/* Low-Bitrate Redundancy (LBRR) encoding. Reuse all parameters but encode excitation at lower bitrate  */
static OPUS_INLINE void silk_LBRR_encode_FIX(
    silk_encoder_state_FIX          *psEnc,                                 /* I/O  Pointer to Silk FIX encoder state                                           */
    silk_encoder_control_FIX        *psEncCtrl,                             /* I/O  Pointer to Silk FIX encoder control struct                                  */
    const opus_int32                xfw_Q3[],                               /* I    Input signal                                                                */
    opus_int                        condCoding                              /* I    The type of conditional coding used so far for this frame                   */
)
{
    opus_int32   TempGains_Q16[ MAX_NB_SUBFR ];
    SideInfoIndices *psIndices_LBRR = &psEnc->sCmn.indices_LBRR[ psEnc->sCmn.nFramesEncoded ];
    silk_nsq_state sNSQ_LBRR;

    /*******************************************/
    /* Control use of inband LBRR              */
    /*******************************************/
    if( psEnc->sCmn.LBRR_enabled && psEnc->sCmn.speech_activity_Q8 > SILK_FIX_CONST( LBRR_SPEECH_ACTIVITY_THRES, 8 ) ) {
        psEnc->sCmn.LBRR_flags[ psEnc->sCmn.nFramesEncoded ] = 1;

        /* Copy noise shaping quantizer state and quantization indices from regular encoding */
        silk_memcpy( &sNSQ_LBRR, &psEnc->sCmn.sNSQ, sizeof( silk_nsq_state ) );
        silk_memcpy( psIndices_LBRR, &psEnc->sCmn.indices, sizeof( SideInfoIndices ) );

        /* Save original gains */
        silk_memcpy( TempGains_Q16, psEncCtrl->Gains_Q16, psEnc->sCmn.nb_subfr * sizeof( opus_int32 ) );

        if( psEnc->sCmn.nFramesEncoded == 0 || psEnc->sCmn.LBRR_flags[ psEnc->sCmn.nFramesEncoded - 1 ] == 0 ) {
            /* First frame in packet or previous frame not LBRR coded */
            psEnc->sCmn.LBRRprevLastGainIndex = psEnc->sShape.LastGainIndex;

            /* Increase Gains to get target LBRR rate */
            psIndices_LBRR->GainsIndices[ 0 ] = psIndices_LBRR->GainsIndices[ 0 ] + psEnc->sCmn.LBRR_GainIncreases;
            psIndices_LBRR->GainsIndices[ 0 ] = silk_min_int( psIndices_LBRR->GainsIndices[ 0 ], N_LEVELS_QGAIN - 1 );
        }

        /* Decode to get gains in sync with decoder         */
        /* Overwrite unquantized gains with quantized gains */
        silk_gains_dequant( psEncCtrl->Gains_Q16, psIndices_LBRR->GainsIndices,
            &psEnc->sCmn.LBRRprevLastGainIndex, condCoding == CODE_CONDITIONALLY, psEnc->sCmn.nb_subfr );

        /*****************************************/
        /* Noise shaping quantization            */
        /*****************************************/
        if( psEnc->sCmn.nStatesDelayedDecision > 1 || psEnc->sCmn.warping_Q16 > 0 ) {
            silk_NSQ_del_dec( &psEnc->sCmn, &sNSQ_LBRR, psIndices_LBRR, xfw_Q3,
                psEnc->sCmn.pulses_LBRR[ psEnc->sCmn.nFramesEncoded ], psEncCtrl->PredCoef_Q12[ 0 ], psEncCtrl->LTPCoef_Q14,
                psEncCtrl->AR2_Q13, psEncCtrl->HarmShapeGain_Q14, psEncCtrl->Tilt_Q14, psEncCtrl->LF_shp_Q14,
                psEncCtrl->Gains_Q16, psEncCtrl->pitchL, psEncCtrl->Lambda_Q10, psEncCtrl->LTP_scale_Q14 );
        } else {
            silk_NSQ( &psEnc->sCmn, &sNSQ_LBRR, psIndices_LBRR, xfw_Q3,
                psEnc->sCmn.pulses_LBRR[ psEnc->sCmn.nFramesEncoded ], psEncCtrl->PredCoef_Q12[ 0 ], psEncCtrl->LTPCoef_Q14,
                psEncCtrl->AR2_Q13, psEncCtrl->HarmShapeGain_Q14, psEncCtrl->Tilt_Q14, psEncCtrl->LF_shp_Q14,
                psEncCtrl->Gains_Q16, psEncCtrl->pitchL, psEncCtrl->Lambda_Q10, psEncCtrl->LTP_scale_Q14 );
        }

        /* Restore original gains */
        silk_memcpy( psEncCtrl->Gains_Q16, TempGains_Q16, psEnc->sCmn.nb_subfr * sizeof( opus_int32 ) );
    }
}

/* Low-Bitrate Redundancy (LBRR) encoding. Reuse all parameters but encode excitation at lower bitrate  */
static void silk_LBRR_encode_FLP(
    silk_encoder_state_FLP          *psEnc,                             /* I/O  Encoder state FLP                           */
    silk_encoder_control_FLP        *psEncCtrl,                         /* I/O  Encoder control FLP                         */
    const silk_float                xfw[],                              /* I    Input signal                                */
    opus_int                        condCoding                          /* I    The type of conditional coding used so far for this frame */
)
{
    opus_int     k;
    opus_int32   Gains_Q16[ MAX_NB_SUBFR ];
    silk_float   TempGains[ MAX_NB_SUBFR ];
    SideInfoIndices *psIndices_LBRR = &psEnc->sCmn.indices_LBRR[ psEnc->sCmn.nFramesEncoded ];
    silk_nsq_state sNSQ_LBRR;

    /*******************************************/
    /* Control use of inband LBRR              */
    /*******************************************/
    if( psEnc->sCmn.LBRR_enabled && psEnc->sCmn.speech_activity_Q8 > SILK_FIX_CONST( LBRR_SPEECH_ACTIVITY_THRES, 8 ) ) {
        psEnc->sCmn.LBRR_flags[ psEnc->sCmn.nFramesEncoded ] = 1;

        /* Copy noise shaping quantizer state and quantization indices from regular encoding */
        silk_memcpy( &sNSQ_LBRR, &psEnc->sCmn.sNSQ, sizeof( silk_nsq_state ) );
        silk_memcpy( psIndices_LBRR, &psEnc->sCmn.indices, sizeof( SideInfoIndices ) );

        /* Save original gains */
        silk_memcpy( TempGains, psEncCtrl->Gains, psEnc->sCmn.nb_subfr * sizeof( silk_float ) );

        if( psEnc->sCmn.nFramesEncoded == 0 || psEnc->sCmn.LBRR_flags[ psEnc->sCmn.nFramesEncoded - 1 ] == 0 ) {
            /* First frame in packet or previous frame not LBRR coded */
            psEnc->sCmn.LBRRprevLastGainIndex = psEnc->sShape.LastGainIndex;

            /* Increase Gains to get target LBRR rate */
            psIndices_LBRR->GainsIndices[ 0 ] += psEnc->sCmn.LBRR_GainIncreases;
            psIndices_LBRR->GainsIndices[ 0 ] = silk_min_int( psIndices_LBRR->GainsIndices[ 0 ], N_LEVELS_QGAIN - 1 );
        }

        /* Decode to get gains in sync with decoder */
        silk_gains_dequant( Gains_Q16, psIndices_LBRR->GainsIndices,
            &psEnc->sCmn.LBRRprevLastGainIndex, condCoding == CODE_CONDITIONALLY, psEnc->sCmn.nb_subfr );

        /* Overwrite unquantized gains with quantized gains and convert back to Q0 from Q16 */
        for( k = 0; k <  psEnc->sCmn.nb_subfr; k++ ) {
            psEncCtrl->Gains[ k ] = Gains_Q16[ k ] * ( 1.0f / 65536.0f );
        }

        /*****************************************/
        /* Noise shaping quantization            */
        /*****************************************/
        silk_NSQ_wrapper_FLP( psEnc, psEncCtrl, psIndices_LBRR, &sNSQ_LBRR,
            psEnc->sCmn.pulses_LBRR[ psEnc->sCmn.nFramesEncoded ], xfw );

        /* Restore original gains */
        silk_memcpy( psEncCtrl->Gains, TempGains, psEnc->sCmn.nb_subfr * sizeof( silk_float ) );
    }
}

/* Helper function, interpolates the filter taps */
static OPUS_INLINE void silk_LP_interpolate_filter_taps(
    opus_int32           B_Q28[ TRANSITION_NB ],
    opus_int32           A_Q28[ TRANSITION_NA ],
    const opus_int       ind,
    const opus_int32     fac_Q16
)
{
    opus_int nb, na;

    if( ind < TRANSITION_INT_NUM - 1 ) {
        if( fac_Q16 > 0 ) {
            if( fac_Q16 < 32768 ) { /* fac_Q16 is in range of a 16-bit int */
                /* Piece-wise linear interpolation of B and A */
                for( nb = 0; nb < TRANSITION_NB; nb++ ) {
                    B_Q28[ nb ] = silk_SMLAWB(
                        silk_Transition_LP_B_Q28[ ind     ][ nb ],
                        silk_Transition_LP_B_Q28[ ind + 1 ][ nb ] -
                        silk_Transition_LP_B_Q28[ ind     ][ nb ],
                        fac_Q16 );
                }
                for( na = 0; na < TRANSITION_NA; na++ ) {
                    A_Q28[ na ] = silk_SMLAWB(
                        silk_Transition_LP_A_Q28[ ind     ][ na ],
                        silk_Transition_LP_A_Q28[ ind + 1 ][ na ] -
                        silk_Transition_LP_A_Q28[ ind     ][ na ],
                        fac_Q16 );
                }
            } else { /* ( fac_Q16 - ( 1 << 16 ) ) is in range of a 16-bit int */
                silk_assert( fac_Q16 - ( 1 << 16 ) == silk_SAT16( fac_Q16 - ( 1 << 16 ) ) );
                /* Piece-wise linear interpolation of B and A */
                for( nb = 0; nb < TRANSITION_NB; nb++ ) {
                    B_Q28[ nb ] = silk_SMLAWB(
                        silk_Transition_LP_B_Q28[ ind + 1 ][ nb ],
                        silk_Transition_LP_B_Q28[ ind + 1 ][ nb ] -
                        silk_Transition_LP_B_Q28[ ind     ][ nb ],
                        fac_Q16 - ( (opus_int32)1 << 16 ) );
                }
                for( na = 0; na < TRANSITION_NA; na++ ) {
                    A_Q28[ na ] = silk_SMLAWB(
                        silk_Transition_LP_A_Q28[ ind + 1 ][ na ],
                        silk_Transition_LP_A_Q28[ ind + 1 ][ na ] -
                        silk_Transition_LP_A_Q28[ ind     ][ na ],
                        fac_Q16 - ( (opus_int32)1 << 16 ) );
                }
            }
        } else {
            silk_memcpy( B_Q28, silk_Transition_LP_B_Q28[ ind ], TRANSITION_NB * sizeof( opus_int32 ) );
            silk_memcpy( A_Q28, silk_Transition_LP_A_Q28[ ind ], TRANSITION_NA * sizeof( opus_int32 ) );
        }
    } else {
        silk_memcpy( B_Q28, silk_Transition_LP_B_Q28[ TRANSITION_INT_NUM - 1 ], TRANSITION_NB * sizeof( opus_int32 ) );
        silk_memcpy( A_Q28, silk_Transition_LP_A_Q28[ TRANSITION_INT_NUM - 1 ], TRANSITION_NA * sizeof( opus_int32 ) );
    }
}

/* Convert adaptive Mid/Side representation to Left/Right stereo signal */
void silk_stereo_MS_to_LR(
    stereo_dec_state            *state,                         /* I/O  State                                       */
    opus_int16                  x1[],                           /* I/O  Left input signal, becomes mid signal       */
    opus_int16                  x2[],                           /* I/O  Right input signal, becomes side signal     */
    const opus_int32            pred_Q13[],                     /* I    Predictors                                  */
    opus_int                    fs_kHz,                         /* I    Samples rate (kHz)                          */
    opus_int                    frame_length                    /* I    Number of samples                           */
)
{
    opus_int   n, denom_Q16, delta0_Q13, delta1_Q13;
    opus_int32 sum, diff, pred0_Q13, pred1_Q13;

    /* Buffering */
    silk_memcpy( x1, state->sMid,  2 * sizeof( opus_int16 ) );
    silk_memcpy( x2, state->sSide, 2 * sizeof( opus_int16 ) );
    silk_memcpy( state->sMid,  &x1[ frame_length ], 2 * sizeof( opus_int16 ) );
    silk_memcpy( state->sSide, &x2[ frame_length ], 2 * sizeof( opus_int16 ) );

    /* Interpolate predictors and add prediction to side channel */
    pred0_Q13  = state->pred_prev_Q13[ 0 ];
    pred1_Q13  = state->pred_prev_Q13[ 1 ];
    denom_Q16  = silk_DIV32_16( (opus_int32)1 << 16, STEREO_INTERP_LEN_MS * fs_kHz );
    delta0_Q13 = silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 0 ] - state->pred_prev_Q13[ 0 ], denom_Q16 ), 16 );
    delta1_Q13 = silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 1 ] - state->pred_prev_Q13[ 1 ], denom_Q16 ), 16 );
    for( n = 0; n < STEREO_INTERP_LEN_MS * fs_kHz; n++ ) {
        pred0_Q13 += delta0_Q13;
        pred1_Q13 += delta1_Q13;
        sum = silk_LSHIFT( silk_ADD_LSHIFT( x1[ n ] + x1[ n + 2 ], x1[ n + 1 ], 1 ), 9 );       /* Q11 */
        sum = silk_SMLAWB( silk_LSHIFT( (opus_int32)x2[ n + 1 ], 8 ), sum, pred0_Q13 );         /* Q8  */
        sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)x1[ n + 1 ], 11 ), pred1_Q13 );        /* Q8  */
        x2[ n + 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) );
    }
    pred0_Q13 = pred_Q13[ 0 ];
    pred1_Q13 = pred_Q13[ 1 ];
    for( n = STEREO_INTERP_LEN_MS * fs_kHz; n < frame_length; n++ ) {
        sum = silk_LSHIFT( silk_ADD_LSHIFT( x1[ n ] + x1[ n + 2 ], x1[ n + 1 ], 1 ), 9 );       /* Q11 */
        sum = silk_SMLAWB( silk_LSHIFT( (opus_int32)x2[ n + 1 ], 8 ), sum, pred0_Q13 );         /* Q8  */
        sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)x1[ n + 1 ], 11 ), pred1_Q13 );        /* Q8  */
        x2[ n + 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) );
    }
    state->pred_prev_Q13[ 0 ] = pred_Q13[ 0 ];
    state->pred_prev_Q13[ 1 ] = pred_Q13[ 1 ];

    /* Convert to left/right signals */
    for( n = 0; n < frame_length; n++ ) {
        sum  = x1[ n + 1 ] + (opus_int32)x2[ n + 1 ];
        diff = x1[ n + 1 ] - (opus_int32)x2[ n + 1 ];
        x1[ n + 1 ] = (opus_int16)silk_SAT16( sum );
        x2[ n + 1 ] = (opus_int16)silk_SAT16( diff );
    }
}

/* Resample with a 2nd order AR filter followed by FIR interpolation */
void silk_resampler_private_down_FIR(
    void                            *SS,            /* I/O  Resampler state             */
    opus_int16                      out[],          /* O    Output signal               */
    const opus_int16                in[],           /* I    Input signal                */
    opus_int32                      inLen           /* I    Number of input samples     */
)
{
    silk_resampler_state_struct *S = (silk_resampler_state_struct *)SS;
    opus_int32 nSamplesIn;
    opus_int32 max_index_Q16, index_increment_Q16;
    VARDECL( opus_int32, buf );
    const opus_int16 *FIR_Coefs;
    SAVE_STACK;

    ALLOC( buf, S->batchSize + S->FIR_Order, opus_int32 );

    /* Copy buffered samples to start of buffer */
    silk_memcpy( buf, S->sFIR.i32, S->FIR_Order * sizeof( opus_int32 ) );

    FIR_Coefs = &S->Coefs[ 2 ];

    /* Iterate over blocks of frameSizeIn input samples */
    index_increment_Q16 = S->invRatio_Q16;
    while( 1 ) {
        nSamplesIn = silk_min( inLen, S->batchSize );

        /* Second-order AR filter (output in Q8) */
        silk_resampler_private_AR2( S->sIIR, &buf[ S->FIR_Order ], in, S->Coefs, nSamplesIn );

        max_index_Q16 = silk_LSHIFT32( nSamplesIn, 16 );

        /* Interpolate filtered signal */
        out = silk_resampler_private_down_FIR_INTERPOL( out, buf, FIR_Coefs, S->FIR_Order,
            S->FIR_Fracs, max_index_Q16, index_increment_Q16 );

        in += nSamplesIn;
        inLen -= nSamplesIn;

        if( inLen > 1 ) {
            /* More iterations to do; copy last part of filtered signal to beginning of buffer */
            silk_memcpy( buf, &buf[ nSamplesIn ], S->FIR_Order * sizeof( opus_int32 ) );
        } else {
            break;
        }
    }

    /* Copy last part of filtered signal to the state for the next call */
    silk_memcpy( S->sFIR.i32, &buf[ nSamplesIn ], S->FIR_Order * sizeof( opus_int32 ) );
    RESTORE_STACK;
}

/* Upsample using a combination of allpass-based 2x upsampling and FIR interpolation */
void silk_resampler_private_IIR_FIR(
    void                            *SS,            /* I/O  Resampler state             */
    opus_int16                      out[],          /* O    Output signal               */
    const opus_int16                in[],           /* I    Input signal                */
    opus_int32                      inLen           /* I    Number of input samples     */
)
{
    silk_resampler_state_struct *S = (silk_resampler_state_struct *)SS;
    opus_int32 nSamplesIn;
    opus_int32 max_index_Q16, index_increment_Q16;
/*    VARDECL( opus_int16, buf );
    SAVE_STACK; */

/*    ALLOC( buf, 2 * S->batchSize + RESAMPLER_ORDER_FIR_12, opus_int16 ); */

    /* worst case = 2*16*10+8 = 328 * 2 = 656bytes */
    opus_int16 buf[2 * S->batchSize + RESAMPLER_ORDER_FIR_12];

    /* Copy buffered samples to start of buffer */
    silk_memcpy( buf, S->sFIR.i16, RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) );

    /* Iterate over blocks of frameSizeIn input samples */
    index_increment_Q16 = S->invRatio_Q16;
    while( 1 ) {
        nSamplesIn = silk_min( inLen, S->batchSize );

        /* Upsample 2x */
        silk_resampler_private_up2_HQ( S->sIIR, &buf[ RESAMPLER_ORDER_FIR_12 ], in, nSamplesIn );

        max_index_Q16 = silk_LSHIFT32( nSamplesIn, 16 + 1 );         /* + 1 because 2x upsampling */
        out = silk_resampler_private_IIR_FIR_INTERPOL( out, buf, max_index_Q16, index_increment_Q16 );
        in += nSamplesIn;
        inLen -= nSamplesIn;

        if( inLen > 0 ) {
            /* More iterations to do; copy last part of filtered signal to beginning of buffer */
            silk_memcpy( buf, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) );
        } else {
            break;
        }
    }

    /* Copy last part of filtered signal to the state for the next call */
    silk_memcpy( S->sFIR.i16, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) );
/*    RESTORE_STACK; */
}

/* Input and output sampling rate are at most 48000 Hz  */
opus_int silk_resampler(
    silk_resampler_state_struct *S,                 /* I/O  Resampler state                                             */
    opus_int16                  out[],              /* O    Output signal                                               */
    const opus_int16            in[],               /* I    Input signal                                                */
    opus_int32                  inLen               /* I    Number of input samples                                     */
)
{
    opus_int nSamples;

    /* Need at least 1 ms of input data */
    silk_assert( inLen >= S->Fs_in_kHz );
    /* Delay can't exceed the 1 ms of buffering */
    silk_assert( S->inputDelay <= S->Fs_in_kHz );

    nSamples = S->Fs_in_kHz - S->inputDelay;

    /* Copy to delay buffer */
    silk_memcpy( &S->delayBuf[ S->inputDelay ], in, nSamples * sizeof( opus_int16 ) );

    switch( S->resampler_function ) {
        case USE_silk_resampler_private_up2_HQ_wrapper:
            silk_resampler_private_up2_HQ_wrapper( S, out, S->delayBuf, S->Fs_in_kHz );
            silk_resampler_private_up2_HQ_wrapper( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz );
            break;
        case USE_silk_resampler_private_IIR_FIR:
            silk_resampler_private_IIR_FIR( S, out, S->delayBuf, S->Fs_in_kHz );
            silk_resampler_private_IIR_FIR( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz );
            break;
        case USE_silk_resampler_private_down_FIR:
            silk_resampler_private_down_FIR( S, out, S->delayBuf, S->Fs_in_kHz );
            silk_resampler_private_down_FIR( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz );
            break;
        default:
            silk_memcpy( out, S->delayBuf, S->Fs_in_kHz * sizeof( opus_int16 ) );
            silk_memcpy( &out[ S->Fs_out_kHz ], &in[ nSamples ], ( inLen - S->Fs_in_kHz ) * sizeof( opus_int16 ) );
    }

    /* Copy to delay buffer */
    silk_memcpy( S->delayBuf, &in[ inLen - S->inputDelay ], S->inputDelay * sizeof( opus_int16 ) );

    return 0;
}

/* Resampler: convert from one sampling rate to another */
opus_int silk_resampler(
    silk_resampler_state_struct *S,                 /* I/O   Resampler state                                            */
    opus_int16                  out[],              /* O     Output signal                                              */
    const opus_int16            in[],               /* I     Input signal                                               */
    opus_int32                  inLen               /* I     Number of input samples                                    */
)
{
    /* Input and output sampling rate are at most 48000 Hz */
    switch( S->resampler_function ) {
    case USE_silk_resampler_private_up2_HQ_wrapper:
        silk_resampler_private_up2_HQ_wrapper( S, out, in, inLen );
        break;
    case USE_silk_resampler_private_IIR_FIR:
        silk_resampler_private_IIR_FIR( S, out, in, inLen );
        break;
    case USE_silk_resampler_private_down_FIR:
        silk_resampler_private_down_FIR( S, out, in, inLen );
        break;
    default:
        silk_memcpy( out, in, inLen * sizeof( opus_int16 ) );
    }
    return 0;
}

/* this code is based on silk_a2k_FLP()                                 */
silk_float silk_LPC_inverse_pred_gain_FLP(  /* O    return inverse prediction gain, energy domain               */
        const silk_float *A,                 /* I    prediction coefficients [order]                             */
        opus_int32 order               /* I    prediction order                                            */
) {
    opus_int k, n;
    double invGain, rc, rc_mult1, rc_mult2;
    silk_float Atmp[2][SILK_MAX_ORDER_LPC];
    silk_float *Aold, *Anew;

    Anew = Atmp[order & 1];
    silk_memcpy(Anew, A, order * sizeof(silk_float));

    invGain = 1.0;
    for (k = order - 1; k > 0; k--) {
        rc = -Anew[k];
        if (rc > RC_THRESHOLD || rc < -RC_THRESHOLD) {
            return 0.0f;
        }
        rc_mult1 = 1.0f - rc * rc;
        rc_mult2 = 1.0f / rc_mult1;
        invGain *= rc_mult1;
        /* swap pointers */
        Aold = Anew;
        Anew = Atmp[k & 1];
        for (n = 0; n < k; n++) {
            Anew[n] = (silk_float)((Aold[n] - Aold[k - n - 1] * rc) * rc_mult2);
        }
    }
    rc = -Anew[0];
    if (rc > RC_THRESHOLD || rc < -RC_THRESHOLD) {
        return 0.0f;
    }
    rc_mult1 = 1.0f - rc * rc;
    invGain *= rc_mult1;
    return (silk_float) invGain;
}

//.........这里部分代码省略.........
        SNR_adj_dB += SPARSE_SNR_INCR_dB * ( psEncCtrl->sparseness - 0.5f );
    }

    /*******************************/
    /* Control bandwidth expansion */
    /*******************************/
    /* More BWE for signals with high prediction gain */
    strength = FIND_PITCH_WHITE_NOISE_FRACTION * psEncCtrl->predGain;           /* between 0.0 and 1.0 */
    BWExp1 = BWExp2 = BANDWIDTH_EXPANSION / ( 1.0f + strength * strength );
    delta  = LOW_RATE_BANDWIDTH_EXPANSION_DELTA * ( 1.0f - 0.75f * psEncCtrl->coding_quality );
    BWExp1 -= delta;
    BWExp2 += delta;
    /* BWExp1 will be applied after BWExp2, so make it relative */
    BWExp1 /= BWExp2;

    if( psEnc->sCmn.warping_Q16 > 0 ) {
        /* Slightly more warping in analysis will move quantization noise up in frequency, where it's better masked */
        warping = (silk_float)psEnc->sCmn.warping_Q16 / 65536.0f + 0.01f * psEncCtrl->coding_quality;
    } else {
        warping = 0.0f;
    }

    /********************************************/
    /* Compute noise shaping AR coefs and gains */
    /********************************************/
    for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) {
        /* Apply window: sine slope followed by flat part followed by cosine slope */
        opus_int shift, slope_part, flat_part;
        flat_part = psEnc->sCmn.fs_kHz * 3;
        slope_part = ( psEnc->sCmn.shapeWinLength - flat_part ) / 2;

        silk_apply_sine_window_FLP( x_windowed, x_ptr, 1, slope_part );
        shift = slope_part;
        silk_memcpy( x_windowed + shift, x_ptr + shift, flat_part * sizeof(silk_float) );
        shift += flat_part;
        silk_apply_sine_window_FLP( x_windowed + shift, x_ptr + shift, 2, slope_part );

        /* Update pointer: next LPC analysis block */
        x_ptr += psEnc->sCmn.subfr_length;

        if( psEnc->sCmn.warping_Q16 > 0 ) {
            /* Calculate warped auto correlation */
            silk_warped_autocorrelation_FLP( auto_corr, x_windowed, warping,
                psEnc->sCmn.shapeWinLength, psEnc->sCmn.shapingLPCOrder );
        } else {
            /* Calculate regular auto correlation */
            silk_autocorrelation_FLP( auto_corr, x_windowed, psEnc->sCmn.shapeWinLength, psEnc->sCmn.shapingLPCOrder + 1 );
        }

        /* Add white noise, as a fraction of energy */
        auto_corr[ 0 ] += auto_corr[ 0 ] * SHAPE_WHITE_NOISE_FRACTION;

        /* Convert correlations to prediction coefficients, and compute residual energy */
        nrg = silk_levinsondurbin_FLP( &psEncCtrl->AR2[ k * MAX_SHAPE_LPC_ORDER ], auto_corr, psEnc->sCmn.shapingLPCOrder );
        psEncCtrl->Gains[ k ] = ( silk_float )sqrt( nrg );

        if( psEnc->sCmn.warping_Q16 > 0 ) {
            /* Adjust gain for warping */
            psEncCtrl->Gains[ k ] *= warped_gain( &psEncCtrl->AR2[ k * MAX_SHAPE_LPC_ORDER ], warping, psEnc->sCmn.shapingLPCOrder );
        }

        /* Bandwidth expansion for synthesis filter shaping */
        silk_bwexpander_FLP( &psEncCtrl->AR2[ k * MAX_SHAPE_LPC_ORDER ], psEnc->sCmn.shapingLPCOrder, BWExp2 );

        /* Compute noise shaping filter coefficients */
        silk_memcpy(

static inline void silk_PLC_conceal(
    silk_decoder_state                  *psDec,             /* I/O Decoder state        */
    silk_decoder_control                *psDecCtrl,         /* I/O Decoder control      */
    opus_int16                          frame[]             /* O LPC residual signal    */
)
{
    opus_int   i, j, k;
    opus_int   lag, idx, sLTP_buf_idx, shift1, shift2;
    opus_int32 rand_seed, harm_Gain_Q15, rand_Gain_Q15, inv_gain_Q16, inv_gain_Q30;
    opus_int32 energy1, energy2, *rand_ptr, *pred_lag_ptr;
    opus_int32 LPC_exc_Q14, LPC_pred_Q10, LTP_pred_Q12;
    opus_int16 rand_scale_Q14;
    opus_int16 *B_Q14, *exc_buf_ptr;
    opus_int32 *sLPC_Q14_ptr;
    opus_int16 exc_buf[ 2 * MAX_SUB_FRAME_LENGTH ];
    opus_int16 A_Q12[ MAX_LPC_ORDER ];
    opus_int16 sLTP[ MAX_FRAME_LENGTH ];
    opus_int32 sLTP_Q14[ 2 * MAX_FRAME_LENGTH ];
    silk_PLC_struct *psPLC = &psDec->sPLC;

    if (psDec->first_frame_after_reset)
       silk_memset(psPLC->prevLPC_Q12, 0, MAX_LPC_ORDER*sizeof(psPLC->prevLPC_Q12[ 0 ]));

    /* Find random noise component */
    /* Scale previous excitation signal */
    exc_buf_ptr = exc_buf;
    for( k = 0; k < 2; k++ ) {
        for( i = 0; i < psPLC->subfr_length; i++ ) {
            exc_buf_ptr[ i ] = ( opus_int16 )silk_RSHIFT(
                silk_SMULWW( psDec->exc_Q10[ i + ( k + psPLC->nb_subfr - 2 ) * psPLC->subfr_length ], psPLC->prevGain_Q16[ k ] ), 10 );
        }
        exc_buf_ptr += psPLC->subfr_length;
    }
    /* Find the subframe with lowest energy of the last two and use that as random noise generator */
    silk_sum_sqr_shift( &energy1, &shift1, exc_buf,                         psPLC->subfr_length );
    silk_sum_sqr_shift( &energy2, &shift2, &exc_buf[ psPLC->subfr_length ], psPLC->subfr_length );

    if( silk_RSHIFT( energy1, shift2 ) < silk_RSHIFT( energy2, shift1 ) ) {
        /* First sub-frame has lowest energy */
        rand_ptr = &psDec->exc_Q10[ silk_max_int( 0, ( psPLC->nb_subfr - 1 ) * psPLC->subfr_length - RAND_BUF_SIZE ) ];
    } else {
        /* Second sub-frame has lowest energy */
        rand_ptr = &psDec->exc_Q10[ silk_max_int( 0, psPLC->nb_subfr * psPLC->subfr_length - RAND_BUF_SIZE ) ];
    }

    /* Setup Gain to random noise component */
    B_Q14          = psPLC->LTPCoef_Q14;
    rand_scale_Q14 = psPLC->randScale_Q14;

    /* Setup attenuation gains */
    harm_Gain_Q15 = HARM_ATT_Q15[ silk_min_int( NB_ATT - 1, psDec->lossCnt ) ];
    if( psDec->prevSignalType == TYPE_VOICED ) {
        rand_Gain_Q15 = PLC_RAND_ATTENUATE_V_Q15[  silk_min_int( NB_ATT - 1, psDec->lossCnt ) ];
    } else {
        rand_Gain_Q15 = PLC_RAND_ATTENUATE_UV_Q15[ silk_min_int( NB_ATT - 1, psDec->lossCnt ) ];
    }

    /* LPC concealment. Apply BWE to previous LPC */
    silk_bwexpander( psPLC->prevLPC_Q12, psDec->LPC_order, SILK_FIX_CONST( BWE_COEF, 16 ) );

    /* Preload LPC coeficients to array on stack. Gives small performance gain */
    silk_memcpy( A_Q12, psPLC->prevLPC_Q12, psDec->LPC_order * sizeof( opus_int16 ) );

    /* First Lost frame */
    if( psDec->lossCnt == 0 ) {
        rand_scale_Q14 = 1 << 14;

        /* Reduce random noise Gain for voiced frames */
        if( psDec->prevSignalType == TYPE_VOICED ) {
            for( i = 0; i < LTP_ORDER; i++ ) {
                rand_scale_Q14 -= B_Q14[ i ];
            }
            rand_scale_Q14 = silk_max_16( 3277, rand_scale_Q14 ); /* 0.2 */
            rand_scale_Q14 = ( opus_int16 )silk_RSHIFT( silk_SMULBB( rand_scale_Q14, psPLC->prevLTP_scale_Q14 ), 14 );
        } else {
            /* Reduce random noise for unvoiced frames with high LPC gain */
            opus_int32 invGain_Q30, down_scale_Q30;

            silk_LPC_inverse_pred_gain( &invGain_Q30, psPLC->prevLPC_Q12, psDec->LPC_order );

            down_scale_Q30 = silk_min_32( silk_RSHIFT( 1 << 30, LOG2_INV_LPC_GAIN_HIGH_THRES ), invGain_Q30 );
            down_scale_Q30 = silk_max_32( silk_RSHIFT( 1 << 30, LOG2_INV_LPC_GAIN_LOW_THRES ), down_scale_Q30 );
            down_scale_Q30 = silk_LSHIFT( down_scale_Q30, LOG2_INV_LPC_GAIN_HIGH_THRES );

            rand_Gain_Q15 = silk_RSHIFT( silk_SMULWB( down_scale_Q30, rand_Gain_Q15 ), 14 );
        }
    }

    rand_seed    = psPLC->rand_seed;
    lag          = silk_RSHIFT_ROUND( psPLC->pitchL_Q8, 8 );
    sLTP_buf_idx = psDec->ltp_mem_length;

    /* Rewhiten LTP state */
    idx = psDec->ltp_mem_length - lag - psDec->LPC_order - LTP_ORDER / 2;
    silk_assert( idx > 0 );
    silk_LPC_analysis_filter( &sLTP[ idx ], &psDec->outBuf[ idx ], A_Q12, psDec->ltp_mem_length - idx, psDec->LPC_order );
    /* Scale LTP state */
    inv_gain_Q16 = silk_INVERSE32_varQ( psPLC->prevGain_Q16[ 1 ], 32 );
    inv_gain_Q16 = silk_min( inv_gain_Q16, silk_int16_MAX );
    inv_gain_Q30 = silk_LSHIFT( inv_gain_Q16, 14 );
//.........这里部分代码省略.........

/* Limit, stabilize, convert and quantize NLSFs */
void silk_process_NLSFs(
    silk_encoder_state          *psEncC,                            /* I/O  Encoder state                               */
    opus_int16                  PredCoef_Q12[ 2 ][ MAX_LPC_ORDER ], /* O    Prediction coefficients                     */
    opus_int16                  pNLSF_Q15[         MAX_LPC_ORDER ], /* I/O  Normalized LSFs (quant out) (0 - (2^15-1))  */
    const opus_int16            prev_NLSFq_Q15[    MAX_LPC_ORDER ]  /* I    Previous Normalized LSFs (0 - (2^15-1))     */
)
{
    opus_int     i, doInterpolate;
    opus_int     NLSF_mu_Q20;
    opus_int32   i_sqr_Q15;
    opus_int16   pNLSF0_temp_Q15[ MAX_LPC_ORDER ];
    opus_int16   pNLSFW_QW[ MAX_LPC_ORDER ];
    opus_int16   pNLSFW0_temp_QW[ MAX_LPC_ORDER ];

    silk_assert(psEncC->speech_activity_Q8 >=   0);
    silk_assert(psEncC->speech_activity_Q8 <= SILK_FIX_CONST(1.0, 8));
    silk_assert(psEncC->useInterpolatedNLSFs == 1 || psEncC->indices.NLSFInterpCoef_Q2 == (1 << 2));

    /***********************/
    /* Calculate mu values */
    /***********************/
    /* NLSF_mu  = 0.003 - 0.0015 * psEnc->speech_activity; */
    NLSF_mu_Q20 = silk_SMLAWB(SILK_FIX_CONST(0.003, 20), SILK_FIX_CONST(-0.001, 28), psEncC->speech_activity_Q8);
    if(psEncC->nb_subfr == 2) {
        /* Multiply by 1.5 for 10 ms packets */
        NLSF_mu_Q20 = silk_ADD_RSHIFT(NLSF_mu_Q20, NLSF_mu_Q20, 1);
    }

    silk_assert(NLSF_mu_Q20 >  0);
    silk_assert(NLSF_mu_Q20 <= SILK_FIX_CONST(0.005, 20));

    /* Calculate NLSF weights */
    silk_NLSF_VQ_weights_laroia(pNLSFW_QW, pNLSF_Q15, psEncC->predictLPCOrder);

    /* Update NLSF weights for interpolated NLSFs */
    doInterpolate = (psEncC->useInterpolatedNLSFs == 1) && (psEncC->indices.NLSFInterpCoef_Q2 < 4);
    if(doInterpolate) {
        /* Calculate the interpolated NLSF vector for the first half */
        silk_interpolate(pNLSF0_temp_Q15, prev_NLSFq_Q15, pNLSF_Q15,
            psEncC->indices.NLSFInterpCoef_Q2, psEncC->predictLPCOrder);

        /* Calculate first half NLSF weights for the interpolated NLSFs */
        silk_NLSF_VQ_weights_laroia(pNLSFW0_temp_QW, pNLSF0_temp_Q15, psEncC->predictLPCOrder);

        /* Update NLSF weights with contribution from first half */
        i_sqr_Q15 = silk_LSHIFT(silk_SMULBB(psEncC->indices.NLSFInterpCoef_Q2, psEncC->indices.NLSFInterpCoef_Q2), 11);
        for(i = 0; i < psEncC->predictLPCOrder; i++) {
            pNLSFW_QW[ i ] = silk_SMLAWB(silk_RSHIFT(pNLSFW_QW[ i ], 1), (opus_int32)pNLSFW0_temp_QW[ i ], i_sqr_Q15);
            silk_assert(pNLSFW_QW[ i ] >= 1);
        }
    }

    silk_NLSF_encode(psEncC->indices.NLSFIndices, pNLSF_Q15, psEncC->psNLSF_CB, pNLSFW_QW,
        NLSF_mu_Q20, psEncC->NLSF_MSVQ_Survivors, psEncC->indices.signalType);

    /* Convert quantized NLSFs back to LPC coefficients */
    silk_NLSF2A(PredCoef_Q12[ 1 ], pNLSF_Q15, psEncC->predictLPCOrder);

    if(doInterpolate) {
        /* Calculate the interpolated, quantized LSF vector for the first half */
        silk_interpolate(pNLSF0_temp_Q15, prev_NLSFq_Q15, pNLSF_Q15,
            psEncC->indices.NLSFInterpCoef_Q2, psEncC->predictLPCOrder);

        /* Convert back to LPC coefficients */
        silk_NLSF2A(PredCoef_Q12[ 0 ], pNLSF0_temp_Q15, psEncC->predictLPCOrder);

    } else {
        /* Copy LPC coefficients for first half from second half */
        silk_memcpy(PredCoef_Q12[ 0 ], PredCoef_Q12[ 1 ], psEncC->predictLPCOrder * sizeof(opus_int16));
    }
}

opus_int silk_encode_frame_FLP(
    silk_encoder_state_FLP          *psEnc,                             /* I/O  Encoder state FLP                           */
    opus_int32                      *pnBytesOut,                        /* O    Number of payload bytes;                    */
    ec_enc                          *psRangeEnc,                        /* I/O  compressor data structure                   */
    opus_int                        condCoding,                         /* I    The type of conditional coding to use       */
    opus_int                        maxBits,                            /* I    If > 0: maximum number of output bits       */
    opus_int                        useCBR                              /* I    Flag to force constant-bitrate operation    */
)
{
    silk_encoder_control_FLP sEncCtrl;
    opus_int     i, iter, maxIter, found_upper, found_lower, ret = 0;
    silk_float   *x_frame, *res_pitch_frame;
    silk_float   xfw[ MAX_FRAME_LENGTH ];
    silk_float   res_pitch[ 2 * MAX_FRAME_LENGTH + LA_PITCH_MAX ];
    ec_enc       sRangeEnc_copy, sRangeEnc_copy2;
    silk_nsq_state sNSQ_copy, sNSQ_copy2;
    opus_int32   seed_copy, nBits, nBits_lower, nBits_upper, gainMult_lower, gainMult_upper;
    opus_int32   gainsID, gainsID_lower, gainsID_upper;
    opus_int16   gainMult_Q8;
    opus_int16   ec_prevLagIndex_copy;
    opus_int     ec_prevSignalType_copy;
    opus_int8    LastGainIndex_copy2;
    opus_int32   pGains_Q16[ MAX_NB_SUBFR ];
    opus_uint8   ec_buf_copy[ 1275 ];

    /* This is totally unnecessary but many compilers (including gcc) are too dumb to realise it */
    LastGainIndex_copy2 = nBits_lower = nBits_upper = gainMult_lower = gainMult_upper = 0;

    psEnc->sCmn.indices.Seed = psEnc->sCmn.frameCounter++ & 3;

    /**************************************************************/
    /* Set up Input Pointers, and insert frame in input buffer    */
    /**************************************************************/
    /* pointers aligned with start of frame to encode */
    x_frame         = psEnc->x_buf + psEnc->sCmn.ltp_mem_length;    /* start of frame to encode */
    res_pitch_frame = res_pitch    + psEnc->sCmn.ltp_mem_length;    /* start of pitch LPC residual frame */

    /***************************************/
    /* Ensure smooth bandwidth transitions */
    /***************************************/
    silk_LP_variable_cutoff( &psEnc->sCmn.sLP, psEnc->sCmn.inputBuf + 1, psEnc->sCmn.frame_length );

    /*******************************************/
    /* Copy new frame to front of input buffer */
    /*******************************************/
    silk_short2float_array( x_frame + LA_SHAPE_MS * psEnc->sCmn.fs_kHz, psEnc->sCmn.inputBuf + 1, psEnc->sCmn.frame_length );

    /* Add tiny signal to avoid high CPU load from denormalized floating point numbers */
    for( i = 0; i < 8; i++ ) {
        x_frame[ LA_SHAPE_MS * psEnc->sCmn.fs_kHz + i * ( psEnc->sCmn.frame_length >> 3 ) ] += ( 1 - ( i & 2 ) ) * 1e-6f;
    }

    if( !psEnc->sCmn.prefillFlag ) {
        /*****************************************/
        /* Find pitch lags, initial LPC analysis */
        /*****************************************/
        silk_find_pitch_lags_FLP( psEnc, &sEncCtrl, res_pitch, x_frame );

        /************************/
        /* Noise shape analysis */
        /************************/
        silk_noise_shape_analysis_FLP( psEnc, &sEncCtrl, res_pitch_frame, x_frame );

        /***************************************************/
        /* Find linear prediction coefficients (LPC + LTP) */
        /***************************************************/
        silk_find_pred_coefs_FLP( psEnc, &sEncCtrl, res_pitch, x_frame, condCoding );

        /****************************************/
        /* Process gains                        */
        /****************************************/
        silk_process_gains_FLP( psEnc, &sEncCtrl, condCoding );

        /*****************************************/
        /* Prefiltering for noise shaper         */
        /*****************************************/
        silk_prefilter_FLP( psEnc, &sEncCtrl, xfw, x_frame );

        /****************************************/
        /* Low Bitrate Redundant Encoding       */
        /****************************************/
        silk_LBRR_encode_FLP( psEnc, &sEncCtrl, xfw, condCoding );

        /* Loop over quantizer and entroy coding to control bitrate */
        maxIter = 6;
        gainMult_Q8 = SILK_FIX_CONST( 1, 8 );
        found_lower = 0;
        found_upper = 0;
        gainsID = silk_gains_ID( psEnc->sCmn.indices.GainsIndices, psEnc->sCmn.nb_subfr );
        gainsID_lower = -1;
        gainsID_upper = -1;
        /* Copy part of the input state */
        silk_memcpy( &sRangeEnc_copy, psRangeEnc, sizeof( ec_enc ) );
        silk_memcpy( &sNSQ_copy, &psEnc->sCmn.sNSQ, sizeof( silk_nsq_state ) );
        seed_copy = psEnc->sCmn.indices.Seed;
        ec_prevLagIndex_copy = psEnc->sCmn.ec_prevLagIndex;
        ec_prevSignalType_copy = psEnc->sCmn.ec_prevSignalType;
        for( iter = 0; ; iter++ ) {
            if( gainsID == gainsID_lower ) {
                nBits = nBits_lower;
//.........这里部分代码省略.........

/* Downsample by a factor 2/3, low quality */
void silk_resampler_down2_3(
    opus_int32                  *S,                 /* I/O  State vector [ 6 ]                                          */
    opus_int16                  *out,               /* O    Output signal [ floor(2*inLen/3) ]                          */
    const opus_int16            *in,                /* I    Input signal [ inLen ]                                      */
    opus_int32                  inLen               /* I    Number of input samples                                     */
)
{
    opus_int32 nSamplesIn, counter, res_Q6;
    VARDECL( opus_int32, buf );
    opus_int32 *buf_ptr;
    SAVE_STACK;

    ALLOC( buf, RESAMPLER_MAX_BATCH_SIZE_IN + ORDER_FIR, opus_int32 );

    /* Copy buffered samples to start of buffer */
    silk_memcpy( buf, S, ORDER_FIR * sizeof( opus_int32 ) );

    /* Iterate over blocks of frameSizeIn input samples */
    while( 1 ) {
        nSamplesIn = silk_min( inLen, RESAMPLER_MAX_BATCH_SIZE_IN );

        /* Second-order AR filter (output in Q8) */
        silk_resampler_private_AR2( &S[ ORDER_FIR ], &buf[ ORDER_FIR ], in,
            silk_Resampler_2_3_COEFS_LQ, nSamplesIn );

        /* Interpolate filtered signal */
        buf_ptr = buf;
        counter = nSamplesIn;
        while( counter > 2 ) {
            /* Inner product */
            res_Q6 = silk_SMULWB(         buf_ptr[ 0 ], silk_Resampler_2_3_COEFS_LQ[ 2 ] );
            res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 1 ], silk_Resampler_2_3_COEFS_LQ[ 3 ] );
            res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 2 ], silk_Resampler_2_3_COEFS_LQ[ 5 ] );
            res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 3 ], silk_Resampler_2_3_COEFS_LQ[ 4 ] );

            /* Scale down, saturate and store in output array */
            *out++ = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( res_Q6, 6 ) );

            res_Q6 = silk_SMULWB(         buf_ptr[ 1 ], silk_Resampler_2_3_COEFS_LQ[ 4 ] );
            res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 2 ], silk_Resampler_2_3_COEFS_LQ[ 5 ] );
            res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 3 ], silk_Resampler_2_3_COEFS_LQ[ 3 ] );
            res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 4 ], silk_Resampler_2_3_COEFS_LQ[ 2 ] );

            /* Scale down, saturate and store in output array */
            *out++ = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( res_Q6, 6 ) );

            buf_ptr += 3;
            counter -= 3;
        }

        in += nSamplesIn;
        inLen -= nSamplesIn;

        if( inLen > 0 ) {
            /* More iterations to do; copy last part of filtered signal to beginning of buffer */
            silk_memcpy( buf, &buf[ nSamplesIn ], ORDER_FIR * sizeof( opus_int32 ) );
        } else {
            break;
        }
    }

    /* Copy last part of filtered signal to the state for the next call */
    silk_memcpy( S, &buf[ nSamplesIn ], ORDER_FIR * sizeof( opus_int32 ) );
    RESTORE_STACK;
}

opus_int silk_encode_frame_FIX(
    silk_encoder_state_FIX          *psEnc,                                 /* I/O  Pointer to Silk FIX encoder state                                           */
    opus_int32                      *pnBytesOut,                            /* O    Pointer to number of payload bytes;                                         */
    ec_enc                          *psRangeEnc,                            /* I/O  compressor data structure                                                   */
    opus_int                        condCoding,                             /* I    The type of conditiona