wz-phone/vendor/audiopus_sys/opus/silk/decode_parameters.c

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#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

#include "main.h"

/* Decode parameters from payload */
void silk_decode_parameters(
    silk_decoder_state          *psDec,                         /* I/O  State                                       */
    silk_decoder_control        *psDecCtrl,                     /* I/O  Decoder control                             */
    opus_int                    condCoding                      /* I    The type of conditional coding to use       */
)
{
    opus_int   i, k, Ix;
    opus_int16 pNLSF_Q15[ MAX_LPC_ORDER ], pNLSF0_Q15[ MAX_LPC_ORDER ];
    const opus_int8 *cbk_ptr_Q7;

    /* Dequant Gains */
    silk_gains_dequant( psDecCtrl->Gains_Q16, psDec->indices.GainsIndices,
        &psDec->LastGainIndex, condCoding == CODE_CONDITIONALLY, psDec->nb_subfr );

    /****************/
    /* Decode NLSFs */
    /****************/
    silk_NLSF_decode( pNLSF_Q15, psDec->indices.NLSFIndices, psDec->psNLSF_CB );

    /* Convert NLSF parameters to AR prediction filter coefficients */
    silk_NLSF2A( psDecCtrl->PredCoef_Q12[ 1 ], pNLSF_Q15, psDec->LPC_order, psDec->arch );

    /* If just reset, e.g., because internal Fs changed, do not allow interpolation */
    /* improves the case of packet loss in the first frame after a switch           */
    if( psDec->first_frame_after_reset == 1 ) {
        psDec->indices.NLSFInterpCoef_Q2 = 4;
    }

    if( psDec->indices.NLSFInterpCoef_Q2 < 4 ) {
        /* Calculation of the interpolated NLSF0 vector from the interpolation factor, */
        /* the previous NLSF1, and the current NLSF1                                   */
        for( i = 0; i < psDec->LPC_order; i++ ) {
            pNLSF0_Q15[ i ] = psDec->prevNLSF_Q15[ i ] + silk_RSHIFT( silk_MUL( psDec->indices.NLSFInterpCoef_Q2,
                pNLSF_Q15[ i ] - psDec->prevNLSF_Q15[ i ] ), 2 );
        }

        /* Convert NLSF parameters to AR prediction filter coefficients */
        silk_NLSF2A( psDecCtrl->PredCoef_Q12[ 0 ], pNLSF0_Q15, psDec->LPC_order, psDec->arch );
    } else {
        /* Copy LPC coefficients for first half from second half */
        silk_memcpy( psDecCtrl->PredCoef_Q12[ 0 ], psDecCtrl->PredCoef_Q12[ 1 ], psDec->LPC_order * sizeof( opus_int16 ) );
    }

    silk_memcpy( psDec->prevNLSF_Q15, pNLSF_Q15, psDec->LPC_order * sizeof( opus_int16 ) );

    /* After a packet loss do BWE of LPC coefs */
    if( psDec->lossCnt ) {
        silk_bwexpander( psDecCtrl->PredCoef_Q12[ 0 ], psDec->LPC_order, BWE_AFTER_LOSS_Q16 );
        silk_bwexpander( psDecCtrl->PredCoef_Q12[ 1 ], psDec->LPC_order, BWE_AFTER_LOSS_Q16 );
    }

    if( psDec->indices.signalType == TYPE_VOICED ) {
        /*********************/
        /* Decode pitch lags */
        /*********************/

        /* Decode pitch values */
        silk_decode_pitch( psDec->indices.lagIndex, psDec->indices.contourIndex, psDecCtrl->pitchL, psDec->fs_kHz, psDec->nb_subfr );

        /* Decode Codebook Index */
        cbk_ptr_Q7 = silk_LTP_vq_ptrs_Q7[ psDec->indices.PERIndex ]; /* set pointer to start of codebook */

        for( k = 0; k < psDec->nb_subfr; k++ ) {
            Ix = psDec->indices.LTPIndex[ k ];
            for( i = 0; i < LTP_ORDER; i++ ) {
                psDecCtrl->LTPCoef_Q14[ k * LTP_ORDER + i ] = silk_LSHIFT( cbk_ptr_Q7[ Ix * LTP_ORDER + i ], 7 );
            }
        }

        /**********************/
        /* Decode LTP scaling */
        /**********************/
        Ix = psDec->indices.LTP_scaleIndex;
        psDecCtrl->LTP_scale_Q14 = silk_LTPScales_table_Q14[ Ix ];
    } else {
        silk_memset( psDecCtrl->pitchL,      0,             psDec->nb_subfr * sizeof( opus_int   ) );
        silk_memset( psDecCtrl->LTPCoef_Q14, 0, LTP_ORDER * psDec->nb_subfr * sizeof( opus_int16 ) );
        psDec->indices.PERIndex  = 0;
        psDecCtrl->LTP_scale_Q14 = 0;
    }
}