizeof(subband_samples[0][0][0]));

    /* 32 subbands QMF */
    for (k = 0; k < s->prim_channels; k++) {
/*        static float pcm_to_double[8] =
            {32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/
         qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * s->channel_order_tab[k]],
                            M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ ,
                            s->add_bias );
    }

    /* Down mixing */

    if (s->prim_channels > dca_channels[s->output & DCA_CHANNEL_MASK]) {
        dca_downmix(s->samples, s->amode, s->downmix_coef);
    }

    /* Generate LFE samples for this subsubframe FIXME!!! */
    if (s->output & DCA_LFE) {
        int lfe_samples = 2 * s->lfe * s->subsubframes;

        lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,
                              s->lfe_data + lfe_samples +
                              2 * s->lfe * subsubframe,
                              &s->samples[256 * dca_lfe_index[s->amode]],
                              (1.0/256.0)*s->scale_bias,  s->add_bias);
        /* Outputs 20bits pcm samples */
    }

static int dca_subframe_footer(DCAContext * s)
{
    int aux_data_count = 0, i;
    int lfe_samples;

    /*
     * Unpack optional information
     */

    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);

    lfe_samples = 2 * s->lfe * s->subsubframes;
    for (i = 0; i < lfe_samples; i++) {
        s->lfe_data[i] = s->lfe_data[i + lfe_samples];
    }

    return 0;
}

/**

    if (!s->current_subsubframe) {
#ifdef TRACE
        av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
#endif
        /* Read subframe header */
        if (dca_subframe_header(s))
            return -1;
    }

    /* Read subsubframe */
#ifdef TRACE

                            AVPacket *avpkt)
{
    const uint8_t *buf = avpkt->data;
    int buf_size = avpkt->size;

    int i;
    int16_t *samples = data;
    DCAContext *s = avctx->priv_data;
    int channels;

        return buf_size;
    }
    //set AVCodec values with parsed data
    avctx->sample_rate = s->sample_rate;
    avctx->bit_rate = s->bit_rate;

    channels = s->prim_channels + !!s->lfe;

    if (s->amode<16) {
        avctx->channel_layout = dca_core_channel_layout[s->amode];

        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;
    for (i = 0; i < (s->sample_blocks / 8); i++) {
        dca_decode_block(s, i);
        s->dsp.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels);
        samples += 256 * channels;
    }

    return buf_size;
}

    dsputil_init(&s->dsp, avctx);
    ff_mdct_init(&s->imdct, 6, 1, 1.0);
    ff_synth_filter_init(&s->synth);
    ff_dcadsp_init(&s->dcadsp);

    for(i = 0; i < 6; 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;

11909:3bfcb8fd3dc9

    int downmix_coef[DCA_PRIM_CHANNELS_MAX][2];                  ///< stereo downmix coefficients
    int dynrange_coef;                                           ///< dynamic range coefficient

    int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];       ///< VQ encoded high frequency subbands

    float lfe_data[2 * DCA_LFE_MAX * (DCA_BLOCKS_MAX + 4)];      ///< Low frequency effect data
    int lfe_scale_factor;

    /* Subband samples history (for ADPCM) */
    float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
    DECLARE_ALIGNED(16, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512];

{
    while(len--)
        *dst++ = get_bits(gb, bits);
}

static int dca_parse_audio_coding_header(DCAContext * s)
{
    int i, j;
    static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
    static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
    static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };

    s->total_channels    = get_bits(&s->gb, 3) + 1;
    s->prim_channels     = s->total_channels;
    if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)
        s->prim_channels = DCA_PRIM_CHANNELS_MAX;   /* We only support DTS core */


    for (i = 0; i < s->prim_channels; i++) {
        s->subband_activity[i] = get_bits(&s->gb, 5) + 2;
        if (s->subband_activity[i] > DCA_SUBBANDS)
            s->subband_activity[i] = DCA_SUBBANDS;
    }
    for (i = 0; i < s->prim_channels; i++) {
        s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
        if (s->vq_start_subband[i] > DCA_SUBBANDS)
            s->vq_start_subband[i] = DCA_SUBBANDS;
    }
    get_array(&s->gb, s->joint_intensity,     s->prim_channels, 3);
    get_array(&s->gb, s->transient_huffman,   s->prim_channels, 2);
    get_array(&s->gb, s->scalefactor_huffman, s->prim_channels, 3);
    get_array(&s->gb, s->bitalloc_huffman,    s->prim_channels, 3);

    /* Get codebooks quantization indexes */
    memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
    for (j = 1; j < 11; j++)
        for (i = 0; i < s->prim_channels; i++)
            s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);

    /* Get scale factor adjustment */
    for (j = 0; j < 11; j++)
        for (i = 0; i < s->prim_channels; i++)
            s->scalefactor_adj[i][j] = 1;

    for (j = 1; j < 11; j++)
        for (i = 0; i < s->prim_channels; i++)
            if (s->quant_index_huffman[i][j] < thr[j])
                s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];

    if (s->crc_present) {
        /* Audio header CRC check */
        get_bits(&s->gb, 16);
    }

    s->current_subframe = 0;
    s->current_subsubframe = 0;

#ifdef TRACE
    av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes);
    av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels);
    for (i = 0; i < s->prim_channels; i++){
        av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", s->subband_activity[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", s->vq_start_subband[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", s->joint_intensity[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", s->transient_huffman[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", s->scalefactor_huffman[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", s->bitalloc_huffman[i]);
        av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:");
        for (j = 0; j < 11; j++)
            av_log(s->avctx, AV_LOG_DEBUG, " %i",
                   s->quant_index_huffman[i][j]);
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
        av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");
        for (j = 0; j < 11; j++)
            av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]);
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
    }
#endif

  return 0;
}

static int dca_parse_frame_header(DCAContext * s)
{
    init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);

    /* Sync code */
    get_bits(&s->gb, 32);

    av_log(s->avctx, AV_LOG_DEBUG, "\n");
#endif

    /* Primary audio coding header */
    s->subframes         = get_bits(&s->gb, 4) + 1;

    return dca_parse_audio_coding_header(s);
}


static inline int get_scale(GetBitContext *gb, int level, int value)
{

   } else if(level < 8)
       value = get_bits(gb, level + 1);
   return value;
}

static int dca_subframe_header(DCAContext * s, int block_index)
{
    /* Primary audio coding side information */
    int j, k;

    s->subsubframes = get_bits(&s->gb, 2) + 1;

            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 * (4 + block_index);
        int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes);
        float lfe_scale;

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

#ifdef TRACE
    av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n", s->subsubframes);

    }
    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 * (4 + block_index);
        int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);

        av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n");
        for (j = lfe_samples; j < lfe_end_sample; j++)
            av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]);
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
    }
#endif

    for (k = 0; k < s->prim_channels; k++)
        for (l = 0; l < s->vq_start_subband[k]; l++)
            memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4],
                        4 * sizeof(subband_samples[0][0][0]));

    return 0;
}

static int dca_filter_channels(DCAContext * s, int block_index)
{
    float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
    int k;

    /* 32 subbands QMF */
    for (k = 0; k < s->prim_channels; k++) {
/*        static float pcm_to_double[8] =
            {32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/
         qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * s->channel_order_tab[k]],
                            M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ ,
                            s->add_bias );
    }

    /* Down mixing */
    if (s->avctx->request_channels == 2 && s->prim_channels > 2) {
        dca_downmix(s->samples, s->amode, s->downmix_coef);
    }

    /* Generate LFE samples for this subsubframe FIXME!!! */
    if (s->output & DCA_LFE) {
        lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,
                              s->lfe_data + 2 * s->lfe * (block_index + 4),
                              &s->samples[256 * dca_lfe_index[s->amode]],
                              (1.0/256.0)*s->scale_bias,  s->add_bias);
        /* Outputs 20bits pcm samples */
    }

static int dca_subframe_footer(DCAContext * s)
{
    int aux_data_count = 0, i;

    /*
     * Unpack optional information
     */

    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;
}

/**

    if (!s->current_subsubframe) {
#ifdef TRACE
        av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
#endif
        /* Read subframe header */
        if (dca_subframe_header(s, block_index))
            return -1;
    }

    /* Read subsubframe */
#ifdef TRACE

                            AVPacket *avpkt)
{
    const uint8_t *buf = avpkt->data;
    int buf_size = avpkt->size;

    int lfe_samples;
    int i;
    int16_t *samples = data;
    DCAContext *s = avctx->priv_data;
    int channels;

        return buf_size;
    }
    //set AVCodec values with parsed data
    avctx->sample_rate = s->sample_rate;
    avctx->bit_rate = s->bit_rate;

    for (i = 0; i < (s->sample_blocks / 8); i++) {
        dca_decode_block(s, i);
    }

    channels = s->prim_channels + !!s->lfe;

    if (s->amode<16) {
        avctx->channel_layout = dca_core_channel_layout[s->amode];

        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++) {
        dca_filter_channels(s, i);
        s->dsp.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels);
        samples += 256 * channels;
    }

    /* update lfe history */
    lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);
    for (i = 0; i < 2 * s->lfe * 4; i++) {
        s->lfe_data[i] = s->lfe_data[i + lfe_samples];
    }

    return buf_size;
}

    dsputil_init(&s->dsp, avctx);
    ff_mdct_init(&s->imdct, 6, 1, 1.0);
    ff_synth_filter_init(&s->synth);
    ff_dcadsp_init(&s->dcadsp);

    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;