예제 #1
0
/* 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 */
    SKP_memcpy( x1, state->sMid,  2 * sizeof( opus_int16 ) );
    SKP_memcpy( x2, state->sSide, 2 * sizeof( opus_int16 ) );
    SKP_memcpy( state->sMid,  &x1[ frame_length ], 2 * sizeof( opus_int16 ) );
    SKP_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  = SKP_DIV32_16( 1 << 16, STEREO_INTERP_LEN_MS * fs_kHz );
    delta0_Q13 = SKP_RSHIFT_ROUND( SKP_SMULBB( pred_Q13[ 0 ] - state->pred_prev_Q13[ 0 ], denom_Q16 ), 16 );
    delta1_Q13 = SKP_RSHIFT_ROUND( SKP_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 = SKP_LSHIFT( SKP_ADD_LSHIFT( x1[ n ] + x1[ n + 2 ], x1[ n + 1 ], 1 ), 9 );         /* Q11 */ 
        sum = SKP_SMLAWB( SKP_LSHIFT( ( opus_int32 )x2[ n + 1 ], 8 ), sum, pred0_Q13 );          /* Q8  */
        sum = SKP_SMLAWB( sum, SKP_LSHIFT( ( opus_int32 )x1[ n + 1 ], 11 ), pred1_Q13 );         /* Q8  */
        x2[ n + 1 ] = (opus_int16)SKP_SAT16( SKP_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 = SKP_LSHIFT( SKP_ADD_LSHIFT( x1[ n ] + x1[ n + 2 ], x1[ n + 1 ], 1 ), 9 );         /* Q11 */ 
        sum = SKP_SMLAWB( SKP_LSHIFT( ( opus_int32 )x2[ n + 1 ], 8 ), sum, pred0_Q13 );          /* Q8  */
        sum = SKP_SMLAWB( sum, SKP_LSHIFT( ( opus_int32 )x1[ n + 1 ], 11 ), pred1_Q13 );         /* Q8  */
        x2[ n + 1 ] = (opus_int16)SKP_SAT16( SKP_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)SKP_SAT16( sum );
        x2[ n + 1 ] = (opus_int16)SKP_SAT16( diff );
    }
}
예제 #2
0
/* Encodes signs of excitation */
void silk_encode_signs(
    ec_enc                      *psRangeEnc,                        /* I/O  Compressor data structure                   */
    const SKP_int8              pulses[],                           /* I    pulse signal                                */
    SKP_int                     length,                             /* I    length of input                             */
    const SKP_int               signalType,                         /* I    Signal type                                 */
    const SKP_int               quantOffsetType,                    /* I    Quantization offset type                    */
    const SKP_int               sum_pulses[ MAX_NB_SHELL_BLOCKS ]   /* I    Sum of absolute pulses per block            */
)
{
    SKP_int         i, j, p;
    SKP_uint8       icdf[ 2 ];
    const SKP_int8  *q_ptr;
    const SKP_uint8 *icdf_ptr;

    icdf[ 1 ] = 0;
    q_ptr = pulses;
    i = SKP_SMULBB( 6, SKP_ADD_LSHIFT( quantOffsetType, signalType, 1 ) );
    icdf_ptr = &silk_sign_iCDF[ i ];
    length = SKP_RSHIFT( length + SHELL_CODEC_FRAME_LENGTH/2, LOG2_SHELL_CODEC_FRAME_LENGTH );
    for( i = 0; i < length; i++ ) {
        p = sum_pulses[ i ];
        if( p > 0 ) {
            icdf[ 0 ] = icdf_ptr[ SKP_min( p - 1, 5 ) ];
            for( j = 0; j < SHELL_CODEC_FRAME_LENGTH; j++ ) {
                if( q_ptr[ j ] != 0 ) {
                    ec_enc_icdf( psRangeEnc, SKP_enc_map( q_ptr[ j ]), icdf, 8 );
                }
            }
        }
        q_ptr += SHELL_CODEC_FRAME_LENGTH;
    }
}
예제 #3
0
/* Decodes signs of excitation */
void silk_decode_signs(
    ec_dec                      *psRangeDec,                        /* I/O  Compressor data structure                   */
    SKP_int                     pulses[],                           /* I/O  pulse signal                                */
    SKP_int                     length,                             /* I    length of input                             */
    const SKP_int               signalType,                         /* I    Signal type                                 */
    const SKP_int               quantOffsetType,                    /* I    Quantization offset type                    */
    const SKP_int               sum_pulses[ MAX_NB_SHELL_BLOCKS ]   /* I    Sum of absolute pulses per block            */
)
{
    SKP_int         i, j, p;
    SKP_uint8       icdf[ 2 ];
    SKP_int         *q_ptr;
    const SKP_uint8 *icdf_ptr;

    icdf[ 1 ] = 0;
    q_ptr = pulses;
    i = SKP_SMULBB( 6, SKP_ADD_LSHIFT( quantOffsetType, signalType, 1 ) );
    icdf_ptr = &silk_sign_iCDF[ i ];
    length = SKP_RSHIFT( length + SHELL_CODEC_FRAME_LENGTH/2, LOG2_SHELL_CODEC_FRAME_LENGTH );
    for( i = 0; i < length; i++ ) {
        p = sum_pulses[ i ];
        if( p > 0 ) {
            icdf[ 0 ] = icdf_ptr[ SKP_min( p - 1, 5 ) ];
            for( j = 0; j < SHELL_CODEC_FRAME_LENGTH; j++ ) {
                if( q_ptr[ j ] > 0 ) {
                    /* attach sign */
#if 0
                    /* conditional implementation */
                    if( ec_dec_icdf( psRangeDec, icdf, 8 ) == 0 ) {
                        q_ptr[ j ] = -q_ptr[ j ];
                    }
#else
                    /* implementation with shift, subtraction, multiplication */
                    q_ptr[ j ] *= SKP_dec_map( ec_dec_icdf( psRangeDec, icdf, 8 ) );
#endif
                }
            }
        }
        q_ptr += SHELL_CODEC_FRAME_LENGTH;
    }
}
/* Find pitch lags */
void SKP_Silk_find_pitch_lags_FIX(
    SKP_Silk_encoder_state_FIX      *psEnc,         /* I/O  encoder state                               */
    SKP_Silk_encoder_control_FIX    *psEncCtrl,     /* I/O  encoder control                             */
    SKP_int16                       res[],          /* O    residual                                    */
    const SKP_int16                 x[]             /* I    Speech signal                               */
)
{
    SKP_Silk_predict_state_FIX *psPredSt = &psEnc->sPred;
    SKP_int   buf_len, i, scale;
    SKP_int32 thrhld_Q15, res_nrg;
    const SKP_int16 *x_buf, *x_buf_ptr;
    SKP_int16 Wsig[      FIND_PITCH_LPC_WIN_MAX ], *Wsig_ptr;
    SKP_int32 auto_corr[ MAX_FIND_PITCH_LPC_ORDER + 1 ];
    SKP_int16 rc_Q15[    MAX_FIND_PITCH_LPC_ORDER ];
    SKP_int32 A_Q24[     MAX_FIND_PITCH_LPC_ORDER ];
    SKP_int32 FiltState[ MAX_FIND_PITCH_LPC_ORDER ];
    SKP_int16 A_Q12[     MAX_FIND_PITCH_LPC_ORDER ];

    /******************************************/
    /* Setup buffer lengths etc based on Fs   */
    /******************************************/
    buf_len = SKP_ADD_LSHIFT( psEnc->sCmn.la_pitch, psEnc->sCmn.frame_length, 1 );

    /* Safty check */
    SKP_assert( buf_len >= psPredSt->pitch_LPC_win_length );

    x_buf = x - psEnc->sCmn.frame_length;

    /*************************************/
    /* Estimate LPC AR coefficients      */
    /*************************************/

    /* Calculate windowed signal */

    /* First LA_LTP samples */
    x_buf_ptr = x_buf + buf_len - psPredSt->pitch_LPC_win_length;
    Wsig_ptr  = Wsig;
    SKP_Silk_apply_sine_window_new( Wsig_ptr, x_buf_ptr, 1, psEnc->sCmn.la_pitch );

    /* Middle un - windowed samples */
    Wsig_ptr  += psEnc->sCmn.la_pitch;
    x_buf_ptr += psEnc->sCmn.la_pitch;
    SKP_memcpy( Wsig_ptr, x_buf_ptr, ( psPredSt->pitch_LPC_win_length - SKP_LSHIFT( psEnc->sCmn.la_pitch, 1 ) ) * sizeof( SKP_int16 ) );

    /* Last LA_LTP samples */
    Wsig_ptr  += psPredSt->pitch_LPC_win_length - SKP_LSHIFT( psEnc->sCmn.la_pitch, 1 );
    x_buf_ptr += psPredSt->pitch_LPC_win_length - SKP_LSHIFT( psEnc->sCmn.la_pitch, 1 );
    SKP_Silk_apply_sine_window_new( Wsig_ptr, x_buf_ptr, 2, psEnc->sCmn.la_pitch );

    /* Calculate autocorrelation sequence */
    SKP_Silk_autocorr( auto_corr, &scale, Wsig, psPredSt->pitch_LPC_win_length, psEnc->sCmn.pitchEstimationLPCOrder + 1 );

    /* Add white noise, as fraction of energy */
    auto_corr[ 0 ] = SKP_SMLAWB( auto_corr[ 0 ], auto_corr[ 0 ], SKP_FIX_CONST( FIND_PITCH_WHITE_NOISE_FRACTION, 16 ) );

    /* Calculate the reflection coefficients using schur */
    res_nrg = SKP_Silk_schur( rc_Q15, auto_corr, psEnc->sCmn.pitchEstimationLPCOrder );

    /* Prediction gain */
    psEncCtrl->predGain_Q16 = SKP_DIV32_varQ( auto_corr[ 0 ], SKP_max_int( res_nrg, 1 ), 16 );

    /* Convert reflection coefficients to prediction coefficients */
    SKP_Silk_k2a( A_Q24, rc_Q15, psEnc->sCmn.pitchEstimationLPCOrder );

    /* Convert From 32 bit Q24 to 16 bit Q12 coefs */
    for( i = 0; i < psEnc->sCmn.pitchEstimationLPCOrder; i++ ) {
        A_Q12[ i ] = ( SKP_int16 )SKP_SAT16( SKP_RSHIFT( A_Q24[ i ], 12 ) );
    }

    /* Do BWE */
    SKP_Silk_bwexpander( A_Q12, psEnc->sCmn.pitchEstimationLPCOrder, SKP_FIX_CONST( FIND_PITCH_BANDWITH_EXPANSION, 16 ) );

    /*****************************************/
    /* LPC analysis filtering                */
    /*****************************************/
    SKP_memset( FiltState, 0, psEnc->sCmn.pitchEstimationLPCOrder * sizeof( SKP_int32 ) ); /* Not really necessary, but Valgrind will complain otherwise */
    SKP_Silk_MA_Prediction( x_buf, A_Q12, FiltState, res, buf_len, psEnc->sCmn.pitchEstimationLPCOrder );
    SKP_memset( res, 0, psEnc->sCmn.pitchEstimationLPCOrder * sizeof( SKP_int16 ) );

    /* Threshold for pitch estimator */
    thrhld_Q15 = SKP_FIX_CONST( 0.45, 15 );
    thrhld_Q15 = SKP_SMLABB( thrhld_Q15, SKP_FIX_CONST( -0.004, 15 ), psEnc->sCmn.pitchEstimationLPCOrder );
    thrhld_Q15 = SKP_SMLABB( thrhld_Q15, SKP_FIX_CONST( -0.1,   7  ), psEnc->speech_activity_Q8 );
    thrhld_Q15 = SKP_SMLABB( thrhld_Q15, SKP_FIX_CONST(  0.15,  15 ), psEnc->sCmn.prev_sigtype );
    thrhld_Q15 = SKP_SMLAWB( thrhld_Q15, SKP_FIX_CONST( -0.1,   16 ), psEncCtrl->input_tilt_Q15 );
    thrhld_Q15 = SKP_SAT16(  thrhld_Q15 );

    /*****************************************/
    /* Call pitch estimator                  */
    /*****************************************/
    psEncCtrl->sCmn.sigtype = SKP_Silk_pitch_analysis_core( res, psEncCtrl->sCmn.pitchL, &psEncCtrl->sCmn.lagIndex,
        &psEncCtrl->sCmn.contourIndex, &psEnc->LTPCorr_Q15, psEnc->sCmn.prevLag, psEnc->sCmn.pitchEstimationThreshold_Q16,
        ( SKP_int16 )thrhld_Q15, psEnc->sCmn.fs_kHz, psEnc->sCmn.pitchEstimationComplexity, SKP_FALSE );
}
예제 #5
0
/* Find pitch lags */
void SKP_Silk_find_pitch_lags_FIX(
    SKP_Silk_encoder_state_FIX      *psEnc,         /* I/O  encoder state                               */
    SKP_Silk_encoder_control_FIX    *psEncCtrl,     /* I/O  encoder control                             */
    SKP_int16                       res[],          /* O    residual                                    */
    const SKP_int16                 x[]             /* I    Speech signal                               */
)
{
    SKP_Silk_predict_state_FIX *psPredSt = &psEnc->sPred;
    SKP_int   buf_len, i;
    SKP_int32 scale;
    SKP_int32 thrhld_Q15;
    const SKP_int16 *x_buf, *x_buf_ptr;
    SKP_int16 Wsig[      FIND_PITCH_LPC_WIN_MAX ], *Wsig_ptr;
    SKP_int32 auto_corr[ FIND_PITCH_LPC_ORDER_MAX + 1 ];
    SKP_int16 rc_Q15[    FIND_PITCH_LPC_ORDER_MAX ];
    SKP_int32 A_Q24[     FIND_PITCH_LPC_ORDER_MAX ];
    SKP_int32 FiltState[ FIND_PITCH_LPC_ORDER_MAX ];
    SKP_int16 A_Q12[     FIND_PITCH_LPC_ORDER_MAX ];

    /******************************************/
    /* Setup buffer lengths etc based of Fs.  */
    /******************************************/
    buf_len = SKP_ADD_LSHIFT( psEnc->sCmn.la_pitch, psEnc->sCmn.frame_length, 1 );

    /* Safty check */
    SKP_assert( buf_len >= psPredSt->pitch_LPC_win_length );

    x_buf = x - psEnc->sCmn.frame_length;

    /*************************************/
    /* Estimate LPC AR coeficients */
    /*************************************/

    /* Calculate windowed signal */

    /* First LA_LTP samples */
    x_buf_ptr = x_buf + buf_len - psPredSt->pitch_LPC_win_length;
    Wsig_ptr  = Wsig;
    SKP_Silk_apply_sine_window( Wsig_ptr, x_buf_ptr, 1, psEnc->sCmn.la_pitch );

    /* Middle un - windowed samples */
    Wsig_ptr  += psEnc->sCmn.la_pitch;
    x_buf_ptr += psEnc->sCmn.la_pitch;
    SKP_memcpy( Wsig_ptr, x_buf_ptr, ( psPredSt->pitch_LPC_win_length - SKP_LSHIFT( psEnc->sCmn.la_pitch, 1 ) ) * sizeof( SKP_int16 ) );

    /* Last LA_LTP samples */
    Wsig_ptr  += psPredSt->pitch_LPC_win_length - SKP_LSHIFT( psEnc->sCmn.la_pitch, 1 );
    x_buf_ptr += psPredSt->pitch_LPC_win_length - SKP_LSHIFT( psEnc->sCmn.la_pitch, 1 );
    SKP_Silk_apply_sine_window( Wsig_ptr, x_buf_ptr, 2, psEnc->sCmn.la_pitch );

    /* Calculate autocorrelation sequence */
    SKP_Silk_autocorr( auto_corr, &scale, Wsig, psPredSt->pitch_LPC_win_length, psEnc->sCmn.pitchEstimationLPCOrder + 1 );

    /* add white noise, as fraction of energy */
    auto_corr[ 0 ] = SKP_SMLAWB( auto_corr[ 0 ], auto_corr[ 0 ], FIND_PITCH_WHITE_NOISE_FRACTION_Q16 );

    /* calculate the reflection coefficients using schur */
    SKP_Silk_schur( rc_Q15, auto_corr, psEnc->sCmn.pitchEstimationLPCOrder );

    /* convert reflection coefficients to prediction coefficients */
    SKP_Silk_k2a( A_Q24, rc_Q15, psEnc->sCmn.pitchEstimationLPCOrder );

    /* Convert From 32 bit Q24 to 16 bit Q12 coefs */
    for( i = 0; i < psEnc->sCmn.pitchEstimationLPCOrder; i++ ) {
        A_Q12[ i ] = ( SKP_int16 )SKP_SAT16( SKP_RSHIFT( A_Q24[ i ], 12 ) );
    }

    /* Do BWE */
    SKP_Silk_bwexpander( A_Q12, psEnc->sCmn.pitchEstimationLPCOrder, FIND_PITCH_BANDWITH_EXPANSION_Q16 );

    /*****************************************/
    /* LPC analysis filtering                */
    /*****************************************/
    SKP_memset( FiltState, 0, psEnc->sCmn.pitchEstimationLPCOrder * sizeof( SKP_int16 ) );
    SKP_Silk_MA_Prediction( x_buf, A_Q12, FiltState, res, buf_len, psEnc->sCmn.pitchEstimationLPCOrder );
    SKP_memset( res, 0, psEnc->sCmn.pitchEstimationLPCOrder * sizeof( SKP_int16 ) );

    /* Threshold for pitch estimator */
    thrhld_Q15 = ( 1 << 14 ); // 0.5f in Q15
    thrhld_Q15 = SKP_SMLABB( thrhld_Q15, -131, psEnc->sCmn.pitchEstimationLPCOrder );
    thrhld_Q15 = SKP_SMLABB( thrhld_Q15,  -13, ( SKP_int16 )SKP_Silk_SQRT_APPROX( SKP_LSHIFT( ( SKP_int32 )psEnc->speech_activity_Q8, 8 ) ) );
    thrhld_Q15 = SKP_SMLABB( thrhld_Q15, 4587, psEnc->sCmn.prev_sigtype );
    thrhld_Q15 = SKP_MLA(    thrhld_Q15,  -31, SKP_RSHIFT( psEncCtrl->input_tilt_Q15, 8 ) );
    thrhld_Q15 = SKP_SAT16(  thrhld_Q15 );

    /*****************************************/
    /* Call Pitch estimator */
    /*****************************************/
    psEncCtrl->sCmn.sigtype = SKP_Silk_pitch_analysis_core( res, psEncCtrl->sCmn.pitchL, &psEncCtrl->sCmn.lagIndex,
        &psEncCtrl->sCmn.contourIndex, &psEnc->LTPCorr_Q15, psEnc->sCmn.prevLag, psEnc->pitchEstimationThreshold_Q16,
        ( SKP_int16 )thrhld_Q15, psEnc->sCmn.fs_kHz, psEnc->sCmn.pitchEstimationComplexity );
}