/* Glues concealed frames with new good recieved frames */
void SKP_Silk_PLC_glue_frames(
SKP_Silk_decoder_state *psDec, /* I/O decoder state */
SKP_Silk_decoder_control *psDecCtrl, /* I/O Decoder control */
SKP_int16 signal[], /* I/O signal */
SKP_int length /* I length of residual */
)
{
SKP_int i, energy_shift;
SKP_int32 energy;
SKP_Silk_PLC_struct *psPLC;
psPLC = &psDec->sPLC;
if( psDec->lossCnt ) {
/* Calculate energy in concealed residual */
SKP_Silk_sum_sqr_shift( &psPLC->conc_energy, &psPLC->conc_energy_shift, signal, length );
psPLC->last_frame_lost = 1;
} else {
if( psDec->sPLC.last_frame_lost ) {
/* Calculate residual in decoded signal if last frame was lost */
SKP_Silk_sum_sqr_shift( &energy, &energy_shift, signal, length );
/* Normalize energies */
if( energy_shift > psPLC->conc_energy_shift ) {
psPLC->conc_energy = SKP_RSHIFT( psPLC->conc_energy, energy_shift - psPLC->conc_energy_shift );
} else if( energy_shift < psPLC->conc_energy_shift ) {
energy = SKP_RSHIFT( energy, psPLC->conc_energy_shift - energy_shift );
}
/* Fade in the energy difference */
if( energy > psPLC->conc_energy ) {
SKP_int32 frac_Q24, LZ;
SKP_int32 gain_Q12, slope_Q12;
LZ = SKP_Silk_CLZ32( psPLC->conc_energy );
LZ = LZ - 1;
psPLC->conc_energy = SKP_LSHIFT( psPLC->conc_energy, LZ );
energy = SKP_RSHIFT( energy, SKP_max_32( 24 - LZ, 0 ) );
frac_Q24 = SKP_DIV32( psPLC->conc_energy, SKP_max( energy, 1 ) );
gain_Q12 = SKP_Silk_SQRT_APPROX( frac_Q24 );
slope_Q12 = SKP_DIV32_16( ( 1 << 12 ) - gain_Q12, length );
for( i = 0; i < length; i++ ) {
signal[ i ] = SKP_RSHIFT( SKP_MUL( gain_Q12, signal[ i ] ), 12 );
gain_Q12 += slope_Q12;
gain_Q12 = SKP_min( gain_Q12, ( 1 << 12 ) );
}
}
}
psPLC->last_frame_lost = 0;
}
}
/* Laroia low complexity NLSF weights */
void SKP_Silk_NLSF_VQ_weights_laroia(
SKP_int *pNLSFW_Q6, /* O: Pointer to input vector weights [D x 1] */
const SKP_int *pNLSF_Q15, /* I: Pointer to input vector [D x 1] */
const SKP_int D /* I: Input vector dimension (even) */
)
{
SKP_int k;
SKP_int32 tmp1_int, tmp2_int;
/* Check that we are guaranteed to end up within the required range */
SKP_assert( D > 0 );
SKP_assert( ( D & 1 ) == 0 );
/* First value */
tmp1_int = SKP_max_int( pNLSF_Q15[ 0 ], MIN_NDELTA );
tmp1_int = SKP_DIV32_16( 1 << ( 15 + Q_OUT ), tmp1_int );
tmp2_int = SKP_max_int( pNLSF_Q15[ 1 ] - pNLSF_Q15[ 0 ], MIN_NDELTA );
tmp2_int = SKP_DIV32_16( 1 << ( 15 + Q_OUT ), tmp2_int );
pNLSFW_Q6[ 0 ] = (SKP_int)SKP_min_int( tmp1_int + tmp2_int, SKP_int16_MAX );
SKP_assert( pNLSFW_Q6[ 0 ] > 0 );
/* Main loop */
for( k = 1; k < D - 1; k += 2 ) {
tmp1_int = SKP_max_int( pNLSF_Q15[ k + 1 ] - pNLSF_Q15[ k ], MIN_NDELTA );
tmp1_int = SKP_DIV32_16( 1 << ( 15 + Q_OUT ), tmp1_int );
pNLSFW_Q6[ k ] = (SKP_int)SKP_min_int( tmp1_int + tmp2_int, SKP_int16_MAX );
SKP_assert( pNLSFW_Q6[ k ] > 0 );
tmp2_int = SKP_max_int( pNLSF_Q15[ k + 2 ] - pNLSF_Q15[ k + 1 ], MIN_NDELTA );
tmp2_int = SKP_DIV32_16( 1 << ( 15 + Q_OUT ), tmp2_int );
pNLSFW_Q6[ k + 1 ] = (SKP_int)SKP_min_int( tmp1_int + tmp2_int, SKP_int16_MAX );
SKP_assert( pNLSFW_Q6[ k + 1 ] > 0 );
}
/* Last value */
tmp1_int = SKP_max_int( ( 1 << 15 ) - pNLSF_Q15[ D - 1 ], MIN_NDELTA );
tmp1_int = SKP_DIV32_16( 1 << ( 15 + Q_OUT ), tmp1_int );
pNLSFW_Q6[ D - 1 ] = (SKP_int)SKP_min_int( tmp1_int + tmp2_int, SKP_int16_MAX );
SKP_assert( pNLSFW_Q6[ D - 1 ] > 0 );
}
/* uses SMLAWB(), requiring armv5E and higher. */
void SKP_Silk_schur(
SKP_int16 *rc_Q15, /* O: reflection coefficients [order] Q15 */
const SKP_int32 *c, /* I: correlations [order+1] */
const SKP_int32 order /* I: prediction order */
)
{
SKP_int k, n, lz;
SKP_int32 C[ SKP_Silk_MAX_ORDER_LPC + 1 ][ 2 ];
SKP_int32 Ctmp1, Ctmp2, rc_tmp_Q15;
/* Get number of leading zeros */
lz = SKP_Silk_CLZ32( c[ 0 ] );
/* Copy correlations and adjust level to Q30 */
if( lz < 2 ) {
/* lz must be 1, so shift one to the right */
for( k = 0; k < order + 1; k++ ) {
C[ k ][ 0 ] = C[ k ][ 1 ] = SKP_RSHIFT( c[ k ], 1 );
}
} else if( lz > 2 ) {
/* Shift to the left */
lz -= 2;
for( k = 0; k < order + 1; k++ ) {
C[ k ][ 0 ] = C[ k ][ 1 ] = SKP_LSHIFT( c[k], lz );
}
} else {
/* No need to shift */
for( k = 0; k < order + 1; k++ ) {
C[ k ][ 0 ] = C[ k ][ 1 ] = c[ k ];
}
}
for( k = 0; k < order; k++ ) {
/* Get reflection coefficient */
rc_tmp_Q15 = -SKP_DIV32_16( C[ k + 1 ][ 0 ], SKP_max_32( SKP_RSHIFT( C[ 0 ][ 1 ], 15 ), 1 ) );
/* Clip (shouldn't happen for properly conditioned inputs) */
rc_tmp_Q15 = SKP_SAT16( rc_tmp_Q15 );
/* Store */
rc_Q15[ k ] = (SKP_int16)rc_tmp_Q15;
/* Update correlations */
for( n = 0; n < order - k; n++ ) {
Ctmp1 = C[ n + k + 1 ][ 0 ];
Ctmp2 = C[ n ][ 1 ];
C[ n + k + 1 ][ 0 ] = SKP_SMLAWB( Ctmp1, SKP_LSHIFT( Ctmp2, 1 ), rc_tmp_Q15 );
C[ n ][ 1 ] = SKP_SMLAWB( Ctmp2, SKP_LSHIFT( Ctmp1, 1 ), rc_tmp_Q15 );
}
}
}
/* Laroia low complexity NLSF weights */
void SKP_Silk_NLSF_VQ_weights_laroia(
SKP_int16 *pNLSFW_Q5, /* O: Pointer to input vector weights [D x 1] */
const SKP_int16 *pNLSF_Q15, /* I: Pointer to input vector [D x 1] */
const SKP_int D /* I: Input vector dimension (even) */
)
{
SKP_int k;
SKP_int32 tmp1_int, tmp2_int;
SKP_assert( D > 0 );
SKP_assert( ( D & 1 ) == 0 );
/* First value */
tmp1_int = SKP_max_int( pNLSF_Q15[ 0 ], 1 );
tmp1_int = SKP_DIV32_16( 1 << ( 15 + Q_OUT ), tmp1_int );
tmp2_int = SKP_max_int( pNLSF_Q15[ 1 ] - pNLSF_Q15[ 0 ], 1 );
tmp2_int = SKP_DIV32_16( 1 << ( 15 + Q_OUT ), tmp2_int );
pNLSFW_Q5[ 0 ] = (SKP_int16)SKP_min_int( tmp1_int + tmp2_int, SKP_int16_MAX );
SKP_assert( pNLSFW_Q5[ 0 ] > 0 );
/* Main loop */
for( k = 1; k < D - 1; k += 2 ) {
tmp1_int = SKP_max_int( pNLSF_Q15[ k + 1 ] - pNLSF_Q15[ k ], 1 );
tmp1_int = SKP_DIV32_16( 1 << ( 15 + Q_OUT ), tmp1_int );
pNLSFW_Q5[ k ] = (SKP_int16)SKP_min_int( tmp1_int + tmp2_int, SKP_int16_MAX );
SKP_assert( pNLSFW_Q5[ k ] > 0 );
tmp2_int = SKP_max_int( pNLSF_Q15[ k + 2 ] - pNLSF_Q15[ k + 1 ], 1 );
tmp2_int = SKP_DIV32_16( 1 << ( 15 + Q_OUT ), tmp2_int );
pNLSFW_Q5[ k + 1 ] = (SKP_int16)SKP_min_int( tmp1_int + tmp2_int, SKP_int16_MAX );
SKP_assert( pNLSFW_Q5[ k + 1 ] > 0 );
}
/* Last value */
tmp1_int = SKP_max_int( ( 1 << 15 ) - pNLSF_Q15[ D - 1 ], 1 );
tmp1_int = SKP_DIV32_16( 1 << ( 15 + Q_OUT ), tmp1_int );
pNLSFW_Q5[ D - 1 ] = (SKP_int16)SKP_min_int( tmp1_int + tmp2_int, SKP_int16_MAX );
SKP_assert( pNLSFW_Q5[ D - 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 );
}
}
void SKP_Silk_CNG_Reset(SKP_Silk_decoder_state * psDec /* I/O Decoder state */
)
{
int i, NLSF_step_Q15, NLSF_acc_Q15;
NLSF_step_Q15 = SKP_DIV32_16(int16_t_MAX, psDec->LPC_order + 1);
NLSF_acc_Q15 = 0;
for (i = 0; i < psDec->LPC_order; i++) {
NLSF_acc_Q15 += NLSF_step_Q15;
psDec->sCNG.CNG_smth_NLSF_Q15[i] = NLSF_acc_Q15;
}
psDec->sCNG.CNG_smth_Gain_Q16 = 0;
psDec->sCNG.rand_seed = 3176576;
}
/* Processing of gains */
void silk_process_gains_FIX(
silk_encoder_state_FIX *psEnc, /* I/O Encoder state_FIX */
silk_encoder_control_FIX *psEncCtrl /* I/O Encoder control_FIX */
)
{
silk_shape_state_FIX *psShapeSt = &psEnc->sShape;
opus_int k;
opus_int32 s_Q16, InvMaxSqrVal_Q16, gain, gain_squared, ResNrg, ResNrgPart, quant_offset_Q10;
/* Gain reduction when LTP coding gain is high */
if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) {
/*s = -0.5f * SKP_sigmoid( 0.25f * ( psEncCtrl->LTPredCodGain - 12.0f ) ); */
s_Q16 = -silk_sigm_Q15( SKP_RSHIFT_ROUND( psEncCtrl->LTPredCodGain_Q7 - SILK_FIX_CONST( 12.0, 7 ), 4 ) );
for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) {
psEncCtrl->Gains_Q16[ k ] = SKP_SMLAWB( psEncCtrl->Gains_Q16[ k ], psEncCtrl->Gains_Q16[ k ], s_Q16 );
}
}
/* Limit the quantized signal */
/* InvMaxSqrVal = pow( 2.0f, 0.33f * ( 21.0f - SNR_dB ) ) / subfr_length; */
InvMaxSqrVal_Q16 = SKP_DIV32_16( silk_log2lin(
SKP_SMULWB( SILK_FIX_CONST( 21 + 16 / 0.33, 7 ) - psEnc->sCmn.SNR_dB_Q7, SILK_FIX_CONST( 0.33, 16 ) ) ), psEnc->sCmn.subfr_length );
for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) {
/* Soft limit on ratio residual energy and squared gains */
ResNrg = psEncCtrl->ResNrg[ k ];
ResNrgPart = SKP_SMULWW( ResNrg, InvMaxSqrVal_Q16 );
if( psEncCtrl->ResNrgQ[ k ] > 0 ) {
ResNrgPart = SKP_RSHIFT_ROUND( ResNrgPart, psEncCtrl->ResNrgQ[ k ] );
} else {
if( ResNrgPart >= SKP_RSHIFT( SKP_int32_MAX, -psEncCtrl->ResNrgQ[ k ] ) ) {
ResNrgPart = SKP_int32_MAX;
} else {
ResNrgPart = SKP_LSHIFT( ResNrgPart, -psEncCtrl->ResNrgQ[ k ] );
}
}
gain = psEncCtrl->Gains_Q16[ k ];
gain_squared = SKP_ADD_SAT32( ResNrgPart, SKP_SMMUL( gain, gain ) );
if( gain_squared < SKP_int16_MAX ) {
/* recalculate with higher precision */
gain_squared = SKP_SMLAWW( SKP_LSHIFT( ResNrgPart, 16 ), gain, gain );
SKP_assert( gain_squared > 0 );
gain = silk_SQRT_APPROX( gain_squared ); /* Q8 */
gain = SKP_min( gain, SKP_int32_MAX >> 8 );
psEncCtrl->Gains_Q16[ k ] = SKP_LSHIFT_SAT32( gain, 8 ); /* Q16 */
} else {
void SKP_Silk_LTP_scale_ctrl_FIX(
SKP_Silk_encoder_state_FIX *psEnc, /* I/O encoder state FIX */
SKP_Silk_encoder_control_FIX *psEncCtrl /* I/O encoder control FIX */
)
{
SKP_int round_loss, frames_per_packet;
SKP_int g_out_Q5, g_limit_Q15, thrld1_Q15, thrld2_Q15;
/* 1st order high-pass filter */
psEnc->HPLTPredCodGain_Q7 = SKP_max_int( psEncCtrl->LTPredCodGain_Q7 - psEnc->prevLTPredCodGain_Q7, 0 )
+ SKP_RSHIFT_ROUND( psEnc->HPLTPredCodGain_Q7, 1 );
psEnc->prevLTPredCodGain_Q7 = psEncCtrl->LTPredCodGain_Q7;
/* combine input and filtered input */
g_out_Q5 = SKP_RSHIFT_ROUND( SKP_RSHIFT( psEncCtrl->LTPredCodGain_Q7, 1 ) + SKP_RSHIFT( psEnc->HPLTPredCodGain_Q7, 1 ), 3 );
g_limit_Q15 = SKP_Silk_sigm_Q15( g_out_Q5 - ( 3 << 5 ) );
/* Default is minimum scaling */
psEncCtrl->sCmn.LTP_scaleIndex = 0;
/* Round the loss measure to whole pct */
round_loss = ( SKP_int )psEnc->sCmn.PacketLoss_perc;
/* Only scale if first frame in packet 0% */
if( psEnc->sCmn.nFramesInPayloadBuf == 0 ) {
frames_per_packet = SKP_DIV32_16( psEnc->sCmn.PacketSize_ms, FRAME_LENGTH_MS );
round_loss += frames_per_packet - 1;
thrld1_Q15 = LTPScaleThresholds_Q15[ SKP_min_int( round_loss, NB_THRESHOLDS - 1 ) ];
thrld2_Q15 = LTPScaleThresholds_Q15[ SKP_min_int( round_loss + 1, NB_THRESHOLDS - 1 ) ];
if( g_limit_Q15 > thrld1_Q15 ) {
/* Maximum scaling */
psEncCtrl->sCmn.LTP_scaleIndex = 2;
} else if( g_limit_Q15 > thrld2_Q15 ) {
/* Medium scaling */
psEncCtrl->sCmn.LTP_scaleIndex = 1;
}
}
psEncCtrl->LTP_scale_Q14 = SKP_Silk_LTPScales_table_Q14[ psEncCtrl->sCmn.LTP_scaleIndex ];
}
void silk_NLSF_decode(
opus_int16 *pNLSF_Q15, /* O Quantized NLSF vector [ LPC_ORDER ] */
opus_int8 *NLSFIndices, /* I Codebook path vector [ LPC_ORDER + 1 ] */
const silk_NLSF_CB_struct *psNLSF_CB /* I Codebook object */
)
{
opus_int i;
opus_uint8 pred_Q8[ MAX_LPC_ORDER ];
opus_int16 ec_ix[ MAX_LPC_ORDER ];
opus_int16 res_Q10[ MAX_LPC_ORDER ];
opus_int16 W_tmp_QW[ MAX_LPC_ORDER ];
opus_int32 W_tmp_Q9, NLSF_Q15_tmp;
const opus_uint8 *pCB_element;
/* Decode first stage */
pCB_element = &psNLSF_CB->CB1_NLSF_Q8[ NLSFIndices[ 0 ] * psNLSF_CB->order ];
for( i = 0; i < psNLSF_CB->order; i++ ) {
pNLSF_Q15[ i ] = SKP_LSHIFT( ( opus_int16 )pCB_element[ i ], 7 );
}
/* Unpack entropy table indices and predictor for current CB1 index */
silk_NLSF_unpack( ec_ix, pred_Q8, psNLSF_CB, NLSFIndices[ 0 ] );
/* Trellis dequantizer */
silk_NLSF_residual_dequant( res_Q10, &NLSFIndices[ 1 ], pred_Q8, psNLSF_CB->quantStepSize_Q16, psNLSF_CB->order );
/* Weights from codebook vector */
silk_NLSF_VQ_weights_laroia( W_tmp_QW, pNLSF_Q15, psNLSF_CB->order );
/* Apply inverse square-rooted weights and add to output */
for( i = 0; i < psNLSF_CB->order; i++ ) {
W_tmp_Q9 = silk_SQRT_APPROX( SKP_LSHIFT( ( opus_int32 )W_tmp_QW[ i ], 18 - NLSF_W_Q ) );
NLSF_Q15_tmp = SKP_ADD32( pNLSF_Q15[ i ], SKP_DIV32_16( SKP_LSHIFT( ( opus_int32 )res_Q10[ i ], 14 ), W_tmp_Q9 ) );
pNLSF_Q15[ i ] = (opus_int16)SKP_LIMIT( NLSF_Q15_tmp, 0, 32767 );
}
/* NLSF stabilization */
silk_NLSF_stabilize( pNLSF_Q15, psNLSF_CB->deltaMin_Q15, psNLSF_CB->order );
}
/* Deactivate by setting psEncC->mode = 0; */
void silk_LP_variable_cutoff(
silk_LP_state *psLP, /* I/O LP filter state */
SKP_int16 *signal, /* I/O Low-pass filtered output signal */
const SKP_int frame_length /* I Frame length */
)
{
SKP_int32 B_Q28[ TRANSITION_NB ], A_Q28[ TRANSITION_NA ], fac_Q16 = 0;
SKP_int ind = 0;
SKP_assert( psLP->transition_frame_no >= 0 && psLP->transition_frame_no <= TRANSITION_FRAMES );
/* Run filter if needed */
if( psLP->mode != 0 ) {
/* Calculate index and interpolation factor for interpolation */
#if( TRANSITION_INT_STEPS == 64 )
fac_Q16 = SKP_LSHIFT( TRANSITION_FRAMES - psLP->transition_frame_no, 16 - 6 );
#else
fac_Q16 = SKP_DIV32_16( SKP_LSHIFT( TRANSITION_FRAMES - psLP->transition_frame_no, 16 ), TRANSITION_FRAMES );
#endif
ind = SKP_RSHIFT( fac_Q16, 16 );
fac_Q16 -= SKP_LSHIFT( ind, 16 );
SKP_assert( ind >= 0 );
SKP_assert( ind < TRANSITION_INT_NUM );
/* Interpolate filter coefficients */
silk_LP_interpolate_filter_taps( B_Q28, A_Q28, ind, fac_Q16 );
/* Update transition frame number for next frame */
psLP->transition_frame_no = SKP_LIMIT( psLP->transition_frame_no + psLP->mode, 0, TRANSITION_FRAMES );
/* ARMA low-pass filtering */
SKP_assert( TRANSITION_NB == 3 && TRANSITION_NA == 2 );
silk_biquad_alt( signal, B_Q28, A_Q28, psLP->In_LP_State, signal, frame_length );
}
}
SKP_int SKP_Silk_VAD_GetSA_Q8( /* O Return value, 0 if success */
SKP_Silk_encoder_state *psEncC, /* I/O Encoder state */
const SKP_int16 pIn[] /* I PCM input */
)
{
SKP_int SA_Q15, pSNR_dB_Q7, input_tilt;
SKP_int decimated_framelength, dec_subframe_length, dec_subframe_offset, SNR_Q7, i, b, s;
SKP_int32 sumSquared, smooth_coef_Q16;
SKP_int16 HPstateTmp;
SKP_int16 X[ VAD_N_BANDS ][ MAX_FRAME_LENGTH / 2 ];
SKP_int32 Xnrg[ VAD_N_BANDS ];
SKP_int32 NrgToNoiseRatio_Q8[ VAD_N_BANDS ];
SKP_int32 speech_nrg, x_tmp;
SKP_int ret = 0;
SKP_Silk_VAD_state *psSilk_VAD = &psEncC->sVAD;
/* Safety checks */
SKP_assert( VAD_N_BANDS == 4 );
SKP_assert( MAX_FRAME_LENGTH >= psEncC->frame_length );
SKP_assert( psEncC->frame_length <= 512 );
SKP_assert( psEncC->frame_length == 8 * SKP_RSHIFT( psEncC->frame_length, 3 ) );
/***********************/
/* Filter and Decimate */
/***********************/
/* 0-8 kHz to 0-4 kHz and 4-8 kHz */
SKP_Silk_ana_filt_bank_1( pIn, &psSilk_VAD->AnaState[ 0 ], &X[ 0 ][ 0 ], &X[ 3 ][ 0 ], psEncC->frame_length );
/* 0-4 kHz to 0-2 kHz and 2-4 kHz */
SKP_Silk_ana_filt_bank_1( &X[ 0 ][ 0 ], &psSilk_VAD->AnaState1[ 0 ], &X[ 0 ][ 0 ], &X[ 2 ][ 0 ], SKP_RSHIFT( psEncC->frame_length, 1 ) );
/* 0-2 kHz to 0-1 kHz and 1-2 kHz */
SKP_Silk_ana_filt_bank_1( &X[ 0 ][ 0 ], &psSilk_VAD->AnaState2[ 0 ], &X[ 0 ][ 0 ], &X[ 1 ][ 0 ], SKP_RSHIFT( psEncC->frame_length, 2 ) );
/*********************************************/
/* HP filter on lowest band (differentiator) */
/*********************************************/
decimated_framelength = SKP_RSHIFT( psEncC->frame_length, 3 );
X[ 0 ][ decimated_framelength - 1 ] = SKP_RSHIFT( X[ 0 ][ decimated_framelength - 1 ], 1 );
HPstateTmp = X[ 0 ][ decimated_framelength - 1 ];
for( i = decimated_framelength - 1; i > 0; i-- ) {
X[ 0 ][ i - 1 ] = SKP_RSHIFT( X[ 0 ][ i - 1 ], 1 );
X[ 0 ][ i ] -= X[ 0 ][ i - 1 ];
}
X[ 0 ][ 0 ] -= psSilk_VAD->HPstate;
psSilk_VAD->HPstate = HPstateTmp;
/*************************************/
/* Calculate the energy in each band */
/*************************************/
for( b = 0; b < VAD_N_BANDS; b++ ) {
/* Find the decimated framelength in the non-uniformly divided bands */
decimated_framelength = SKP_RSHIFT( psEncC->frame_length, SKP_min_int( VAD_N_BANDS - b, VAD_N_BANDS - 1 ) );
/* Split length into subframe lengths */
dec_subframe_length = SKP_RSHIFT( decimated_framelength, VAD_INTERNAL_SUBFRAMES_LOG2 );
dec_subframe_offset = 0;
/* Compute energy per sub-frame */
/* initialize with summed energy of last subframe */
Xnrg[ b ] = psSilk_VAD->XnrgSubfr[ b ];
for( s = 0; s < VAD_INTERNAL_SUBFRAMES; s++ ) {
sumSquared = 0;
for( i = 0; i < dec_subframe_length; i++ ) {
/* The energy will be less than dec_subframe_length * ( SKP_int16_MIN / 8 ) ^ 2. */
/* Therefore we can accumulate with no risk of overflow (unless dec_subframe_length > 128) */
x_tmp = SKP_RSHIFT( X[ b ][ i + dec_subframe_offset ], 3 );
sumSquared = SKP_SMLABB( sumSquared, x_tmp, x_tmp );
/* Safety check */
SKP_assert( sumSquared >= 0 );
}
/* Add/saturate summed energy of current subframe */
if( s < VAD_INTERNAL_SUBFRAMES - 1 ) {
Xnrg[ b ] = SKP_ADD_POS_SAT32( Xnrg[ b ], sumSquared );
} else {
/* Look-ahead subframe */
Xnrg[ b ] = SKP_ADD_POS_SAT32( Xnrg[ b ], SKP_RSHIFT( sumSquared, 1 ) );
}
dec_subframe_offset += dec_subframe_length;
}
psSilk_VAD->XnrgSubfr[ b ] = sumSquared;
}
/********************/
/* Noise estimation */
/********************/
SKP_Silk_VAD_GetNoiseLevels( &Xnrg[ 0 ], psSilk_VAD );
/***********************************************/
/* Signal-plus-noise to noise ratio estimation */
/***********************************************/
sumSquared = 0;
input_tilt = 0;
for( b = 0; b < VAD_N_BANDS; b++ ) {
speech_nrg = Xnrg[ b ] - psSilk_VAD->NL[ b ];
if( speech_nrg > 0 ) {
/* Divide, with sufficient resolution */
if( ( Xnrg[ b ] & 0xFF800000 ) == 0 ) {
NrgToNoiseRatio_Q8[ b ] = SKP_DIV32( SKP_LSHIFT( Xnrg[ b ], 8 ), psSilk_VAD->NL[ b ] + 1 );
} else {
NrgToNoiseRatio_Q8[ b ] = SKP_DIV32( Xnrg[ b ], SKP_RSHIFT( psSilk_VAD->NL[ b ], 8 ) + 1 );
}
/* Convert to log domain */
SNR_Q7 = SKP_Silk_lin2log( NrgToNoiseRatio_Q8[ b ] ) - 8 * 128;
/* Sum-of-squares */
sumSquared = SKP_SMLABB( sumSquared, SNR_Q7, SNR_Q7 ); /* Q14 */
/* Tilt measure */
if( speech_nrg < ( 1 << 20 ) ) {
/* Scale down SNR value for small subband speech energies */
SNR_Q7 = SKP_SMULWB( SKP_LSHIFT( SKP_Silk_SQRT_APPROX( speech_nrg ), 6 ), SNR_Q7 );
}
input_tilt = SKP_SMLAWB( input_tilt, tiltWeights[ b ], SNR_Q7 );
} else {
NrgToNoiseRatio_Q8[ b ] = 256;
}
}
/* Mean-of-squares */
sumSquared = SKP_DIV32_16( sumSquared, VAD_N_BANDS ); /* Q14 */
/* Root-mean-square approximation, scale to dBs, and write to output pointer */
pSNR_dB_Q7 = ( SKP_int16 )( 3 * SKP_Silk_SQRT_APPROX( sumSquared ) ); /* Q7 */
/*********************************/
/* Speech Probability Estimation */
/*********************************/
SA_Q15 = SKP_Silk_sigm_Q15( SKP_SMULWB( VAD_SNR_FACTOR_Q16, pSNR_dB_Q7 ) - VAD_NEGATIVE_OFFSET_Q5 );
/**************************/
/* Frequency Tilt Measure */
/**************************/
psEncC->input_tilt_Q15 = SKP_LSHIFT( SKP_Silk_sigm_Q15( input_tilt ) - 16384, 1 );
/**************************************************/
/* Scale the sigmoid output based on power levels */
/**************************************************/
speech_nrg = 0;
for( b = 0; b < VAD_N_BANDS; b++ ) {
/* Accumulate signal-without-noise energies, higher frequency bands have more weight */
speech_nrg += ( b + 1 ) * SKP_RSHIFT( Xnrg[ b ] - psSilk_VAD->NL[ b ], 4 );
}
/* Power scaling */
if( speech_nrg <= 0 ) {
SA_Q15 = SKP_RSHIFT( SA_Q15, 1 );
} else if( speech_nrg < 32768 ) {
if( psEncC->frame_length == 10 * psEncC->fs_kHz ) {
speech_nrg = SKP_LSHIFT_SAT32( speech_nrg, 16 );
} else {
speech_nrg = SKP_LSHIFT_SAT32( speech_nrg, 15 );
}
/* square-root */
speech_nrg = SKP_Silk_SQRT_APPROX( speech_nrg );
SA_Q15 = SKP_SMULWB( 32768 + speech_nrg, SA_Q15 );
}
/* Copy the resulting speech activity in Q8 */
psEncC->speech_activity_Q8 = SKP_min_int( SKP_RSHIFT( SA_Q15, 7 ), SKP_uint8_MAX );
/***********************************/
/* Energy Level and SNR estimation */
/***********************************/
/* Smoothing coefficient */
smooth_coef_Q16 = SKP_SMULWB( VAD_SNR_SMOOTH_COEF_Q18, SKP_SMULWB( SA_Q15, SA_Q15 ) );
if( psEncC->frame_length == 10 * psEncC->fs_kHz ) {
smooth_coef_Q16 >>= 1;
}
opus_int silk_setup_resamplers(
silk_encoder_state_Fxx *psEnc, /* I/O */
opus_int fs_kHz /* I */
)
{
opus_int ret = SILK_NO_ERROR;
opus_int32 nSamples_temp;
if( psEnc->sCmn.fs_kHz != fs_kHz || psEnc->sCmn.prev_API_fs_Hz != psEnc->sCmn.API_fs_Hz )
{
if( psEnc->sCmn.fs_kHz == 0 ) {
/* Initialize the resampler for enc_API.c preparing resampling from API_fs_Hz to fs_kHz */
ret += silk_resampler_init( &psEnc->sCmn.resampler_state, psEnc->sCmn.API_fs_Hz, fs_kHz * 1000 );
} else {
/* Allocate worst case space for temporary upsampling, 8 to 48 kHz, so a factor 6 */
opus_int16 x_buf_API_fs_Hz[ ( 2 * MAX_FRAME_LENGTH_MS + LA_SHAPE_MS ) * MAX_API_FS_KHZ ];
#ifdef FIXED_POINT
opus_int16 *x_bufFIX = psEnc->x_buf;
#else
opus_int16 x_bufFIX[ 2 * MAX_FRAME_LENGTH + LA_SHAPE_MAX ];
#endif
nSamples_temp = SKP_LSHIFT( psEnc->sCmn.frame_length, 1 ) + LA_SHAPE_MS * psEnc->sCmn.fs_kHz;
#ifndef FIXED_POINT
SKP_float2short_array( x_bufFIX, psEnc->x_buf, nSamples_temp );
#endif
if( SKP_SMULBB( fs_kHz, 1000 ) < psEnc->sCmn.API_fs_Hz && psEnc->sCmn.fs_kHz != 0 ) {
/* Resample buffered data in x_buf to API_fs_Hz */
silk_resampler_state_struct temp_resampler_state;
/* Initialize resampler for temporary resampling of x_buf data to API_fs_Hz */
ret += silk_resampler_init( &temp_resampler_state, SKP_SMULBB( psEnc->sCmn.fs_kHz, 1000 ), psEnc->sCmn.API_fs_Hz );
/* Temporary resampling of x_buf data to API_fs_Hz */
ret += silk_resampler( &temp_resampler_state, x_buf_API_fs_Hz, x_bufFIX, nSamples_temp );
/* Calculate number of samples that has been temporarily upsampled */
nSamples_temp = SKP_DIV32_16( nSamples_temp * psEnc->sCmn.API_fs_Hz, SKP_SMULBB( psEnc->sCmn.fs_kHz, 1000 ) );
/* Initialize the resampler for enc_API.c preparing resampling from API_fs_Hz to fs_kHz */
ret += silk_resampler_init( &psEnc->sCmn.resampler_state, psEnc->sCmn.API_fs_Hz, SKP_SMULBB( fs_kHz, 1000 ) );
} else {
/* Copy data */
SKP_memcpy( x_buf_API_fs_Hz, x_bufFIX, nSamples_temp * sizeof( opus_int16 ) );
}
if( 1000 * fs_kHz != psEnc->sCmn.API_fs_Hz ) {
/* Correct resampler state (unless resampling by a factor 1) by resampling buffered data from API_fs_Hz to fs_kHz */
ret += silk_resampler( &psEnc->sCmn.resampler_state, x_bufFIX, x_buf_API_fs_Hz, nSamples_temp );
}
#ifndef FIXED_POINT
SKP_short2float_array( psEnc->x_buf, x_bufFIX, ( 2 * MAX_FRAME_LENGTH_MS + LA_SHAPE_MS ) * fs_kHz );
#endif
}
}
psEnc->sCmn.prev_API_fs_Hz = psEnc->sCmn.API_fs_Hz;
return ret;
}
void SKP_Silk_noise_shape_analysis_FIX(
SKP_Silk_encoder_state_FIX *psEnc, /* I/O Encoder state FIX */
SKP_Silk_encoder_control_FIX *psEncCtrl, /* I/O Encoder control FIX */
const SKP_int16 *pitch_res, /* I LPC residual from pitch analysis */
const SKP_int16 *x /* I Input signal [ frame_length + la_shape ] */
)
{
SKP_Silk_shape_state_FIX *psShapeSt = &psEnc->sShape;
SKP_int k, i, nSamples, Qnrg, b_Q14, warping_Q16, scale = 0;
SKP_int32 SNR_adj_dB_Q7, HarmBoost_Q16, HarmShapeGain_Q16, Tilt_Q16, tmp32;
SKP_int32 nrg, pre_nrg_Q30, log_energy_Q7, log_energy_prev_Q7, energy_variation_Q7;
SKP_int32 delta_Q16, BWExp1_Q16, BWExp2_Q16, gain_mult_Q16, gain_add_Q16, strength_Q16, b_Q8;
SKP_int32 auto_corr[ MAX_SHAPE_LPC_ORDER + 1 ];
SKP_int32 refl_coef_Q16[ MAX_SHAPE_LPC_ORDER ];
SKP_int32 AR1_Q24[ MAX_SHAPE_LPC_ORDER ];
SKP_int32 AR2_Q24[ MAX_SHAPE_LPC_ORDER ];
SKP_int16 x_windowed[ SHAPE_LPC_WIN_MAX ];
const SKP_int16 *x_ptr, *pitch_res_ptr;
SKP_int32 sqrt_nrg[ NB_SUBFR ], Qnrg_vec[ NB_SUBFR ];
/* Point to start of first LPC analysis block */
x_ptr = x - psEnc->sCmn.la_shape;
/****************/
/* CONTROL SNR */
/****************/
/* Reduce SNR_dB values if recent bitstream has exceeded TargetRate */
psEncCtrl->current_SNR_dB_Q7 = psEnc->SNR_dB_Q7 - SKP_SMULWB( SKP_LSHIFT( ( SKP_int32 )psEnc->BufferedInChannel_ms, 7 ),
SKP_FIX_CONST( 0.05, 16 ) );
/* Reduce SNR_dB if inband FEC used */
if( psEnc->speech_activity_Q8 > SKP_FIX_CONST( LBRR_SPEECH_ACTIVITY_THRES, 8 ) ) {
psEncCtrl->current_SNR_dB_Q7 -= SKP_RSHIFT( psEnc->inBandFEC_SNR_comp_Q8, 1 );
}
/****************/
/* GAIN CONTROL */
/****************/
/* Input quality is the average of the quality in the lowest two VAD bands */
psEncCtrl->input_quality_Q14 = ( SKP_int )SKP_RSHIFT( ( SKP_int32 )psEncCtrl->input_quality_bands_Q15[ 0 ]
+ psEncCtrl->input_quality_bands_Q15[ 1 ], 2 );
/* Coding quality level, between 0.0_Q0 and 1.0_Q0, but in Q14 */
psEncCtrl->coding_quality_Q14 = SKP_RSHIFT( SKP_Silk_sigm_Q15( SKP_RSHIFT_ROUND( psEncCtrl->current_SNR_dB_Q7 -
SKP_FIX_CONST( 18.0, 7 ), 4 ) ), 1 );
/* Reduce coding SNR during low speech activity */
b_Q8 = SKP_FIX_CONST( 1.0, 8 ) - psEnc->speech_activity_Q8;
b_Q8 = SKP_SMULWB( SKP_LSHIFT( b_Q8, 8 ), b_Q8 );
SNR_adj_dB_Q7 = SKP_SMLAWB( psEncCtrl->current_SNR_dB_Q7,
SKP_SMULBB( SKP_FIX_CONST( -BG_SNR_DECR_dB, 7 ) >> ( 4 + 1 ), b_Q8 ), // Q11
SKP_SMULWB( SKP_FIX_CONST( 1.0, 14 ) + psEncCtrl->input_quality_Q14, psEncCtrl->coding_quality_Q14 ) ); // Q12
if( psEncCtrl->sCmn.sigtype == SIG_TYPE_VOICED ) {
/* Reduce gains for periodic signals */
SNR_adj_dB_Q7 = SKP_SMLAWB( SNR_adj_dB_Q7, SKP_FIX_CONST( HARM_SNR_INCR_dB, 8 ), psEnc->LTPCorr_Q15 );
} else {
/* For unvoiced signals and low-quality input, adjust the quality slower than SNR_dB setting */
SNR_adj_dB_Q7 = SKP_SMLAWB( SNR_adj_dB_Q7,
SKP_SMLAWB( SKP_FIX_CONST( 6.0, 9 ), -SKP_FIX_CONST( 0.4, 18 ), psEncCtrl->current_SNR_dB_Q7 ),
SKP_FIX_CONST( 1.0, 14 ) - psEncCtrl->input_quality_Q14 );
}
/*************************/
/* SPARSENESS PROCESSING */
/*************************/
/* Set quantizer offset */
if( psEncCtrl->sCmn.sigtype == SIG_TYPE_VOICED ) {
/* Initally set to 0; may be overruled in process_gains(..) */
psEncCtrl->sCmn.QuantOffsetType = 0;
psEncCtrl->sparseness_Q8 = 0;
} else {
/* Sparseness measure, based on relative fluctuations of energy per 2 milliseconds */
nSamples = SKP_LSHIFT( psEnc->sCmn.fs_kHz, 1 );
energy_variation_Q7 = 0;
log_energy_prev_Q7 = 0;
pitch_res_ptr = pitch_res;
for( k = 0; k < FRAME_LENGTH_MS / 2; k++ ) {
SKP_Silk_sum_sqr_shift( &nrg, &scale, pitch_res_ptr, nSamples );
nrg += SKP_RSHIFT( nSamples, scale ); // Q(-scale)
log_energy_Q7 = SKP_Silk_lin2log( nrg );
if( k > 0 ) {
energy_variation_Q7 += SKP_abs( log_energy_Q7 - log_energy_prev_Q7 );
}
log_energy_prev_Q7 = log_energy_Q7;
pitch_res_ptr += nSamples;
}
psEncCtrl->sparseness_Q8 = SKP_RSHIFT( SKP_Silk_sigm_Q15( SKP_SMULWB( energy_variation_Q7 -
SKP_FIX_CONST( 5.0, 7 ), SKP_FIX_CONST( 0.1, 16 ) ) ), 7 );
/* Set quantization offset depending on sparseness measure */
if( psEncCtrl->sparseness_Q8 > SKP_FIX_CONST( SPARSENESS_THRESHOLD_QNT_OFFSET, 8 ) ) {
psEncCtrl->sCmn.QuantOffsetType = 0;
} else {
psEncCtrl->sCmn.QuantOffsetType = 1;
}
/* Increase coding SNR for sparse signals */
SNR_adj_dB_Q7 = SKP_SMLAWB( SNR_adj_dB_Q7, SKP_FIX_CONST( SPARSE_SNR_INCR_dB, 15 ), psEncCtrl->sparseness_Q8 - SKP_FIX_CONST( 0.5, 8 ) );
}
/*******************************/
/* Control bandwidth expansion */
/*******************************/
/* More BWE for signals with high prediction gain */
strength_Q16 = SKP_SMULWB( psEncCtrl->predGain_Q16, SKP_FIX_CONST( FIND_PITCH_WHITE_NOISE_FRACTION, 16 ) );
BWExp1_Q16 = BWExp2_Q16 = SKP_DIV32_varQ( SKP_FIX_CONST( BANDWIDTH_EXPANSION, 16 ),
SKP_SMLAWW( SKP_FIX_CONST( 1.0, 16 ), strength_Q16, strength_Q16 ), 16 );
delta_Q16 = SKP_SMULWB( SKP_FIX_CONST( 1.0, 16 ) - SKP_SMULBB( 3, psEncCtrl->coding_quality_Q14 ),
SKP_FIX_CONST( LOW_RATE_BANDWIDTH_EXPANSION_DELTA, 16 ) );
BWExp1_Q16 = SKP_SUB32( BWExp1_Q16, delta_Q16 );
BWExp2_Q16 = SKP_ADD32( BWExp2_Q16, delta_Q16 );
/* BWExp1 will be applied after BWExp2, so make it relative */
BWExp1_Q16 = SKP_DIV32_16( SKP_LSHIFT( BWExp1_Q16, 14 ), SKP_RSHIFT( BWExp2_Q16, 2 ) );
if( psEnc->sCmn.warping_Q16 > 0 ) {
/* Slightly more warping in analysis will move quantization noise up in frequency, where it's better masked */
warping_Q16 = SKP_SMLAWB( psEnc->sCmn.warping_Q16, psEncCtrl->coding_quality_Q14, SKP_FIX_CONST( 0.01, 18 ) );
} else {
warping_Q16 = 0;
}
/********************************************/
/* Compute noise shaping AR coefs and gains */
/********************************************/
for( k = 0; k < NB_SUBFR; k++ ) {
/* Apply window: sine slope followed by flat part followed by cosine slope */
SKP_int shift, slope_part, flat_part;
flat_part = psEnc->sCmn.fs_kHz * 5;
slope_part = SKP_RSHIFT( psEnc->sCmn.shapeWinLength - flat_part, 1 );
SKP_Silk_apply_sine_window_new( x_windowed, x_ptr, 1, slope_part );
shift = slope_part;
SKP_memcpy( x_windowed + shift, x_ptr + shift, flat_part * sizeof(SKP_int16) );
shift += flat_part;
SKP_Silk_apply_sine_window_new( 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 */
SKP_Silk_warped_autocorrelation_FIX( auto_corr, &scale, x_windowed, warping_Q16, psEnc->sCmn.shapeWinLength, psEnc->sCmn.shapingLPCOrder );
} else {
/* Calculate regular auto correlation */
SKP_Silk_autocorr( auto_corr, &scale, x_windowed, psEnc->sCmn.shapeWinLength, psEnc->sCmn.shapingLPCOrder + 1 );
}
/* Add white noise, as a fraction of energy */
auto_corr[0] = SKP_ADD32( auto_corr[0], SKP_max_32( SKP_SMULWB( SKP_RSHIFT( auto_corr[ 0 ], 4 ),
SKP_FIX_CONST( SHAPE_WHITE_NOISE_FRACTION, 20 ) ), 1 ) );
/* Calculate the reflection coefficients using schur */
nrg = SKP_Silk_schur64( refl_coef_Q16, auto_corr, psEnc->sCmn.shapingLPCOrder );
SKP_assert( nrg >= 0 );
/* Convert reflection coefficients to prediction coefficients */
SKP_Silk_k2a_Q16( AR2_Q24, refl_coef_Q16, psEnc->sCmn.shapingLPCOrder );
Qnrg = -scale; // range: -12...30
SKP_assert( Qnrg >= -12 );
SKP_assert( Qnrg <= 30 );
/* Make sure that Qnrg is an even number */
if( Qnrg & 1 ) {
Qnrg -= 1;
nrg >>= 1;
}
tmp32 = SKP_Silk_SQRT_APPROX( nrg );
Qnrg >>= 1; // range: -6...15
sqrt_nrg[ k ] = tmp32;
Qnrg_vec[ k ] = Qnrg;
psEncCtrl->Gains_Q16[ k ] = SKP_LSHIFT_SAT32( tmp32, 16 - Qnrg );
if( psEnc->sCmn.warping_Q16 > 0 ) {
/* Adjust gain for warping */
gain_mult_Q16 = warped_gain( AR2_Q24, warping_Q16, psEnc->sCmn.shapingLPCOrder );
SKP_assert( psEncCtrl->Gains_Q16[ k ] >= 0 );
psEncCtrl->Gains_Q16[ k ] = SKP_SMULWW( psEncCtrl->Gains_Q16[ k ], gain_mult_Q16 );
if( psEncCtrl->Gains_Q16[ k ] < 0 ) {
psEncCtrl->Gains_Q16[ k ] = SKP_int32_MAX;
}
}
/* Bandwidth expansion for synthesis filter shaping */
SKP_Silk_bwexpander_32( AR2_Q24, psEnc->sCmn.shapingLPCOrder, BWExp2_Q16 );
/* Compute noise shaping filter coefficients */
SKP_memcpy( AR1_Q24, AR2_Q24, psEnc->sCmn.shapingLPCOrder * sizeof( SKP_int32 ) );
/* Bandwidth expansion for analysis filter shaping */
SKP_assert( BWExp1_Q16 <= SKP_FIX_CONST( 1.0, 16 ) );
SKP_Silk_bwexpander_32( AR1_Q24, psEnc->sCmn.shapingLPCOrder, BWExp1_Q16 );
/* Ratio of prediction gains, in energy domain */
SKP_Silk_LPC_inverse_pred_gain_Q24( &pre_nrg_Q30, AR2_Q24, psEnc->sCmn.shapingLPCOrder );
SKP_Silk_LPC_inverse_pred_gain_Q24( &nrg, AR1_Q24, psEnc->sCmn.shapingLPCOrder );
//psEncCtrl->GainsPre[ k ] = 1.0f - 0.7f * ( 1.0f - pre_nrg / nrg ) = 0.3f + 0.7f * pre_nrg / nrg;
pre_nrg_Q30 = SKP_LSHIFT32( SKP_SMULWB( pre_nrg_Q30, SKP_FIX_CONST( 0.7, 15 ) ), 1 );
psEncCtrl->GainsPre_Q14[ k ] = ( SKP_int ) SKP_FIX_CONST( 0.3, 14 ) + SKP_DIV32_varQ( pre_nrg_Q30, nrg, 14 );
/* Convert to monic warped prediction coefficients and limit absolute values */
limit_warped_coefs( AR2_Q24, AR1_Q24, warping_Q16, SKP_FIX_CONST( 3.999, 24 ), psEnc->sCmn.shapingLPCOrder );
/* Convert from Q24 to Q13 and store in int16 */
for( i = 0; i < psEnc->sCmn.shapingLPCOrder; i++ ) {
psEncCtrl->AR1_Q13[ k * MAX_SHAPE_LPC_ORDER + i ] = (SKP_int16)SKP_SAT16( SKP_RSHIFT_ROUND( AR1_Q24[ i ], 11 ) );
psEncCtrl->AR2_Q13[ k * MAX_SHAPE_LPC_ORDER + i ] = (SKP_int16)SKP_SAT16( SKP_RSHIFT_ROUND( AR2_Q24[ i ], 11 ) );
}
}
/* Downsamples by a factor 3 */
void SKP_Silk_resample_1_3(
SKP_int16 *out, /* O: Fs_low signal [inLen/3] */
SKP_int32 *S, /* I/O: State vector [7] */
const SKP_int16 *in, /* I: Fs_high signal [inLen] */
const SKP_int32 inLen /* I: Input length, must be a multiple of 3 */
)
{
SKP_int k, outLen, LSubFrameIn, LSubFrameOut;
SKP_int32 out_tmp, limit = 102258000; // (102258000 + 1560) * 21 * 2^(-16) = 32767.5
SKP_int32 scratch0[ 3 * OUT_SUBFR_LEN ];
SKP_int32 scratch10[ OUT_SUBFR_LEN ], scratch11[ OUT_SUBFR_LEN ], scratch12[ OUT_SUBFR_LEN ];
/* coefficients for 3-fold resampling */
const SKP_int16 A30[ 2 ] = { 1773, 17818 };
const SKP_int16 A31[ 2 ] = { 4942, 25677 };
const SKP_int16 A32[ 2 ] = { 11786, 29304 };
/* Check that input is multiple of 3 */
SKP_assert( inLen % 3 == 0 );
outLen = SKP_DIV32_16( inLen, 3 );
while( outLen > 0 ) {
LSubFrameOut = SKP_min_int( OUT_SUBFR_LEN, outLen );
LSubFrameIn = SKP_SMULBB( 3, LSubFrameOut );
/* Low-pass filter, Q15 -> Q25 */
SKP_Silk_lowpass_short( in, S, scratch0, LSubFrameIn );
/* De-interleave three allpass inputs */
for( k = 0; k < LSubFrameOut; k++ ) {
scratch10[ k ] = scratch0[ 3 * k ];
scratch11[ k ] = scratch0[ 3 * k + 1 ];
scratch12[ k ] = scratch0[ 3 * k + 2 ];
}
/* Allpass filters */
SKP_Silk_allpass_int( scratch10, S + 1, A32[ 0 ], scratch0, LSubFrameOut );
SKP_Silk_allpass_int( scratch0, S + 2, A32[ 1 ], scratch10, LSubFrameOut );
SKP_Silk_allpass_int( scratch11, S + 3, A31[ 0 ], scratch0, LSubFrameOut );
SKP_Silk_allpass_int( scratch0, S + 4, A31[ 1 ], scratch11, LSubFrameOut );
SKP_Silk_allpass_int( scratch12, S + 5, A30[ 0 ], scratch0, LSubFrameOut );
SKP_Silk_allpass_int( scratch0, S + 6, A30[ 1 ], scratch12, LSubFrameOut );
/* Add three allpass outputs */
for( k = 0; k < LSubFrameOut; k++ ) {
out_tmp = scratch10[ k ] + scratch11[ k ] + scratch12[ k ];
if( out_tmp - limit > 0 ) {
out[ k ] = SKP_int16_MAX;
} else if( out_tmp + limit < 0 ) {
out[ k ] = SKP_int16_MIN;
} else {
out[ k ] = (SKP_int16) SKP_SMULWB( out_tmp + 1560, 21 );
}
}
in += LSubFrameIn;
out += LSubFrameOut;
outLen -= LSubFrameOut;
}
}
/* Control internal sampling rate */
SKP_int SKP_Silk_control_audio_bandwidth(
SKP_Silk_encoder_state *psEncC, /* I/O Pointer to Silk encoder state */
const SKP_int32 TargetRate_bps /* I Target max bitrate (bps) */
)
{
SKP_int fs_kHz;
fs_kHz = psEncC->fs_kHz;
if( fs_kHz == 0 ) {
/* Encoder has just been initialized */
if( TargetRate_bps >= SWB2WB_BITRATE_BPS ) {
fs_kHz = 24;
} else if( TargetRate_bps >= WB2MB_BITRATE_BPS ) {
fs_kHz = 16;
} else if( TargetRate_bps >= MB2NB_BITRATE_BPS ) {
fs_kHz = 12;
} else {
fs_kHz = 8;
}
/* Make sure internal rate is not higher than external rate or maximum allowed, or lower than minimum allowed */
fs_kHz = SKP_min( fs_kHz, SKP_DIV32_16( psEncC->API_fs_Hz, 1000 ) );
fs_kHz = SKP_min( fs_kHz, psEncC->maxInternal_fs_kHz );
} else if( SKP_SMULBB( fs_kHz, 1000 ) > psEncC->API_fs_Hz || fs_kHz > psEncC->maxInternal_fs_kHz ) {
/* Make sure internal rate is not higher than external rate or maximum allowed */
fs_kHz = SKP_DIV32_16( psEncC->API_fs_Hz, 1000 );
fs_kHz = SKP_min( fs_kHz, psEncC->maxInternal_fs_kHz );
} else {
/* State machine for the internal sampling rate switching */
if( psEncC->API_fs_Hz > 8000 ) {
/* Accumulate the difference between the target rate and limit for switching down */
psEncC->bitrateDiff += SKP_MUL( psEncC->PacketSize_ms, psEncC->TargetRate_bps - psEncC->bitrate_threshold_down );
psEncC->bitrateDiff = SKP_min( psEncC->bitrateDiff, 0 );
if( psEncC->vadFlag == NO_VOICE_ACTIVITY ) { /* Low speech activity */
/* Check if we should switch down */
#if SWITCH_TRANSITION_FILTERING
if( ( psEncC->sLP.transition_frame_no == 0 ) && /* Transition phase not active */
( psEncC->bitrateDiff <= -ACCUM_BITS_DIFF_THRESHOLD || /* Bitrate threshold is met */
( psEncC->sSWBdetect.WB_detected * psEncC->fs_kHz == 24 ) ) ) { /* Forced down-switching due to WB input */
psEncC->sLP.transition_frame_no = 1; /* Begin transition phase */
psEncC->sLP.mode = 0; /* Switch down */
} else if(
( psEncC->sLP.transition_frame_no >= TRANSITION_FRAMES_DOWN ) && /* Transition phase complete */
( psEncC->sLP.mode == 0 ) ) { /* Ready to switch down */
psEncC->sLP.transition_frame_no = 0; /* Ready for new transition phase */
#else
if( psEncC->bitrateDiff <= -ACCUM_BITS_DIFF_THRESHOLD ) { /* Bitrate threshold is met */
#endif
psEncC->bitrateDiff = 0;
/* Switch to a lower sample frequency */
if( psEncC->fs_kHz == 24 ) {
fs_kHz = 16;
} else if( psEncC->fs_kHz == 16 ) {
fs_kHz = 12;
} else {
SKP_assert( psEncC->fs_kHz == 12 );
fs_kHz = 8;
}
}
/* Check if we should switch up */
if( ( ( psEncC->fs_kHz * 1000 < psEncC->API_fs_Hz ) &&
( psEncC->TargetRate_bps >= psEncC->bitrate_threshold_up ) &&
( psEncC->sSWBdetect.WB_detected * psEncC->fs_kHz < 16 ) ) &&
( ( ( psEncC->fs_kHz == 16 ) && ( psEncC->maxInternal_fs_kHz >= 24 ) ) ||
( ( psEncC->fs_kHz == 12 ) && ( psEncC->maxInternal_fs_kHz >= 16 ) ) ||
( ( psEncC->fs_kHz == 8 ) && ( psEncC->maxInternal_fs_kHz >= 12 ) ) )
#if SWITCH_TRANSITION_FILTERING
&& ( psEncC->sLP.transition_frame_no == 0 ) ) { /* No transition phase running, ready to switch */
psEncC->sLP.mode = 1; /* Switch up */
#else
) {
#endif
psEncC->bitrateDiff = 0;
/* Switch to a higher sample frequency */
if( psEncC->fs_kHz == 8 ) {
fs_kHz = 12;
} else if( psEncC->fs_kHz == 12 ) {
fs_kHz = 16;
} else {
SKP_assert( psEncC->fs_kHz == 16 );
fs_kHz = 24;
}
}
}
}
/* Resamples input data with a factor 2/3 */
void SKP_Silk_resample_2_3_coarse(int16_t * out, /* O: Output signal */
int16_t * S, /* I/O: Resampler state [ SigProc_Resample_2_3_coarse_NUM_FIR_COEFS - 1 ] */
const int16_t * in, /* I: Input signal */
const int frameLenIn, /* I: Number of input samples */
int16_t * scratch /* I: Scratch memory [ frameLenIn + SigProc_Resample_2_3_coarse_NUM_FIR_COEFS - 1 ] */
)
{
int32_t n, ind, interpol_ind, tmp, index_Q16;
int16_t *in_ptr;
int frameLenOut;
const int16_t *interpol_ptr;
/* Copy buffered samples to start of scratch */
SKP_memcpy(scratch, S,
(SigProc_Resample_2_3_coarse_NUM_FIR_COEFS -
1) * sizeof(int16_t));
/* Then append by the input signal */
SKP_memcpy(&scratch[SigProc_Resample_2_3_coarse_NUM_FIR_COEFS - 1], in,
frameLenIn * sizeof(int16_t));
frameLenOut = SKP_DIV32_16(SKP_MUL(2, frameLenIn), 3);
index_Q16 = 0;
assert(frameLenIn == ((frameLenOut * 3) / 2));
/* Interpolate */
for (n = frameLenOut; n > 0; n--) {
/* Integer part */
ind = SKP_RSHIFT(index_Q16, 16);
/* Pointer to buffered input */
in_ptr = scratch + ind;
/* Fractional part */
interpol_ind =
(SKP_SMULWB
(index_Q16,
SigProc_Resample_2_3_coarse_NUM_INTERPOLATORS) &
(SigProc_Resample_2_3_coarse_NUM_INTERPOLATORS - 1));
/* Pointer to FIR taps */
interpol_ptr =
SigProc_Resample_2_3_coarse_INTERPOL[interpol_ind];
/* Interpolate */
/* Hardcoded for 32 FIR taps */
assert(SigProc_Resample_2_3_coarse_NUM_FIR_COEFS == 32);
tmp =
(int32_t) interpol_ptr[0] * in_ptr[0] +
(int32_t) interpol_ptr[1] * in_ptr[1] +
(int32_t) interpol_ptr[2] * in_ptr[2] +
(int32_t) interpol_ptr[3] * in_ptr[3] +
(int32_t) interpol_ptr[4] * in_ptr[4] +
(int32_t) interpol_ptr[5] * in_ptr[5] +
(int32_t) interpol_ptr[6] * in_ptr[6] +
(int32_t) interpol_ptr[7] * in_ptr[7] +
(int32_t) interpol_ptr[8] * in_ptr[8] +
(int32_t) interpol_ptr[9] * in_ptr[9] +
(int32_t) interpol_ptr[10] * in_ptr[10] +
(int32_t) interpol_ptr[11] * in_ptr[11] +
(int32_t) interpol_ptr[12] * in_ptr[12] +
(int32_t) interpol_ptr[13] * in_ptr[13] +
(int32_t) interpol_ptr[14] * in_ptr[14] +
(int32_t) interpol_ptr[15] * in_ptr[15] +
(int32_t) interpol_ptr[16] * in_ptr[16] +
(int32_t) interpol_ptr[17] * in_ptr[17] +
(int32_t) interpol_ptr[18] * in_ptr[18] +
(int32_t) interpol_ptr[19] * in_ptr[19] +
(int32_t) interpol_ptr[20] * in_ptr[20] +
(int32_t) interpol_ptr[21] * in_ptr[21] +
(int32_t) interpol_ptr[22] * in_ptr[22] +
(int32_t) interpol_ptr[23] * in_ptr[23] +
(int32_t) interpol_ptr[24] * in_ptr[24] +
(int32_t) interpol_ptr[25] * in_ptr[25] +
(int32_t) interpol_ptr[26] * in_ptr[26] +
(int32_t) interpol_ptr[27] * in_ptr[27] +
(int32_t) interpol_ptr[28] * in_ptr[28] +
(int32_t) interpol_ptr[29] * in_ptr[29];
/* Round, saturate and store to output array */
*out++ = (int16_t) SKP_SAT16(SKP_RSHIFT_ROUND(tmp, 15));
/* Update index */
index_Q16 += ((1 << 16) + (1 << 15)); // (3/2)_Q0;
}
/* Move last part of input signal to the sample buffer to prepare for the next call */
SKP_memcpy(S,
&in[frameLenIn -
(SigProc_Resample_2_3_coarse_NUM_FIR_COEFS - 1)],
(SigProc_Resample_2_3_coarse_NUM_FIR_COEFS -
1) * sizeof(int16_t));
}
opus_int silk_setup_fs(
silk_encoder_state_Fxx *psEnc, /* I/O */
opus_int fs_kHz, /* I */
opus_int PacketSize_ms /* I */
)
{
opus_int ret = SILK_NO_ERROR;
/* Set packet size */
if( PacketSize_ms != psEnc->sCmn.PacketSize_ms ) {
if( ( PacketSize_ms != 10 ) &&
( PacketSize_ms != 20 ) &&
( PacketSize_ms != 40 ) &&
( PacketSize_ms != 60 ) ) {
ret = SILK_ENC_PACKET_SIZE_NOT_SUPPORTED;
}
if( PacketSize_ms <= 10 ) {
psEnc->sCmn.nFramesPerPacket = 1;
psEnc->sCmn.nb_subfr = PacketSize_ms == 10 ? 2 : 1;
psEnc->sCmn.frame_length = SKP_SMULBB( PacketSize_ms, fs_kHz );
psEnc->sCmn.pitch_LPC_win_length = SKP_SMULBB( FIND_PITCH_LPC_WIN_MS_2_SF, fs_kHz );
if( psEnc->sCmn.fs_kHz == 8 ) {
psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_NB_iCDF;
} else {
psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_iCDF;
}
} else {
psEnc->sCmn.nFramesPerPacket = SKP_DIV32_16( PacketSize_ms, MAX_FRAME_LENGTH_MS );
psEnc->sCmn.nb_subfr = MAX_NB_SUBFR;
psEnc->sCmn.frame_length = SKP_SMULBB( 20, fs_kHz );
psEnc->sCmn.pitch_LPC_win_length = SKP_SMULBB( FIND_PITCH_LPC_WIN_MS, fs_kHz );
if( psEnc->sCmn.fs_kHz == 8 ) {
psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_NB_iCDF;
} else {
psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_iCDF;
}
}
psEnc->sCmn.PacketSize_ms = PacketSize_ms;
psEnc->sCmn.TargetRate_bps = 0; /* trigger new SNR computation */
}
/* Set internal sampling frequency */
SKP_assert( fs_kHz == 8 || fs_kHz == 12 || fs_kHz == 16 );
SKP_assert( psEnc->sCmn.nb_subfr == 2 || psEnc->sCmn.nb_subfr == 4 );
if( psEnc->sCmn.fs_kHz != fs_kHz ) {
/* reset part of the state */
#ifdef FIXED_POINT
SKP_memset( &psEnc->sShape, 0, sizeof( silk_shape_state_FIX ) );
SKP_memset( &psEnc->sPrefilt, 0, sizeof( silk_prefilter_state_FIX ) );
#else
SKP_memset( &psEnc->sShape, 0, sizeof( silk_shape_state_FLP ) );
SKP_memset( &psEnc->sPrefilt, 0, sizeof( silk_prefilter_state_FLP ) );
#endif
SKP_memset( &psEnc->sCmn.sNSQ, 0, sizeof( silk_nsq_state ) );
SKP_memset( psEnc->sCmn.prev_NLSFq_Q15, 0, sizeof( psEnc->sCmn.prev_NLSFq_Q15 ) );
SKP_memset( &psEnc->sCmn.sLP.In_LP_State, 0, sizeof( psEnc->sCmn.sLP.In_LP_State ) );
psEnc->sCmn.inputBufIx = 0;
psEnc->sCmn.nFramesEncoded = 0;
psEnc->sCmn.TargetRate_bps = 0; /* trigger new SNR computation */
/* Initialize non-zero parameters */
psEnc->sCmn.prevLag = 100;
psEnc->sCmn.first_frame_after_reset = 1;
psEnc->sPrefilt.lagPrev = 100;
psEnc->sShape.LastGainIndex = 10;
psEnc->sCmn.sNSQ.lagPrev = 100;
psEnc->sCmn.sNSQ.prev_inv_gain_Q16 = 65536;
psEnc->sCmn.fs_kHz = fs_kHz;
if( psEnc->sCmn.fs_kHz == 8 ) {
if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) {
psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_NB_iCDF;
} else {
psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_NB_iCDF;
}
} else {
if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) {
psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_iCDF;
} else {
psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_iCDF;
}
}
if( psEnc->sCmn.fs_kHz == 8 || psEnc->sCmn.fs_kHz == 12 ) {
psEnc->sCmn.predictLPCOrder = MIN_LPC_ORDER;
psEnc->sCmn.psNLSF_CB = &silk_NLSF_CB_NB_MB;
} else {
psEnc->sCmn.predictLPCOrder = MAX_LPC_ORDER;
psEnc->sCmn.psNLSF_CB = &silk_NLSF_CB_WB;
}
psEnc->sCmn.subfr_length = SUB_FRAME_LENGTH_MS * fs_kHz;
psEnc->sCmn.frame_length = SKP_SMULBB( psEnc->sCmn.subfr_length, psEnc->sCmn.nb_subfr );
psEnc->sCmn.ltp_mem_length = SKP_SMULBB( LTP_MEM_LENGTH_MS, fs_kHz );
psEnc->sCmn.la_pitch = SKP_SMULBB( LA_PITCH_MS, fs_kHz );
psEnc->sCmn.max_pitch_lag = SKP_SMULBB( 18, fs_kHz );
if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) {
psEnc->sCmn.pitch_LPC_win_length = SKP_SMULBB( FIND_PITCH_LPC_WIN_MS, fs_kHz );
} else {
psEnc->sCmn.pitch_LPC_win_length = SKP_SMULBB( FIND_PITCH_LPC_WIN_MS_2_SF, fs_kHz );
}
if( psEnc->sCmn.fs_kHz == 16 ) {
psEnc->sCmn.mu_LTP_Q9 = SILK_FIX_CONST( MU_LTP_QUANT_WB, 9 );
psEnc->sCmn.pitch_lag_low_bits_iCDF = silk_uniform8_iCDF;
} else if( psEnc->sCmn.fs_kHz == 12 ) {
psEnc->sCmn.mu_LTP_Q9 = SILK_FIX_CONST( MU_LTP_QUANT_MB, 9 );
psEnc->sCmn.pitch_lag_low_bits_iCDF = silk_uniform6_iCDF;
} else {
psEnc->sCmn.mu_LTP_Q9 = SILK_FIX_CONST( MU_LTP_QUANT_NB, 9 );
psEnc->sCmn.pitch_lag_low_bits_iCDF = silk_uniform4_iCDF;
}
}
/* Check that settings are valid */
SKP_assert( ( psEnc->sCmn.subfr_length * psEnc->sCmn.nb_subfr ) == psEnc->sCmn.frame_length );
return ret;
}
/* Processing of gains */
void SKP_Silk_process_gains_FIX(
SKP_Silk_encoder_state_FIX *psEnc, /* I/O Encoder state_FIX */
SKP_Silk_encoder_control_FIX *psEncCtrl /* I/O Encoder control_FIX */
)
{
SKP_Silk_shape_state_FIX *psShapeSt = &psEnc->sShape;
SKP_int k;
SKP_int32 s_Q16, InvMaxSqrVal_Q16, gain, gain_squared, ResNrg, ResNrgPart;
/* Gain reduction when LTP coding gain is high */
if( psEncCtrl->sCmn.sigtype == SIG_TYPE_VOICED ) {
/*s = -0.5f * SKP_sigmoid( 0.25f * ( psEncCtrl->LTPredCodGain - 12.0f ) ); */
s_Q16 = -SKP_Silk_sigm_Q15( SKP_RSHIFT_ROUND( psEncCtrl->LTPredCodGain_Q7 - (12 << 7), 4 ) );
for( k = 0; k < NB_SUBFR; k++ ) {
psEncCtrl->Gains_Q16[ k ] = SKP_SMLAWB( psEncCtrl->Gains_Q16[ k ], psEncCtrl->Gains_Q16[ k ], s_Q16 );
}
}
/* Limit the quantized signal */
/* 69 = 21.0f + 16/0.33 */
InvMaxSqrVal_Q16 = SKP_DIV32_16( SKP_Silk_log2lin(
SKP_SMULWB( (69 << 7) - psEncCtrl->current_SNR_dB_Q7, SKP_FIX_CONST( 0.33, 16 )) ), psEnc->sCmn.subfr_length );
for( k = 0; k < NB_SUBFR; k++ ) {
/* Soft limit on ratio residual energy and squared gains */
ResNrg = psEncCtrl->ResNrg[ k ];
ResNrgPart = SKP_SMULWW( ResNrg, InvMaxSqrVal_Q16 );
if( psEncCtrl->ResNrgQ[ k ] > 0 ) {
if( psEncCtrl->ResNrgQ[ k ] < 32 ) {
ResNrgPart = SKP_RSHIFT_ROUND( ResNrgPart, psEncCtrl->ResNrgQ[ k ] );
} else {
ResNrgPart = 0;
}
} else if( psEncCtrl->ResNrgQ[k] != 0 ) {
if( ResNrgPart > SKP_RSHIFT( SKP_int32_MAX, -psEncCtrl->ResNrgQ[ k ] ) ) {
ResNrgPart = SKP_int32_MAX;
} else {
ResNrgPart = SKP_LSHIFT( ResNrgPart, -psEncCtrl->ResNrgQ[ k ] );
}
}
gain = psEncCtrl->Gains_Q16[ k ];
gain_squared = SKP_ADD_SAT32( ResNrgPart, SKP_SMMUL( gain, gain ) );
if( gain_squared < SKP_int16_MAX ) {
/* recalculate with higher precision */
gain_squared = SKP_SMLAWW( SKP_LSHIFT( ResNrgPart, 16 ), gain, gain );
SKP_assert( gain_squared > 0 );
gain = SKP_Silk_SQRT_APPROX( gain_squared ); /* Q8 */
psEncCtrl->Gains_Q16[ k ] = SKP_LSHIFT_SAT32( gain, 8 ); /* Q16 */
} else {
gain = SKP_Silk_SQRT_APPROX( gain_squared ); /* Q0 */
psEncCtrl->Gains_Q16[ k ] = SKP_LSHIFT_SAT32( gain, 16 ); /* Q16 */
}
}
/* Noise shaping quantization */
SKP_Silk_gains_quant( psEncCtrl->sCmn.GainsIndices, psEncCtrl->Gains_Q16,
&psShapeSt->LastGainIndex, psEnc->sCmn.nFramesInPayloadBuf );
/* Set quantizer offset for voiced signals. Larger offset when LTP coding gain is low or tilt is high (ie low-pass) */
if( psEncCtrl->sCmn.sigtype == SIG_TYPE_VOICED ) {
if( psEncCtrl->LTPredCodGain_Q7 + SKP_RSHIFT( psEncCtrl->input_tilt_Q15, 8 ) > ( 1 << 7 ) ) {
psEncCtrl->sCmn.QuantOffsetType = 0;
} else {
psEncCtrl->sCmn.QuantOffsetType = 1;
}
}
/* Quantizer boundary adjustment */
if( psEncCtrl->sCmn.sigtype == SIG_TYPE_VOICED ) {
psEncCtrl->Lambda_Q10 = SKP_FIX_CONST( 1.3, 10 )
- SKP_SMULWB( SKP_FIX_CONST( 0.5, 18 ), psEnc->speech_activity_Q8 )
- SKP_SMULWB( SKP_FIX_CONST( 0.3, 12 ), psEncCtrl->input_quality_Q14 )
+ SKP_SMULBB( SKP_FIX_CONST( 0.2, 10 ), psEncCtrl->sCmn.QuantOffsetType )
- SKP_SMULWB( SKP_FIX_CONST( 0.1, 12 ), psEncCtrl->coding_quality_Q14 );
} else {
psEncCtrl->Lambda_Q10 = SKP_FIX_CONST( 1.3, 10 )
- SKP_SMULWB( SKP_FIX_CONST( 0.5, 18 ), psEnc->speech_activity_Q8 )
- SKP_SMULWB( SKP_FIX_CONST( 0.4, 12 ), psEncCtrl->input_quality_Q14 )
+ SKP_SMULBB( SKP_FIX_CONST( 0.4, 10 ), psEncCtrl->sCmn.QuantOffsetType )
- SKP_SMULWB( SKP_FIX_CONST( 0.1, 12 ), psEncCtrl->coding_quality_Q14 );
}
SKP_assert( psEncCtrl->Lambda_Q10 >= 0 );
SKP_assert( psEncCtrl->Lambda_Q10 < SKP_FIX_CONST( 2, 10 ) );
}
void SKP_Silk_noise_shape_analysis_FIX(
SKP_Silk_encoder_state_FIX *psEnc, /* I/O Encoder state FIX */
SKP_Silk_encoder_control_FIX *psEncCtrl, /* I/O Encoder control FIX */
const SKP_int16 *pitch_res, /* I LPC residual from pitch analysis */
const SKP_int16 *x /* I Input signal [ 2 * frame_length + la_shape ]*/
)
{
SKP_Silk_shape_state_FIX *psShapeSt = &psEnc->sShape;
SKP_int k, nSamples, lz, Qnrg, b_Q14, scale = 0, sz;
SKP_int32 SNR_adj_dB_Q7, HarmBoost_Q16, HarmShapeGain_Q16, Tilt_Q16, tmp32;
SKP_int32 nrg, pre_nrg_Q30, log_energy_Q7, log_energy_prev_Q7, energy_variation_Q7;
SKP_int32 delta_Q16, BWExp1_Q16, BWExp2_Q16, gain_mult_Q16, gain_add_Q16, strength_Q16, b_Q8;
SKP_int32 auto_corr[ SHAPE_LPC_ORDER_MAX + 1 ];
SKP_int32 refl_coef_Q16[ SHAPE_LPC_ORDER_MAX ];
SKP_int32 AR_Q24[ SHAPE_LPC_ORDER_MAX ];
SKP_int16 x_windowed[ SHAPE_LPC_WIN_MAX ];
const SKP_int16 *x_ptr, *pitch_res_ptr;
SKP_int32 sqrt_nrg[ NB_SUBFR ], Qnrg_vec[ NB_SUBFR ];
/* Point to start of first LPC analysis block */
x_ptr = x + psEnc->sCmn.la_shape - SKP_SMULBB( SHAPE_LPC_WIN_MS, psEnc->sCmn.fs_kHz ) + psEnc->sCmn.frame_length / NB_SUBFR;
/****************/
/* CONTROL SNR */
/****************/
/* Reduce SNR_dB values if recent bitstream has exceeded TargetRate */
psEncCtrl->current_SNR_dB_Q7 = psEnc->SNR_dB_Q7 - SKP_SMULWB( SKP_LSHIFT( ( SKP_int32 )psEnc->BufferedInChannel_ms, 7 ), 3277 );
/* Reduce SNR_dB if inband FEC used */
if( psEnc->speech_activity_Q8 > LBRR_SPEECH_ACTIVITY_THRES_Q8 ) {
psEncCtrl->current_SNR_dB_Q7 -= SKP_RSHIFT( psEnc->inBandFEC_SNR_comp_Q8, 1 );
}
/****************/
/* GAIN CONTROL */
/****************/
/* Input quality is the average of the quality in the lowest two VAD bands */
psEncCtrl->input_quality_Q14 = ( SKP_int )SKP_RSHIFT( ( SKP_int32 )psEncCtrl->input_quality_bands_Q15[ 0 ]
+ psEncCtrl->input_quality_bands_Q15[ 1 ], 2 );
/* Coding quality level, between 0.0_Q0 and 1.0_Q0, but in Q14 */
psEncCtrl->coding_quality_Q14 = SKP_RSHIFT( SKP_Silk_sigm_Q15( SKP_RSHIFT_ROUND( psEncCtrl->current_SNR_dB_Q7 - ( 18 << 7 ), 4 ) ), 1 );
/* Reduce coding SNR during low speech activity */
b_Q8 = ( 1 << 8 ) - psEnc->speech_activity_Q8;
b_Q8 = SKP_SMULWB( SKP_LSHIFT( b_Q8, 8 ), b_Q8 );
SNR_adj_dB_Q7 = SKP_SMLAWB( psEncCtrl->current_SNR_dB_Q7,
SKP_SMULBB( -BG_SNR_DECR_dB_Q7 >> ( 4 + 1 ), b_Q8 ), // Q11
SKP_SMULWB( ( 1 << 14 ) + psEncCtrl->input_quality_Q14, psEncCtrl->coding_quality_Q14 ) ); // Q12
if( psEncCtrl->sCmn.sigtype == SIG_TYPE_VOICED ) {
/* Reduce gains for periodic signals */
SNR_adj_dB_Q7 = SKP_SMLAWB( SNR_adj_dB_Q7, HARM_SNR_INCR_dB_Q7 << 1, psEnc->LTPCorr_Q15 );
} else {
/* For unvoiced signals and low-quality input, adjust the quality slower than SNR_dB setting */
SNR_adj_dB_Q7 = SKP_SMLAWB( SNR_adj_dB_Q7,
SKP_SMLAWB( 6 << ( 7 + 2 ), -104856, psEncCtrl->current_SNR_dB_Q7 ), //-104856_Q18 = -0.4_Q0, Q9
( 1 << 14 ) - psEncCtrl->input_quality_Q14 ); // Q14
}
/*************************/
/* SPARSENESS PROCESSING */
/*************************/
/* Set quantizer offset */
if( psEncCtrl->sCmn.sigtype == SIG_TYPE_VOICED ) {
/* Initally set to 0; may be overruled in process_gains(..) */
psEncCtrl->sCmn.QuantOffsetType = 0;
psEncCtrl->sparseness_Q8 = 0;
} else {
/* Sparseness measure, based on relative fluctuations of energy per 2 milliseconds */
nSamples = SKP_LSHIFT( psEnc->sCmn.fs_kHz, 1 );
energy_variation_Q7 = 0;
log_energy_prev_Q7 = 0;
pitch_res_ptr = pitch_res;
for( k = 0; k < FRAME_LENGTH_MS / 2; k++ ) {
SKP_Silk_sum_sqr_shift( &nrg, &scale, pitch_res_ptr, nSamples );
nrg += SKP_RSHIFT( nSamples, scale ); // Q(-scale)
log_energy_Q7 = SKP_Silk_lin2log( nrg );
if( k > 0 ) {
energy_variation_Q7 += SKP_abs( log_energy_Q7 - log_energy_prev_Q7 );
}
log_energy_prev_Q7 = log_energy_Q7;
pitch_res_ptr += nSamples;
}
psEncCtrl->sparseness_Q8 = SKP_RSHIFT( SKP_Silk_sigm_Q15( SKP_SMULWB( energy_variation_Q7 - ( 5 << 7 ), 6554 ) ), 7 ); // 6554_Q16 = 0.1_Q0
/* Set quantization offset depending on sparseness measure */
if( psEncCtrl->sparseness_Q8 > SPARSENESS_THRESHOLD_QNT_OFFSET_Q8 ) {
psEncCtrl->sCmn.QuantOffsetType = 0;
} else {
psEncCtrl->sCmn.QuantOffsetType = 1;
}
/* Increase coding SNR for sparse signals */
SNR_adj_dB_Q7 = SKP_SMLAWB( SNR_adj_dB_Q7, SPARSE_SNR_INCR_dB_Q7 << 8, psEncCtrl->sparseness_Q8 - ( 1 << 7 ) );
}
/*******************************/
/* Control bandwidth expansion */
/*******************************/
delta_Q16 = SKP_SMULWB( ( 1 << 16 ) - SKP_SMULBB( 3, psEncCtrl->coding_quality_Q14 ), LOW_RATE_BANDWIDTH_EXPANSION_DELTA_Q16 );
BWExp1_Q16 = BANDWIDTH_EXPANSION_Q16 - delta_Q16;
BWExp2_Q16 = BANDWIDTH_EXPANSION_Q16 + delta_Q16;
if( psEnc->sCmn.fs_kHz == 24 ) {
/* Less bandwidth expansion for super wideband */
BWExp1_Q16 = ( 1 << 16 ) - SKP_SMULWB( SWB_BANDWIDTH_EXPANSION_REDUCTION_Q16, ( 1 << 16 ) - BWExp1_Q16 );
BWExp2_Q16 = ( 1 << 16 ) - SKP_SMULWB( SWB_BANDWIDTH_EXPANSION_REDUCTION_Q16, ( 1 << 16 ) - BWExp2_Q16 );
}
/* BWExp1 will be applied after BWExp2, so make it relative */
BWExp1_Q16 = SKP_DIV32_16( SKP_LSHIFT( BWExp1_Q16, 14 ), SKP_RSHIFT( BWExp2_Q16, 2 ) );
/********************************************/
/* Compute noise shaping AR coefs and gains */
/********************************************/
sz = ( SKP_int )SKP_SMULBB( SHAPE_LPC_WIN_MS, psEnc->sCmn.fs_kHz );
for( k = 0; k < NB_SUBFR; k++ ) {
/* Apply window */
SKP_Silk_apply_sine_window( x_windowed, x_ptr, 0, SHAPE_LPC_WIN_MS * psEnc->sCmn.fs_kHz );
/* Update pointer: next LPC analysis block */
x_ptr += psEnc->sCmn.frame_length / NB_SUBFR;
/* Calculate auto correlation */
SKP_Silk_autocorr( auto_corr, &scale, x_windowed, sz, psEnc->sCmn.shapingLPCOrder + 1 );
/* Add white noise, as a fraction of energy */
auto_corr[0] = SKP_ADD32( auto_corr[0], SKP_max_32( SKP_SMULWB( SKP_RSHIFT( auto_corr[ 0 ], 4 ), SHAPE_WHITE_NOISE_FRACTION_Q20 ), 1 ) );
/* Calculate the reflection coefficients using schur */
nrg = SKP_Silk_schur64( refl_coef_Q16, auto_corr, psEnc->sCmn.shapingLPCOrder );
/* Convert reflection coefficients to prediction coefficients */
SKP_Silk_k2a_Q16( AR_Q24, refl_coef_Q16, psEnc->sCmn.shapingLPCOrder );
/* Bandwidth expansion for synthesis filter shaping */
SKP_Silk_bwexpander_32( AR_Q24, psEnc->sCmn.shapingLPCOrder, BWExp2_Q16 );
/* Make sure to fit in Q13 SKP_int16 */
SKP_Silk_LPC_fit( &psEncCtrl->AR2_Q13[ k * SHAPE_LPC_ORDER_MAX ], AR_Q24, 13, psEnc->sCmn.shapingLPCOrder );
/* Compute noise shaping filter coefficients */
SKP_memcpy(
&psEncCtrl->AR1_Q13[ k * SHAPE_LPC_ORDER_MAX ],
&psEncCtrl->AR2_Q13[ k * SHAPE_LPC_ORDER_MAX ],
psEnc->sCmn.shapingLPCOrder * sizeof( SKP_int16 ) );
/* Bandwidth expansion for analysis filter shaping */
SKP_assert( BWExp1_Q16 <= ( 1 << 16 ) ); // If ever breaking, use LPC_stabilize() in these cases to stay within range
SKP_Silk_bwexpander( &psEncCtrl->AR1_Q13[ k * SHAPE_LPC_ORDER_MAX ], psEnc->sCmn.shapingLPCOrder, BWExp1_Q16 );
/* Increase residual energy */
nrg = SKP_SMLAWB( nrg, SKP_RSHIFT( auto_corr[ 0 ], 8 ), SHAPE_MIN_ENERGY_RATIO_Q24 );
Qnrg = -scale; // range: -12...30
SKP_assert( Qnrg >= -12 );
SKP_assert( Qnrg <= 30 );
/* Make sure that Qnrg is an even number */
if( Qnrg & 1 ) {
Qnrg -= 1;
nrg >>= 1;
}
tmp32 = SKP_Silk_SQRT_APPROX( nrg );
Qnrg >>= 1; // range: -6...15
sqrt_nrg[ k ] = tmp32;
Qnrg_vec[ k ] = Qnrg;
psEncCtrl->Gains_Q16[ k ] = SKP_LSHIFT_SAT32( tmp32, 16 - Qnrg );
/* Ratio of prediction gains, in energy domain */
SKP_Silk_LPC_inverse_pred_gain_Q13( &pre_nrg_Q30, &psEncCtrl->AR2_Q13[ k * SHAPE_LPC_ORDER_MAX ], psEnc->sCmn.shapingLPCOrder );
SKP_Silk_LPC_inverse_pred_gain_Q13( &nrg, &psEncCtrl->AR1_Q13[ k * SHAPE_LPC_ORDER_MAX ], psEnc->sCmn.shapingLPCOrder );
lz = SKP_min_32( SKP_Silk_CLZ32( pre_nrg_Q30 ) - 1, 19 );
pre_nrg_Q30 = SKP_DIV32( SKP_LSHIFT( pre_nrg_Q30, lz ), SKP_RSHIFT( nrg, 20 - lz ) + 1 ); // Q20
pre_nrg_Q30 = SKP_RSHIFT( SKP_LSHIFT_SAT32( pre_nrg_Q30, 9 ), 1 ); /* Q28 */
psEncCtrl->GainsPre_Q14[ k ] = ( SKP_int )SKP_Silk_SQRT_APPROX( pre_nrg_Q30 );
}
/* Control encoder SNR */
SKP_int SKP_Silk_control_encoder_FIX(
SKP_Silk_encoder_state_FIX *psEnc, /* I/O Pointer to Silk encoder state */
const SKP_int32 API_fs_Hz, /* I External (API) sampling rate (Hz) */
const SKP_int max_internal_fs_kHz,/* I Maximum internal sampling rate (kHz) */
const SKP_int PacketSize_ms, /* I Packet length (ms) */
SKP_int32 TargetRate_bps, /* I Target max bitrate (bps) (used if SNR_dB == 0) */
const SKP_int PacketLoss_perc, /* I Packet loss rate (in percent) */
const SKP_int INBandFEC_enabled, /* I Enable (1) / disable (0) inband FEC */
const SKP_int DTX_enabled, /* I Enable / disable DTX */
const SKP_int InputFramesize_ms, /* I Inputframe in ms */
const SKP_int Complexity /* I Complexity (0->low; 1->medium; 2->high) */
)
{
SKP_int32 LBRRRate_thres_bps;
SKP_int k, fs_kHz, ret = 0;
SKP_int32 frac_Q6;
const SKP_int32 *rateTable;
/* State machine for the SWB/WB switching */
fs_kHz = psEnc->sCmn.fs_kHz;
/* Only switch during low speech activity, when no frames are sitting in the payload buffer */
if( API_fs_Hz == 8000 || fs_kHz == 0 || API_fs_Hz < SKP_SMULBB( fs_kHz, 1000 ) || fs_kHz > max_internal_fs_kHz ) {
/* Switching is not possible, encoder just initialized, internal mode higher than external, */
/* or internal mode higher than maximum allowed internal mode */
fs_kHz = SKP_min( SKP_DIV32_16( API_fs_Hz, 1000 ), max_internal_fs_kHz );
} else {
/* Accumulate the difference between the target rate and limit for switching down */
psEnc->sCmn.bitrateDiff += SKP_MUL( InputFramesize_ms, TargetRate_bps - psEnc->sCmn.bitrate_threshold_down );
psEnc->sCmn.bitrateDiff = SKP_min( psEnc->sCmn.bitrateDiff, 0 );
if( psEnc->speech_activity_Q8 < 128 && psEnc->sCmn.nFramesInPayloadBuf == 0 ) { /* Low speech activity and payload buffer empty */
/* Check if we should switch down */
#if SWITCH_TRANSITION_FILTERING
if( ( psEnc->sCmn.sLP.transition_frame_no == 0 ) && /* Transition phase not active */
( psEnc->sCmn.bitrateDiff <= -ACCUM_BITS_DIFF_THRESHOLD || /* Bitrate threshold is met */
( psEnc->sCmn.sSWBdetect.WB_detected * psEnc->sCmn.fs_kHz == 24 ) ) ) { /* Forced down-switching due to WB input */
psEnc->sCmn.sLP.transition_frame_no = 1; /* Begin transition phase */
psEnc->sCmn.sLP.mode = 0; /* Switch down */
} else if(
( psEnc->sCmn.sLP.transition_frame_no >= TRANSITION_FRAMES_DOWN ) && /* Transition phase complete */
( psEnc->sCmn.sLP.mode == 0 ) ) { /* Ready to switch down */
psEnc->sCmn.sLP.transition_frame_no = 0; /* Ready for new transition phase */
#else
if( psEnc->sCmn.bitrateDiff <= -ACCUM_BITS_DIFF_THRESHOLD ) { /* Bitrate threshold is met */
#endif
psEnc->sCmn.bitrateDiff = 0;
/* Switch to a lower sample frequency */
if( psEnc->sCmn.fs_kHz == 24 ) {
fs_kHz = 16;
} else if( psEnc->sCmn.fs_kHz == 16 ) {
fs_kHz = 12;
} else {
SKP_assert( psEnc->sCmn.fs_kHz == 12 );
fs_kHz = 8;
}
}
/* Check if we should switch up */
if( ( ( SKP_SMULBB( psEnc->sCmn.fs_kHz, 1000 ) < API_fs_Hz ) &&
( TargetRate_bps >= psEnc->sCmn.bitrate_threshold_up ) &&
( psEnc->sCmn.sSWBdetect.WB_detected * psEnc->sCmn.fs_kHz != 16 ) ) &&
( ( psEnc->sCmn.fs_kHz == 16 ) && ( max_internal_fs_kHz >= 24 ) ||
( psEnc->sCmn.fs_kHz == 12 ) && ( max_internal_fs_kHz >= 16 ) ||
( psEnc->sCmn.fs_kHz == 8 ) && ( max_internal_fs_kHz >= 12 ) )
#if SWITCH_TRANSITION_FILTERING
&& ( psEnc->sCmn.sLP.transition_frame_no == 0 ) ) { /* No transition phase running, ready to switch */
psEnc->sCmn.sLP.mode = 1; /* Switch up */
#else
) {
#endif
psEnc->sCmn.bitrateDiff = 0;
/* Switch to a higher sample frequency */
if( psEnc->sCmn.fs_kHz == 8 ) {
fs_kHz = 12;
} else if( psEnc->sCmn.fs_kHz == 12 ) {
fs_kHz = 16;
} else {
SKP_assert( psEnc->sCmn.fs_kHz == 16 );
fs_kHz = 24;
}
}
}
}
/* Encode side-information parameters to payload */
void silk_encode_indices(
silk_encoder_state *psEncC, /* I/O Encoder state */
ec_enc *psRangeEnc, /* I/O Compressor data structure */
opus_int FrameIndex, /* I Frame number */
opus_int encode_LBRR /* I Flag indicating LBRR data is being encoded */
)
{
opus_int i, k, condCoding, typeOffset;
opus_int encode_absolute_lagIndex, delta_lagIndex;
opus_int16 ec_ix[ MAX_LPC_ORDER ];
opus_uint8 pred_Q8[ MAX_LPC_ORDER ];
const SideInfoIndices *psIndices;
#if SAVE_ALL_INTERNAL_DATA
opus_int nBytes_lagIndex, nBytes_contourIndex, nBytes_LTP;
opus_int nBytes_after, nBytes_before;
#endif
/* Use conditional coding if previous frame available */
if( FrameIndex > 0 && ( encode_LBRR == 0 || psEncC->LBRR_flags[ FrameIndex - 1 ] == 1 ) ) {
condCoding = 1;
} else {
condCoding = 0;
}
if( encode_LBRR ) {
psIndices = &psEncC->indices_LBRR[ FrameIndex ];
} else {
psIndices = &psEncC->indices;
}
/*******************************************/
/* Encode signal type and quantizer offset */
/*******************************************/
typeOffset = 2 * psIndices->signalType + psIndices->quantOffsetType;
SKP_assert( typeOffset >= 0 && typeOffset < 6 );
SKP_assert( encode_LBRR == 0 || typeOffset >= 2 );
if( encode_LBRR || typeOffset >= 2 ) {
ec_enc_icdf( psRangeEnc, typeOffset - 2, silk_type_offset_VAD_iCDF, 8 );
} else {
ec_enc_icdf( psRangeEnc, typeOffset, silk_type_offset_no_VAD_iCDF, 8 );
}
/****************/
/* Encode gains */
/****************/
#ifdef SAVE_ALL_INTERNAL_DATA
nBytes_before = SKP_RSHIFT( ec_tell( psRangeEnc ) + 7, 3 );
#endif
/* first subframe */
if( condCoding ) {
/* conditional coding */
SKP_assert( psIndices->GainsIndices[ 0 ] >= 0 && psIndices->GainsIndices[ 0 ] < MAX_DELTA_GAIN_QUANT - MIN_DELTA_GAIN_QUANT + 1 );
ec_enc_icdf( psRangeEnc, psIndices->GainsIndices[ 0 ], silk_delta_gain_iCDF, 8 );
} else {
/* independent coding, in two stages: MSB bits followed by 3 LSBs */
SKP_assert( psIndices->GainsIndices[ 0 ] >= 0 && psIndices->GainsIndices[ 0 ] < N_LEVELS_QGAIN );
ec_enc_icdf( psRangeEnc, SKP_RSHIFT( psIndices->GainsIndices[ 0 ], 3 ), silk_gain_iCDF[ psIndices->signalType ], 8 );
ec_enc_icdf( psRangeEnc, psIndices->GainsIndices[ 0 ] & 7, silk_uniform8_iCDF, 8 );
}
/* remaining subframes */
for( i = 1; i < psEncC->nb_subfr; i++ ) {
SKP_assert( psIndices->GainsIndices[ i ] >= 0 && psIndices->GainsIndices[ i ] < MAX_DELTA_GAIN_QUANT - MIN_DELTA_GAIN_QUANT + 1 );
ec_enc_icdf( psRangeEnc, psIndices->GainsIndices[ i ], silk_delta_gain_iCDF, 8 );
}
#ifdef SAVE_ALL_INTERNAL_DATA
nBytes_after = SKP_RSHIFT( ec_tell( psRangeEnc ) + 7, 3 );
nBytes_after -= nBytes_before; // bytes just added
DEBUG_STORE_DATA( nBytes_gains.dat, &nBytes_after, sizeof( opus_int ) );
#endif
/****************/
/* Encode NLSFs */
/****************/
#ifdef SAVE_ALL_INTERNAL_DATA
nBytes_before = SKP_RSHIFT( ec_tell( psRangeEnc ) + 7, 3 );
#endif
ec_enc_icdf( psRangeEnc, psIndices->NLSFIndices[ 0 ], &psEncC->psNLSF_CB->CB1_iCDF[ ( psIndices->signalType >> 1 ) * psEncC->psNLSF_CB->nVectors ], 8 );
silk_NLSF_unpack( ec_ix, pred_Q8, psEncC->psNLSF_CB, psIndices->NLSFIndices[ 0 ] );
SKP_assert( psEncC->psNLSF_CB->order == psEncC->predictLPCOrder );
for( i = 0; i < psEncC->psNLSF_CB->order; i++ ) {
if( psIndices->NLSFIndices[ i+1 ] >= NLSF_QUANT_MAX_AMPLITUDE ) {
ec_enc_icdf( psRangeEnc, 2 * NLSF_QUANT_MAX_AMPLITUDE, &psEncC->psNLSF_CB->ec_iCDF[ ec_ix[ i ] ], 8 );
ec_enc_icdf( psRangeEnc, psIndices->NLSFIndices[ i+1 ] - NLSF_QUANT_MAX_AMPLITUDE, silk_NLSF_EXT_iCDF, 8 );
} else if( psIndices->NLSFIndices[ i+1 ] <= -NLSF_QUANT_MAX_AMPLITUDE ) {
ec_enc_icdf( psRangeEnc, 0, &psEncC->psNLSF_CB->ec_iCDF[ ec_ix[ i ] ], 8 );
ec_enc_icdf( psRangeEnc, -psIndices->NLSFIndices[ i+1 ] - NLSF_QUANT_MAX_AMPLITUDE, silk_NLSF_EXT_iCDF, 8 );
} else {
ec_enc_icdf( psRangeEnc, psIndices->NLSFIndices[ i+1 ] + NLSF_QUANT_MAX_AMPLITUDE, &psEncC->psNLSF_CB->ec_iCDF[ ec_ix[ i ] ], 8 );
}
}
/* Encode NLSF interpolation factor */
if( psEncC->nb_subfr == MAX_NB_SUBFR ) {
SKP_assert( psEncC->useInterpolatedNLSFs == 1 || psIndices->NLSFInterpCoef_Q2 == ( 1 << 2 ) );
SKP_assert( psIndices->NLSFInterpCoef_Q2 >= 0 && psIndices->NLSFInterpCoef_Q2 < 5 );
ec_enc_icdf( psRangeEnc, psIndices->NLSFInterpCoef_Q2, silk_NLSF_interpolation_factor_iCDF, 8 );
}
#ifdef SAVE_ALL_INTERNAL_DATA
DEBUG_STORE_DATA( lsf_interpol.dat, &psIndices->NLSFInterpCoef_Q2, sizeof(int) );
nBytes_after = SKP_RSHIFT( ec_tell( psRangeEnc ) + 7, 3 );
nBytes_after -= nBytes_before; // bytes just added
DEBUG_STORE_DATA( nBytes_LSF.dat, &nBytes_after, sizeof( opus_int ) );
#endif
if( psIndices->signalType == TYPE_VOICED )
{
/*********************/
/* Encode pitch lags */
/*********************/
#ifdef SAVE_ALL_INTERNAL_DATA
nBytes_before = SKP_RSHIFT( ec_tell( psRangeEnc ) + 7, 3 );
#endif
/* lag index */
encode_absolute_lagIndex = 1;
if( condCoding && psEncC->ec_prevSignalType == TYPE_VOICED ) {
/* Delta Encoding */
delta_lagIndex = psIndices->lagIndex - psEncC->ec_prevLagIndex;
if( delta_lagIndex < -8 || delta_lagIndex > 11 ) {
delta_lagIndex = 0;
} else {
delta_lagIndex = delta_lagIndex + 9;
encode_absolute_lagIndex = 0; /* Only use delta */
}
SKP_assert( delta_lagIndex >= 0 && delta_lagIndex < 21 );
ec_enc_icdf( psRangeEnc, delta_lagIndex, silk_pitch_delta_iCDF, 8 );
}
if( encode_absolute_lagIndex ) {
/* Absolute encoding */
opus_int32 pitch_high_bits, pitch_low_bits;
pitch_high_bits = SKP_DIV32_16( psIndices->lagIndex, SKP_RSHIFT( psEncC->fs_kHz, 1 ) );
pitch_low_bits = psIndices->lagIndex - SKP_SMULBB( pitch_high_bits, SKP_RSHIFT( psEncC->fs_kHz, 1 ) );
SKP_assert( pitch_low_bits < psEncC->fs_kHz / 2 );
SKP_assert( pitch_high_bits < 32 );
ec_enc_icdf( psRangeEnc, pitch_high_bits, silk_pitch_lag_iCDF, 8 );
ec_enc_icdf( psRangeEnc, pitch_low_bits, psEncC->pitch_lag_low_bits_iCDF, 8 );
}
psEncC->ec_prevLagIndex = psIndices->lagIndex;
#ifdef SAVE_ALL_INTERNAL_DATA
nBytes_after = SKP_RSHIFT( ec_tell( psRangeEnc ) + 7, 3 );
nBytes_lagIndex = nBytes_after - nBytes_before; // bytes just added
#endif
#ifdef SAVE_ALL_INTERNAL_DATA
nBytes_before = SKP_RSHIFT( ec_tell( psRangeEnc ) + 7, 3 );
#endif
/* Countour index */
SKP_assert( psIndices->contourIndex >= 0 );
SKP_assert( ( psIndices->contourIndex < 34 && psEncC->fs_kHz > 8 && psEncC->nb_subfr == 4 ) ||
( psIndices->contourIndex < 11 && psEncC->fs_kHz == 8 && psEncC->nb_subfr == 4 ) ||
( psIndices->contourIndex < 12 && psEncC->fs_kHz > 8 && psEncC->nb_subfr == 2 ) ||
( psIndices->contourIndex < 3 && psEncC->fs_kHz == 8 && psEncC->nb_subfr == 2 ) );
ec_enc_icdf( psRangeEnc, psIndices->contourIndex, psEncC->pitch_contour_iCDF, 8 );
#ifdef SAVE_ALL_INTERNAL_DATA
nBytes_after = SKP_RSHIFT( ec_tell( psRangeEnc ) + 7, 3 );
nBytes_contourIndex = nBytes_after - nBytes_before; // bytes just added
#endif
/********************/
/* Encode LTP gains */
/********************/
#ifdef SAVE_ALL_INTERNAL_DATA
nBytes_before = SKP_RSHIFT( ec_tell( psRangeEnc ) + 7, 3 );
#endif
/* PERIndex value */
SKP_assert( psIndices->PERIndex >= 0 && psIndices->PERIndex < 3 );
ec_enc_icdf( psRangeEnc, psIndices->PERIndex, silk_LTP_per_index_iCDF, 8 );
/* Codebook Indices */
for( k = 0; k < psEncC->nb_subfr; k++ ) {
SKP_assert( psIndices->LTPIndex[ k ] >= 0 && psIndices->LTPIndex[ k ] < ( 8 << psIndices->PERIndex ) );
ec_enc_icdf( psRangeEnc, psIndices->LTPIndex[ k ], silk_LTP_gain_iCDF_ptrs[ psIndices->PERIndex ], 8 );
}
/**********************/
/* Encode LTP scaling */
/**********************/
if( !condCoding ) {
SKP_assert( psIndices->LTP_scaleIndex >= 0 && psIndices->LTP_scaleIndex < 3 );
ec_enc_icdf( psRangeEnc, psIndices->LTP_scaleIndex, silk_LTPscale_iCDF, 8 );
}
SKP_assert( !condCoding || psIndices->LTP_scaleIndex == 0 );
#ifdef SAVE_ALL_INTERNAL_DATA
nBytes_after = SKP_RSHIFT( ec_tell( psRangeEnc ) + 7, 3 );
nBytes_LTP = nBytes_after - nBytes_before; // bytes just added
#endif
}
#ifdef SAVE_ALL_INTERNAL_DATA
else {
// Unvoiced speech
nBytes_lagIndex = 0;
nBytes_contourIndex = 0;
nBytes_LTP = 0;
}
DEBUG_STORE_DATA( nBytes_lagIndex.dat, &nBytes_lagIndex, sizeof( opus_int ) );
DEBUG_STORE_DATA( nBytes_contourIndex.dat, &nBytes_contourIndex, sizeof( opus_int ) );
DEBUG_STORE_DATA( nBytes_LTP.dat, &nBytes_LTP, sizeof( opus_int ) );
#endif
psEncC->ec_prevSignalType = psIndices->signalType;
#ifdef SAVE_ALL_INTERNAL_DATA
nBytes_before = SKP_RSHIFT( ec_tell( psRangeEnc ) + 7, 3 );
#endif
/***************/
/* Encode seed */
/***************/
SKP_assert( psIndices->Seed >= 0 && psIndices->Seed < 4 );
ec_enc_icdf( psRangeEnc, psIndices->Seed, silk_uniform4_iCDF, 8 );
}
/* Control internal sampling rate */
SKP_int SKP_Silk_control_audio_bandwidth(
SKP_Silk_encoder_state *psEncC, /* I/O Pointer to Silk encoder state */
SKP_int32 TargetRate_bps /* I Target max bitrate (bps) */
)
{
SKP_int fs_kHz;
fs_kHz = psEncC->fs_kHz;
/* Reduce bitrate for 10 ms modes in these calculations */
if( psEncC->nb_subfr == 2 ) {
TargetRate_bps -= REDUCE_BITRATE_10_MS_BPS;
}
if( fs_kHz == 0 ) {
/* Encoder has just been initialized */
if( TargetRate_bps >= WB2MB_BITRATE_BPS ) {
fs_kHz = 16;
} else if( TargetRate_bps >= MB2NB_BITRATE_BPS ) {
fs_kHz = 12;
} else {
fs_kHz = 8;
}
/* Make sure internal rate is not higher than external rate or maximum allowed, or lower than minimum allowed */
fs_kHz = SKP_min( fs_kHz, SKP_DIV32_16( psEncC->API_fs_Hz, 1000 ) );
fs_kHz = SKP_min( fs_kHz, psEncC->maxInternal_fs_kHz );
fs_kHz = SKP_max( fs_kHz, psEncC->minInternal_fs_kHz );
} else if( SKP_SMULBB( fs_kHz, 1000 ) > psEncC->API_fs_Hz || fs_kHz > psEncC->maxInternal_fs_kHz || fs_kHz < psEncC->minInternal_fs_kHz ) {
/* Make sure internal rate is not higher than external rate or maximum allowed, or lower than minimum allowed */
fs_kHz = SKP_DIV32_16( psEncC->API_fs_Hz, 1000 );
fs_kHz = SKP_min( fs_kHz, psEncC->maxInternal_fs_kHz );
fs_kHz = SKP_max( fs_kHz, psEncC->minInternal_fs_kHz );
} else {
/* State machine for the internal sampling rate switching */
if( psEncC->API_fs_Hz > 8000 && psEncC->prevSignalType == TYPE_NO_VOICE_ACTIVITY ) { /* Low speech activity */
/* Check if we should switch down */
if( ( psEncC->fs_kHz == 12 && TargetRate_bps < MB2NB_BITRATE_BPS && psEncC->minInternal_fs_kHz <= 8 ) ||
( psEncC->fs_kHz == 16 && TargetRate_bps < WB2MB_BITRATE_BPS && psEncC->minInternal_fs_kHz <= 12 ) )
{
/* Switch down */
if( SWITCH_TRANSITION_FILTERING && psEncC->sLP.mode == 0 ) {
/* New transition */
psEncC->sLP.transition_frame_no = TRANSITION_FRAMES;
/* Reset transition filter state */
SKP_memset( psEncC->sLP.In_LP_State, 0, sizeof( psEncC->sLP.In_LP_State ) );
}
if( psEncC->sLP.transition_frame_no <= 0 ) {
/* Stop transition phase */
psEncC->sLP.mode = 0;
/* Switch to a lower sample frequency */
fs_kHz = psEncC->fs_kHz == 16 ? 12 : 8;
} else {
/* Direction: down (at double speed) */
psEncC->sLP.mode = -2;
}
}
else
if( ( psEncC->fs_kHz == 8 && TargetRate_bps > NB2MB_BITRATE_BPS && psEncC->maxInternal_fs_kHz >= 12 && psEncC->API_fs_Hz >= 12000 ) ||
( psEncC->fs_kHz == 12 && TargetRate_bps > MB2WB_BITRATE_BPS && psEncC->maxInternal_fs_kHz >= 16 && psEncC->API_fs_Hz >= 16000 ) )
{
/* Switch up */
if( SWITCH_TRANSITION_FILTERING && psEncC->sLP.mode == 0 ) {
/* Switch to a higher sample frequency */
fs_kHz = psEncC->fs_kHz == 8 ? 12 : 16;
/* New transition */
psEncC->sLP.transition_frame_no = 0;
}
if( psEncC->sLP.transition_frame_no >= TRANSITION_FRAMES ) {
/* Stop transition phase */
psEncC->sLP.mode = 0;
} else {
/* Direction: up */
psEncC->sLP.mode = 1;
}
}
}
}
#ifdef FORCE_INTERNAL_FS_KHZ
fs_kHz = FORCE_INTERNAL_FS_KHZ;
#endif
return fs_kHz;
}
void SKP_Silk_NLSF_MSVQ_encode_FIX(
SKP_int *NLSFIndices, /* O Codebook path vector [ CB_STAGES ] */
SKP_int *pNLSF_Q15, /* I/O Quantized NLSF vector [ LPC_ORDER ] */
const SKP_Silk_NLSF_CB_struct *psNLSF_CB, /* I Codebook object */
const SKP_int *pNLSF_q_Q15_prev, /* I Prev. quantized NLSF vector [LPC_ORDER] */
const SKP_int *pW_Q6, /* I NLSF weight vector [ LPC_ORDER ] */
const SKP_int NLSF_mu_Q15, /* I Rate weight for the RD optimization */
const SKP_int NLSF_mu_fluc_red_Q16, /* I Fluctuation reduction error weight */
const SKP_int NLSF_MSVQ_Survivors, /* I Max survivors from each stage */
const SKP_int LPC_order, /* I LPC order */
const SKP_int deactivate_fluc_red /* I Deactivate fluctuation reduction */
)
{
SKP_int i, s, k, cur_survivors = 0, prev_survivors, input_index, cb_index, bestIndex;
SKP_int32 rateDistThreshold_Q18;
SKP_int pNLSF_in_Q15[ MAX_LPC_ORDER ];
#if( NLSF_MSVQ_FLUCTUATION_REDUCTION == 1 )
SKP_int32 se_Q15, wsse_Q20, bestRateDist_Q20;
#endif
#if( LOW_COMPLEXITY_ONLY == 1 )
SKP_int32 pRateDist_Q18[ NLSF_MSVQ_TREE_SEARCH_MAX_VECTORS_EVALUATED_LC_MODE ];
SKP_int32 pRate_Q5[ MAX_NLSF_MSVQ_SURVIVORS_LC_MODE ];
SKP_int32 pRate_new_Q5[ MAX_NLSF_MSVQ_SURVIVORS_LC_MODE ];
SKP_int pTempIndices[ MAX_NLSF_MSVQ_SURVIVORS_LC_MODE ];
SKP_int pPath[ MAX_NLSF_MSVQ_SURVIVORS_LC_MODE * NLSF_MSVQ_MAX_CB_STAGES ];
SKP_int pPath_new[ MAX_NLSF_MSVQ_SURVIVORS_LC_MODE * NLSF_MSVQ_MAX_CB_STAGES ];
SKP_int pRes_Q15[ MAX_NLSF_MSVQ_SURVIVORS_LC_MODE * MAX_LPC_ORDER ];
SKP_int pRes_new_Q15[ MAX_NLSF_MSVQ_SURVIVORS_LC_MODE * MAX_LPC_ORDER ];
#else
SKP_int32 pRateDist_Q18[ NLSF_MSVQ_TREE_SEARCH_MAX_VECTORS_EVALUATED ];
SKP_int32 pRate_Q5[ MAX_NLSF_MSVQ_SURVIVORS ];
SKP_int32 pRate_new_Q5[ MAX_NLSF_MSVQ_SURVIVORS ];
SKP_int pTempIndices[ MAX_NLSF_MSVQ_SURVIVORS ];
SKP_int pPath[ MAX_NLSF_MSVQ_SURVIVORS * NLSF_MSVQ_MAX_CB_STAGES ];
SKP_int pPath_new[ MAX_NLSF_MSVQ_SURVIVORS * NLSF_MSVQ_MAX_CB_STAGES ];
SKP_int pRes_Q15[ MAX_NLSF_MSVQ_SURVIVORS * MAX_LPC_ORDER ];
SKP_int pRes_new_Q15[ MAX_NLSF_MSVQ_SURVIVORS * MAX_LPC_ORDER ];
#endif
const SKP_int *pConstInt;
SKP_int *pInt;
const SKP_int16 *pCB_element;
const SKP_Silk_NLSF_CBS *pCurrentCBStage;
SKP_assert( NLSF_MSVQ_Survivors <= MAX_NLSF_MSVQ_SURVIVORS );
SKP_assert( ( LOW_COMPLEXITY_ONLY == 0 ) || ( NLSF_MSVQ_Survivors <= MAX_NLSF_MSVQ_SURVIVORS_LC_MODE ) );
/* Copy the input vector */
SKP_memcpy( pNLSF_in_Q15, pNLSF_Q15, LPC_order * sizeof( SKP_int ) );
/****************************************************/
/* Tree search for the multi-stage vector quantizer */
/****************************************************/
/* Clear accumulated rates */
SKP_memset( pRate_Q5, 0, NLSF_MSVQ_Survivors * sizeof( SKP_int32 ) );
/* Copy NLSFs into residual signal vector */
for( i = 0; i < LPC_order; i++ ) {
pRes_Q15[ i ] = pNLSF_Q15[ i ];
}
/* Set first stage values */
prev_survivors = 1;
/* Loop over all stages */
for( s = 0; s < psNLSF_CB->nStages; s++ ) {
/* Set a pointer to the current stage codebook */
pCurrentCBStage = &psNLSF_CB->CBStages[ s ];
/* Calculate the number of survivors in the current stage */
cur_survivors = SKP_min_32( NLSF_MSVQ_Survivors, SKP_SMULBB( prev_survivors, pCurrentCBStage->nVectors ) );
#if( NLSF_MSVQ_FLUCTUATION_REDUCTION == 0 )
/* Find a single best survivor in the last stage, if we */
/* do not need candidates for fluctuation reduction */
if( s == psNLSF_CB->nStages - 1 ) {
cur_survivors = 1;
}
#endif
/* Nearest neighbor clustering for multiple input data vectors */
SKP_Silk_NLSF_VQ_rate_distortion_FIX( pRateDist_Q18, pCurrentCBStage, pRes_Q15, pW_Q6,
pRate_Q5, NLSF_mu_Q15, prev_survivors, LPC_order );
/* Sort the rate-distortion errors */
SKP_Silk_insertion_sort_increasing( pRateDist_Q18, pTempIndices,
prev_survivors * pCurrentCBStage->nVectors, cur_survivors );
/* Discard survivors with rate-distortion values too far above the best one */
if( pRateDist_Q18[ 0 ] < SKP_int32_MAX / NLSF_MSVQ_SURV_MAX_REL_RD ) {
rateDistThreshold_Q18 = SKP_MUL( NLSF_MSVQ_SURV_MAX_REL_RD, pRateDist_Q18[ 0 ] );
while( pRateDist_Q18[ cur_survivors - 1 ] > rateDistThreshold_Q18 && cur_survivors > 1 ) {
cur_survivors--;
}
}
/* Update accumulated codebook contributions for the 'cur_survivors' best codebook indices */
for( k = 0; k < cur_survivors; k++ ) {
if( s > 0 ) {
/* Find the indices of the input and the codebook vector */
if( pCurrentCBStage->nVectors == 8 ) {
input_index = SKP_RSHIFT( pTempIndices[ k ], 3 );
cb_index = pTempIndices[ k ] & 7;
} else {
input_index = SKP_DIV32_16( pTempIndices[ k ], pCurrentCBStage->nVectors );
cb_index = pTempIndices[ k ] - SKP_SMULBB( input_index, pCurrentCBStage->nVectors );
}
} else {
/* Find the indices of the input and the codebook vector */
input_index = 0;
cb_index = pTempIndices[ k ];
}
/* Subtract new contribution from the previous residual vector for each of 'cur_survivors' */
pConstInt = &pRes_Q15[ SKP_SMULBB( input_index, LPC_order ) ];
pCB_element = &pCurrentCBStage->CB_NLSF_Q15[ SKP_SMULBB( cb_index, LPC_order ) ];
pInt = &pRes_new_Q15[ SKP_SMULBB( k, LPC_order ) ];
for( i = 0; i < LPC_order; i++ ) {
pInt[ i ] = pConstInt[ i ] - ( SKP_int )pCB_element[ i ];
}
/* Update accumulated rate for stage 1 to the current */
pRate_new_Q5[ k ] = pRate_Q5[ input_index ] + pCurrentCBStage->Rates_Q5[ cb_index ];
/* Copy paths from previous matrix, starting with the best path */
pConstInt = &pPath[ SKP_SMULBB( input_index, psNLSF_CB->nStages ) ];
pInt = &pPath_new[ SKP_SMULBB( k, psNLSF_CB->nStages ) ];
for( i = 0; i < s; i++ ) {
pInt[ i ] = pConstInt[ i ];
}
/* Write the current stage indices for the 'cur_survivors' to the best path matrix */
pInt[ s ] = cb_index;
}
if( s < psNLSF_CB->nStages - 1 ) {
/* Copy NLSF residual matrix for next stage */
SKP_memcpy( pRes_Q15, pRes_new_Q15, SKP_SMULBB( cur_survivors, LPC_order ) * sizeof( SKP_int ) );
/* Copy rate vector for next stage */
SKP_memcpy( pRate_Q5, pRate_new_Q5, cur_survivors * sizeof( SKP_int32 ) );
/* Copy best path matrix for next stage */
SKP_memcpy( pPath, pPath_new, SKP_SMULBB( cur_survivors, psNLSF_CB->nStages ) * sizeof( SKP_int ) );
}
prev_survivors = cur_survivors;
}
/* (Preliminary) index of the best survivor, later to be decoded */
bestIndex = 0;
#if( NLSF_MSVQ_FLUCTUATION_REDUCTION == 1 )
/******************************/
/* NLSF fluctuation reduction */
/******************************/
if( deactivate_fluc_red != 1 ) {
/* Search among all survivors, now taking also weighted fluctuation errors into account */
bestRateDist_Q20 = SKP_int32_MAX;
for( s = 0; s < cur_survivors; s++ ) {
/* Decode survivor to compare with previous quantized NLSF vector */
SKP_Silk_NLSF_MSVQ_decode( pNLSF_Q15, psNLSF_CB, &pPath_new[ SKP_SMULBB( s, psNLSF_CB->nStages ) ], LPC_order );
/* Compare decoded NLSF vector with the previously quantized vector */
wsse_Q20 = 0;
for( i = 0; i < LPC_order; i += 2 ) {
/* Compute weighted squared quantization error for index i */
se_Q15 = pNLSF_Q15[ i ] - pNLSF_q_Q15_prev[ i ]; // range: [ -32767 : 32767 ]
wsse_Q20 = SKP_SMLAWB( wsse_Q20, SKP_SMULBB( se_Q15, se_Q15 ), pW_Q6[ i ] );
/* Compute weighted squared quantization error for index i + 1 */
se_Q15 = pNLSF_Q15[ i + 1 ] - pNLSF_q_Q15_prev[ i + 1 ]; // range: [ -32767 : 32767 ]
wsse_Q20 = SKP_SMLAWB( wsse_Q20, SKP_SMULBB( se_Q15, se_Q15 ), pW_Q6[ i + 1 ] );
}
SKP_assert( wsse_Q20 >= 0 );
/* Add the fluctuation reduction penalty to the rate distortion error */
wsse_Q20 = SKP_ADD_POS_SAT32( pRateDist_Q18[ s ], SKP_SMULWB( wsse_Q20, NLSF_mu_fluc_red_Q16 ) );
/* Keep index of best survivor */
if( wsse_Q20 < bestRateDist_Q20 ) {
bestRateDist_Q20 = wsse_Q20;
bestIndex = s;
}
}
}
#endif
/* Copy best path to output argument */
SKP_memcpy( NLSFIndices, &pPath_new[ SKP_SMULBB( bestIndex, psNLSF_CB->nStages ) ], psNLSF_CB->nStages * sizeof( SKP_int ) );
/* Decode and stabilize the best survivor */
SKP_Silk_NLSF_MSVQ_decode( pNLSF_Q15, psNLSF_CB, NLSFIndices, LPC_order );
}
SKP_int SKP_Silk_SDK_Encode(
void *encState, /* I/O: State */
SKP_Silk_EncodeControlStruct *encControl, /* I: Control structure */
const SKP_int16 *samplesIn, /* I: Speech sample input vector */
SKP_int nSamplesIn, /* I: Number of samples in input vector */
ec_enc *psRangeEnc, /* I/O Compressor data structure */
SKP_int32 *nBytesOut, /* I/O: Number of bytes in payload (input: Max bytes) */
const SKP_int prefillFlag /* I: Flag to indicate prefilling buffers no coding */
)
{
SKP_int tmp_payloadSize_ms, tmp_complexity, ret = 0;
SKP_int nSamplesToBuffer, nBlocksOf10ms, nSamplesFromInput = 0;
SKP_Silk_encoder_state_Fxx *psEnc = ( SKP_Silk_encoder_state_Fxx* )encState;
ret = process_enc_control_struct( psEnc, encControl );
nBlocksOf10ms = SKP_DIV32( 100 * nSamplesIn, psEnc->sCmn.API_fs_Hz );
if( prefillFlag ) {
/* Only accept input length of 10 ms */
if( nBlocksOf10ms != 1 ) {
ret = SKP_SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
SKP_assert( 0 );
return( ret );
}
/* Reset Encoder */
if( ret = SKP_Silk_init_encoder_Fxx( psEnc ) ) {
SKP_assert( 0 );
}
tmp_payloadSize_ms = encControl->payloadSize_ms;
encControl->payloadSize_ms = 10;
tmp_complexity = encControl->complexity;
encControl->complexity = 0;
ret = process_enc_control_struct( psEnc, encControl );
psEnc->sCmn.prefillFlag = 1;
} else {
/* Only accept input lengths that are a multiple of 10 ms */
if( nBlocksOf10ms * psEnc->sCmn.API_fs_Hz != 100 * nSamplesIn || nSamplesIn < 0 ) {
ret = SKP_SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
SKP_assert( 0 );
return( ret );
}
/* Make sure no more than one packet can be produced */
if( 1000 * (SKP_int32)nSamplesIn > psEnc->sCmn.PacketSize_ms * psEnc->sCmn.API_fs_Hz ) {
ret = SKP_SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
SKP_assert( 0 );
return( ret );
}
}
/* Input buffering/resampling and encoding */
while( 1 ) {
nSamplesToBuffer = psEnc->sCmn.frame_length - psEnc->sCmn.inputBufIx;
if( psEnc->sCmn.API_fs_Hz == SKP_SMULBB( 1000, psEnc->sCmn.fs_kHz ) ) {
nSamplesToBuffer = SKP_min_int( nSamplesToBuffer, nSamplesIn );
nSamplesFromInput = nSamplesToBuffer;
/* Copy to buffer */
SKP_memcpy( &psEnc->sCmn.inputBuf[ psEnc->sCmn.inputBufIx ], samplesIn, nSamplesFromInput * sizeof( SKP_int16 ) );
} else {
nSamplesToBuffer = SKP_min( nSamplesToBuffer, 10 * nBlocksOf10ms * psEnc->sCmn.fs_kHz );
nSamplesFromInput = SKP_DIV32_16( nSamplesToBuffer * psEnc->sCmn.API_fs_Hz, psEnc->sCmn.fs_kHz * 1000 );
/* Resample and write to buffer */
ret += SKP_Silk_resampler( &psEnc->sCmn.resampler_state, &psEnc->sCmn.inputBuf[ psEnc->sCmn.inputBufIx ], samplesIn, nSamplesFromInput );
}
samplesIn += nSamplesFromInput;
nSamplesIn -= nSamplesFromInput;
psEnc->sCmn.inputBufIx += nSamplesToBuffer;
/* Silk encoder */
if( psEnc->sCmn.inputBufIx >= psEnc->sCmn.frame_length ) {
SKP_assert( psEnc->sCmn.inputBufIx == psEnc->sCmn.frame_length );
/* Enough data in input buffer, so encode */
if( ( ret = SKP_Silk_encode_frame_Fxx( psEnc, nBytesOut, psRangeEnc ) ) != 0 ) {
SKP_assert( 0 );
}
psEnc->sCmn.inputBufIx = 0;
psEnc->sCmn.controlled_since_last_payload = 0;
if( nSamplesIn == 0 ) {
break;
}
} else {
break;
}
}
if( prefillFlag ) {
encControl->payloadSize_ms = tmp_payloadSize_ms;
encControl->complexity = tmp_complexity;
ret = process_enc_control_struct( psEnc, encControl );
psEnc->sCmn.prefillFlag = 0;
}
return ret;
}
/* Deactivate by setting psEncC->transition_frame_no = 0; */
void SKP_Silk_LP_variable_cutoff(
SKP_Silk_LP_state *psLP, /* I/O LP filter state */
SKP_int16 *out, /* O Low-pass filtered output signal */
const SKP_int16 *in, /* I Input signal */
const SKP_int frame_length /* I Frame length */
)
{
SKP_int32 B_Q28[ TRANSITION_NB ], A_Q28[ TRANSITION_NA ], fac_Q16 = 0;
SKP_int ind = 0;
SKP_assert( psLP->transition_frame_no >= 0 );
SKP_assert( ( ( ( psLP->transition_frame_no <= TRANSITION_FRAMES_DOWN ) && ( psLP->mode == 0 ) ) ||
( ( psLP->transition_frame_no <= TRANSITION_FRAMES_UP ) && ( psLP->mode == 1 ) ) ) );
/* Interpolate filter coefficients if needed */
if( psLP->transition_frame_no > 0 ) {
if( psLP->mode == 0 ) {
if( psLP->transition_frame_no < TRANSITION_FRAMES_DOWN ) {
/* Calculate index and interpolation factor for interpolation */
#if( TRANSITION_INT_STEPS_DOWN == 32 )
fac_Q16 = SKP_LSHIFT( psLP->transition_frame_no, 16 - 5 );
#else
fac_Q16 = SKP_DIV32_16( SKP_LSHIFT( psLP->transition_frame_no, 16 ), TRANSITION_INT_STEPS_DOWN );
#endif
ind = SKP_RSHIFT( fac_Q16, 16 );
fac_Q16 -= SKP_LSHIFT( ind, 16 );
SKP_assert( ind >= 0 );
SKP_assert( ind < TRANSITION_INT_NUM );
/* Interpolate filter coefficients */
SKP_Silk_LP_interpolate_filter_taps( B_Q28, A_Q28, ind, fac_Q16 );
/* Increment transition frame number for next frame */
psLP->transition_frame_no++;
} else if( psLP->transition_frame_no == TRANSITION_FRAMES_DOWN ) {
/* End of transition phase */
SKP_Silk_LP_interpolate_filter_taps( B_Q28, A_Q28, TRANSITION_INT_NUM - 1, 0 );
}
} else if( psLP->mode == 1 ) {
if( psLP->transition_frame_no < TRANSITION_FRAMES_UP ) {
/* Calculate index and interpolation factor for interpolation */
#if( TRANSITION_INT_STEPS_UP == 64 )
fac_Q16 = SKP_LSHIFT( TRANSITION_FRAMES_UP - psLP->transition_frame_no, 16 - 6 );
#else
fac_Q16 = SKP_DIV32_16( SKP_LSHIFT( TRANSITION_FRAMES_UP - psLP->transition_frame_no, 16 ), TRANSITION_INT_STEPS_UP );
#endif
ind = SKP_RSHIFT( fac_Q16, 16 );
fac_Q16 -= SKP_LSHIFT( ind, 16 );
SKP_assert( ind >= 0 );
SKP_assert( ind < TRANSITION_INT_NUM );
/* Interpolate filter coefficients */
SKP_Silk_LP_interpolate_filter_taps( B_Q28, A_Q28, ind, fac_Q16 );
/* Increment transition frame number for next frame */
psLP->transition_frame_no++;
} else if( psLP->transition_frame_no == TRANSITION_FRAMES_UP ) {
/* End of transition phase */
SKP_Silk_LP_interpolate_filter_taps( B_Q28, A_Q28, 0, 0 );
}
}
}
if( psLP->transition_frame_no > 0 ) {
/* ARMA low-pass filtering */
SKP_assert( TRANSITION_NB == 3 && TRANSITION_NA == 2 );
SKP_Silk_biquad_alt( in, B_Q28, A_Q28, psLP->In_LP_State, out, frame_length );
} else {
/* Instead of using the filter, copy input directly to output */
SKP_memcpy( out, in, frame_length * sizeof( SKP_int16 ) );
}
}
/* High-pass filter with cutoff frequency adaptation based on pitch lag statistics */
void SKP_Silk_HP_variable_cutoff_FIX(
SKP_Silk_encoder_state_FIX *psEnc, /* I/O Encoder state FIX */
SKP_Silk_encoder_control_FIX *psEncCtrl, /* I/O Encoder control FIX */
SKP_int16 *out, /* O high-pass filtered output signal */
const SKP_int16 *in /* I input signal */
)
{
SKP_int quality_Q15;
SKP_int32 B_Q28[ 3 ], A_Q28[ 2 ];
SKP_int32 Fc_Q19, r_Q28, r_Q22;
SKP_int32 pitch_freq_Hz_Q16, pitch_freq_log_Q7, delta_freq_Q7;
/*********************************************/
/* Estimate Low End of Pitch Frequency Range */
/*********************************************/
if( psEnc->sCmn.prev_sigtype == SIG_TYPE_VOICED ) {
/* difference, in log domain */
pitch_freq_Hz_Q16 = SKP_DIV32_16( SKP_LSHIFT( SKP_MUL( psEnc->sCmn.fs_kHz, 1000 ), 16 ), psEnc->sCmn.prevLag );
pitch_freq_log_Q7 = SKP_Silk_lin2log( pitch_freq_Hz_Q16 ) - ( 16 << 7 ); //0x70
/* adjustment based on quality */
quality_Q15 = psEncCtrl->input_quality_bands_Q15[ 0 ];
pitch_freq_log_Q7 = SKP_SUB32( pitch_freq_log_Q7, SKP_SMULWB( SKP_SMULWB( SKP_LSHIFT( quality_Q15, 2 ), quality_Q15 ),
pitch_freq_log_Q7 - SKP_LOG2_VARIABLE_HP_MIN_FREQ_Q7 ) );
pitch_freq_log_Q7 = SKP_ADD32( pitch_freq_log_Q7, SKP_RSHIFT( SKP_FIX_CONST( 0.6, 15 ) - quality_Q15, 9 ) );
//delta_freq = pitch_freq_log - psEnc->variable_HP_smth1;
delta_freq_Q7 = pitch_freq_log_Q7 - SKP_RSHIFT( psEnc->variable_HP_smth1_Q15, 8 );
if( delta_freq_Q7 < 0 ) {
/* less smoothing for decreasing pitch frequency, to track something close to the minimum */
delta_freq_Q7 = SKP_MUL( delta_freq_Q7, 3 );
}
/* limit delta, to reduce impact of outliers */
delta_freq_Q7 = SKP_LIMIT_32( delta_freq_Q7, -SKP_FIX_CONST( VARIABLE_HP_MAX_DELTA_FREQ, 7 ), SKP_FIX_CONST( VARIABLE_HP_MAX_DELTA_FREQ, 7 ) );
/* update smoother */
psEnc->variable_HP_smth1_Q15 = SKP_SMLAWB( psEnc->variable_HP_smth1_Q15,
SKP_MUL( SKP_LSHIFT( psEnc->speech_activity_Q8, 1 ), delta_freq_Q7 ), SKP_FIX_CONST( VARIABLE_HP_SMTH_COEF1, 16 ) );
}
/* second smoother */
psEnc->variable_HP_smth2_Q15 = SKP_SMLAWB( psEnc->variable_HP_smth2_Q15,
psEnc->variable_HP_smth1_Q15 - psEnc->variable_HP_smth2_Q15, SKP_FIX_CONST( VARIABLE_HP_SMTH_COEF2, 16 ) );
/* convert from log scale to Hertz */
psEncCtrl->pitch_freq_low_Hz = SKP_Silk_log2lin( SKP_RSHIFT( psEnc->variable_HP_smth2_Q15, 8 ) );
/* limit frequency range */
psEncCtrl->pitch_freq_low_Hz = SKP_LIMIT_32( psEncCtrl->pitch_freq_low_Hz,
SKP_FIX_CONST( VARIABLE_HP_MIN_FREQ, 0 ), SKP_FIX_CONST( VARIABLE_HP_MAX_FREQ, 0 ) );
/********************************/
/* Compute Filter Coefficients */
/********************************/
/* compute cut-off frequency, in radians */
//Fc_num = (SKP_float)( 0.45f * 2.0f * 3.14159265359 * psEncCtrl->pitch_freq_low_Hz );
//Fc_denom = (SKP_float)( 1e3f * psEnc->sCmn.fs_kHz );
SKP_assert( psEncCtrl->pitch_freq_low_Hz <= SKP_int32_MAX / SKP_RADIANS_CONSTANT_Q19 );
Fc_Q19 = SKP_DIV32_16( SKP_SMULBB( SKP_RADIANS_CONSTANT_Q19, psEncCtrl->pitch_freq_low_Hz ), psEnc->sCmn.fs_kHz ); // range: 3704 - 27787, 11-15 bits
SKP_assert( Fc_Q19 >= 3704 );
SKP_assert( Fc_Q19 <= 27787 );
r_Q28 = SKP_FIX_CONST( 1.0, 28 ) - SKP_MUL( SKP_FIX_CONST( 0.92, 9 ), Fc_Q19 );
SKP_assert( r_Q28 >= 255347779 );
SKP_assert( r_Q28 <= 266690872 );
/* b = r * [ 1; -2; 1 ]; */
/* a = [ 1; -2 * r * ( 1 - 0.5 * Fc^2 ); r^2 ]; */
B_Q28[ 0 ] = r_Q28;
B_Q28[ 1 ] = SKP_LSHIFT( -r_Q28, 1 );
B_Q28[ 2 ] = r_Q28;
// -r * ( 2 - Fc * Fc );
r_Q22 = SKP_RSHIFT( r_Q28, 6 );
A_Q28[ 0 ] = SKP_SMULWW( r_Q22, SKP_SMULWW( Fc_Q19, Fc_Q19 ) - SKP_FIX_CONST( 2.0, 22 ) );
A_Q28[ 1 ] = SKP_SMULWW( r_Q22, r_Q22 );
/********************************/
/* High-Pass Filter */
/********************************/
SKP_Silk_biquad_alt( in, B_Q28, A_Q28, psEnc->sCmn.In_HP_State, out, psEnc->sCmn.frame_length );
}
/* Control internal sampling rate */
SKP_int silk_control_audio_bandwidth(
silk_encoder_state *psEncC /* I/O Pointer to Silk encoder state */
)
{
SKP_int fs_kHz;
SKP_int32 fs_Hz;
fs_kHz = psEncC->fs_kHz;
fs_Hz = SKP_SMULBB( fs_kHz, 1000 );
if( fs_Hz == 0 ) {
/* Encoder has just been initialized */
fs_Hz = SKP_min( psEncC->desiredInternal_fs_Hz, psEncC->API_fs_Hz );
fs_kHz = SKP_DIV32_16( fs_Hz, 1000 );
} else if( fs_Hz > psEncC->API_fs_Hz || fs_Hz > psEncC->maxInternal_fs_Hz || fs_Hz < psEncC->minInternal_fs_Hz ) {
/* Make sure internal rate is not higher than external rate or maximum allowed, or lower than minimum allowed */
fs_Hz = psEncC->API_fs_Hz;
fs_Hz = SKP_min( fs_Hz, psEncC->maxInternal_fs_Hz );
fs_Hz = SKP_max( fs_Hz, psEncC->minInternal_fs_Hz );
fs_kHz = SKP_DIV32_16( fs_Hz, 1000 );
} else {
/* State machine for the internal sampling rate switching */
if( psEncC->sLP.transition_frame_no >= TRANSITION_FRAMES ) {
/* Stop transition phase */
psEncC->sLP.mode = 0;
}
if( psEncC->allow_bandwidth_switch ) {
/* Check if we should switch down */
if( SKP_SMULBB( psEncC->fs_kHz, 1000 ) > psEncC->desiredInternal_fs_Hz )
{
/* Switch down */
if( psEncC->sLP.mode == 0 ) {
/* New transition */
psEncC->sLP.transition_frame_no = TRANSITION_FRAMES;
/* Reset transition filter state */
SKP_memset( psEncC->sLP.In_LP_State, 0, sizeof( psEncC->sLP.In_LP_State ) );
}
if( psEncC->sLP.transition_frame_no <= 0 ) {
/* Stop transition phase */
psEncC->sLP.mode = 0;
/* Switch to a lower sample frequency */
fs_kHz = psEncC->fs_kHz == 16 ? 12 : 8;
} else {
/* Direction: down (at double speed) */
psEncC->sLP.mode = -2;
}
}
else
/* Check if we should switch up */
if( SKP_SMULBB( psEncC->fs_kHz, 1000 ) < psEncC->desiredInternal_fs_Hz )
{
/* Switch up */
if( psEncC->sLP.mode == 0 ) {
/* Switch to a higher sample frequency */
fs_kHz = psEncC->fs_kHz == 8 ? 12 : 16;
/* New transition */
psEncC->sLP.transition_frame_no = 0;
/* Reset transition filter state */
SKP_memset( psEncC->sLP.In_LP_State, 0, sizeof( psEncC->sLP.In_LP_State ) );
}
/* Direction: up */
psEncC->sLP.mode = 1;
}
}
}
#ifdef FORCE_INTERNAL_FS_KHZ
fs_kHz = FORCE_INTERNAL_FS_KHZ;
#endif
return fs_kHz;
}
