/* Decode mid/side predictors */
void silk_stereo_decode_pred(
ec_dec *psRangeDec, /* I/O Compressor data structure */
opus_int32 pred_Q13[] /* O Predictors */
)
{
opus_int n, ix[ 2 ][ 3 ];
opus_int32 low_Q13, step_Q13;
/* Entropy decoding */
n = ec_dec_icdf( psRangeDec, silk_stereo_pred_joint_iCDF, 8 );
ix[ 0 ][ 2 ] = silk_DIV32_16( n, 5 );
ix[ 1 ][ 2 ] = n - 5 * ix[ 0 ][ 2 ];
for( n = 0; n < 2; n++ ) {
ix[ n ][ 0 ] = ec_dec_icdf( psRangeDec, silk_uniform3_iCDF, 8 );
ix[ n ][ 1 ] = ec_dec_icdf( psRangeDec, silk_uniform5_iCDF, 8 );
}
/* Dequantize */
for( n = 0; n < 2; n++ ) {
ix[ n ][ 0 ] += 3 * ix[ n ][ 2 ];
low_Q13 = silk_stereo_pred_quant_Q13[ ix[ n ][ 0 ] ];
step_Q13 = silk_SMULWB( silk_stereo_pred_quant_Q13[ ix[ n ][ 0 ] + 1 ] - low_Q13,
SILK_FIX_CONST( 0.5 / STEREO_QUANT_SUB_STEPS, 16 ) );
pred_Q13[ n ] = silk_SMLABB( low_Q13, step_Q13, 2 * ix[ n ][ 1 ] + 1 );
}
/* Subtract second from first predictor (helps when actually applying these) */
pred_Q13[ 0 ] -= pred_Q13[ 1 ];
}
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_int32 NLSF_Q15_tmp;
const opus_uint8 *pCB_element;
const opus_int16 *pCB_Wght_Q9;
/* Unpack entropy table indices and predictor for current CB1 index */
silk_NLSF_unpack( ec_ix, pred_Q8, psNLSF_CB, NLSFIndices[ 0 ] );
/* Predictive residual dequantizer */
silk_NLSF_residual_dequant( res_Q10, &NLSFIndices[ 1 ], pred_Q8, psNLSF_CB->quantStepSize_Q16, psNLSF_CB->order );
/* Apply inverse square-rooted weights to first stage and add to output */
pCB_element = &psNLSF_CB->CB1_NLSF_Q8[ NLSFIndices[ 0 ] * psNLSF_CB->order ];
pCB_Wght_Q9 = &psNLSF_CB->CB1_Wght_Q9[ NLSFIndices[ 0 ] * psNLSF_CB->order ];
for( i = 0; i < psNLSF_CB->order; i++ ) {
NLSF_Q15_tmp = silk_ADD_LSHIFT32( silk_DIV32_16( silk_LSHIFT( (opus_int32)res_Q10[ i ], 14 ), pCB_Wght_Q9[ i ] ), (opus_int16)pCB_element[ i ], 7 );
pNLSF_Q15[ i ] = (opus_int16)silk_LIMIT( NLSF_Q15_tmp, 0, 32767 );
}
/* NLSF stabilization */
silk_NLSF_stabilize( pNLSF_Q15, psNLSF_CB->deltaMin_Q15, psNLSF_CB->order );
}
opus_int silk_VAD_Init( /* O Return value, 0 if success */
silk_VAD_state *psSilk_VAD /* I/O Pointer to Silk VAD state */
)
{
opus_int b, ret = 0;
/* reset state memory */
silk_memset( psSilk_VAD, 0, sizeof( silk_VAD_state ) );
/* init noise levels */
/* Initialize array with approx pink noise levels (psd proportional to inverse of frequency) */
for( b = 0; b < VAD_N_BANDS; b++ ) {
psSilk_VAD->NoiseLevelBias[ b ] = silk_max_32( silk_DIV32_16( VAD_NOISE_LEVELS_BIAS, b + 1 ), 1 );
}
/* Initialize state */
for( b = 0; b < VAD_N_BANDS; b++ ) {
psSilk_VAD->NL[ b ] = silk_MUL( 100, psSilk_VAD->NoiseLevelBias[ b ] );
psSilk_VAD->inv_NL[ b ] = silk_DIV32( silk_int32_MAX, psSilk_VAD->NL[ b ] );
}
psSilk_VAD->counter = 15;
/* init smoothed energy-to-noise ratio*/
for( b = 0; b < VAD_N_BANDS; b++ ) {
psSilk_VAD->NrgRatioSmth_Q8[ b ] = 100 * 256; /* 100 * 256 --> 20 dB SNR */
}
return( ret );
}
/* High-pass filter with cutoff frequency adaptation based on pitch lag statistics */
void silk_HP_variable_cutoff(silk_encoder_state_Fxx state_Fxx[] /* I/O Encoder states */
)
{
int quality_Q15;
int32_t pitch_freq_Hz_Q16, pitch_freq_log_Q7, delta_freq_Q7;
silk_encoder_state *psEncC1 = &state_Fxx[0].sCmn;
/* Adaptive cutoff frequency: estimate low end of pitch frequency range */
if (psEncC1->prevSignalType == TYPE_VOICED) {
/* difference, in log domain */
pitch_freq_Hz_Q16 =
silk_DIV32_16(silk_LSHIFT
(silk_MUL(psEncC1->fs_kHz, 1000), 16),
psEncC1->prevLag);
pitch_freq_log_Q7 = silk_lin2log(pitch_freq_Hz_Q16) - (16 << 7);
/* adjustment based on quality */
quality_Q15 = psEncC1->input_quality_bands_Q15[0];
pitch_freq_log_Q7 =
silk_SMLAWB(pitch_freq_log_Q7,
silk_SMULWB(silk_LSHIFT(-quality_Q15, 2),
quality_Q15),
pitch_freq_log_Q7 -
(silk_lin2log
(SILK_FIX_CONST(VARIABLE_HP_MIN_CUTOFF_HZ, 16))
- (16 << 7)));
/* delta_freq = pitch_freq_log - psEnc->variable_HP_smth1; */
delta_freq_Q7 =
pitch_freq_log_Q7 -
silk_RSHIFT(psEncC1->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 = silk_MUL(delta_freq_Q7, 3);
}
/* limit delta, to reduce impact of outliers in pitch estimation */
delta_freq_Q7 =
silk_LIMIT_32(delta_freq_Q7,
-SILK_FIX_CONST(VARIABLE_HP_MAX_DELTA_FREQ,
7),
SILK_FIX_CONST(VARIABLE_HP_MAX_DELTA_FREQ,
7));
/* update smoother */
psEncC1->variable_HP_smth1_Q15 =
silk_SMLAWB(psEncC1->variable_HP_smth1_Q15,
silk_SMULBB(psEncC1->speech_activity_Q8,
delta_freq_Q7),
SILK_FIX_CONST(VARIABLE_HP_SMTH_COEF1, 16));
/* limit frequency range */
psEncC1->variable_HP_smth1_Q15 =
silk_LIMIT_32(psEncC1->variable_HP_smth1_Q15,
silk_LSHIFT(silk_lin2log
(VARIABLE_HP_MIN_CUTOFF_HZ), 8),
silk_LSHIFT(silk_lin2log
(VARIABLE_HP_MAX_CUTOFF_HZ), 8));
}
}
/* Glues concealed frames with new good received frames */
void silk_PLC_glue_frames(
silk_decoder_state *psDec, /* I/O decoder state */
opus_int16 frame[], /* I/O signal */
opus_int length /* I length of signal */
)
{
opus_int i, energy_shift;
opus_int32 energy;
silk_PLC_struct *psPLC;
psPLC = &psDec->sPLC;
if( psDec->lossCnt ) {
/* Calculate energy in concealed residual */
silk_sum_sqr_shift( &psPLC->conc_energy, &psPLC->conc_energy_shift, frame, length );
psPLC->last_frame_lost = 1;
} else {
if( psDec->sPLC.last_frame_lost ) {
/* Calculate residual in decoded signal if last frame was lost */
silk_sum_sqr_shift( &energy, &energy_shift, frame, length );
/* Normalize energies */
if( energy_shift > psPLC->conc_energy_shift ) {
psPLC->conc_energy = silk_RSHIFT( psPLC->conc_energy, energy_shift - psPLC->conc_energy_shift );
} else if( energy_shift < psPLC->conc_energy_shift ) {
energy = silk_RSHIFT( energy, psPLC->conc_energy_shift - energy_shift );
}
/* Fade in the energy difference */
if( energy > psPLC->conc_energy ) {
opus_int32 frac_Q24, LZ;
opus_int32 gain_Q16, slope_Q16;
LZ = silk_CLZ32( psPLC->conc_energy );
LZ = LZ - 1;
psPLC->conc_energy = silk_LSHIFT( psPLC->conc_energy, LZ );
energy = silk_RSHIFT( energy, silk_max_32( 24 - LZ, 0 ) );
frac_Q24 = silk_DIV32( psPLC->conc_energy, silk_max( energy, 1 ) );
gain_Q16 = silk_LSHIFT( silk_SQRT_APPROX( frac_Q24 ), 4 );
slope_Q16 = silk_DIV32_16( ( (opus_int32)1 << 16 ) - gain_Q16, length );
/* Make slope 4x steeper to avoid missing onsets after DTX */
slope_Q16 = silk_LSHIFT( slope_Q16, 2 );
for( i = 0; i < length; i++ ) {
frame[ i ] = silk_SMULWB( gain_Q16, frame[ i ] );
gain_Q16 += slope_Q16;
if( gain_Q16 > (opus_int32)1 << 16 ) {
break;
}
}
}
}
psPLC->last_frame_lost = 0;
}
}
/* uses SMLAWB(), requiring armv5E and higher. */
opus_int32 silk_schur( /* O Returns residual energy */
opus_int16 *rc_Q15, /* O reflection coefficients [order] Q15 */
const opus_int32 *c, /* I correlations [order+1] */
const opus_int32 order /* I prediction order */
)
{
opus_int k, n, lz;
opus_int32 C[ SILK_MAX_ORDER_LPC + 1 ][ 2 ];
opus_int32 Ctmp1, Ctmp2, rc_tmp_Q15;
silk_assert( order==6||order==8||order==10||order==12||order==14||order==16 );
/* Get number of leading zeros */
lz = 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 ] = silk_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 ] = silk_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 = -silk_DIV32_16( C[ k + 1 ][ 0 ], silk_max_32( silk_RSHIFT( C[ 0 ][ 1 ], 15 ), 1 ) );
/* Clip (shouldn't happen for properly conditioned inputs) */
rc_tmp_Q15 = silk_SAT16( rc_tmp_Q15 );
/* Store */
rc_Q15[ k ] = (opus_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 ] = silk_SMLAWB( Ctmp1, silk_LSHIFT( Ctmp2, 1 ), rc_tmp_Q15 );
C[ n ][ 1 ] = silk_SMLAWB( Ctmp2, silk_LSHIFT( Ctmp1, 1 ), rc_tmp_Q15 );
}
}
/* return residual energy */
return C[ 0 ][ 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, 1 );
} 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 ];
silk_resampler_state_struct temp_resampler_state;
#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 = silk_LSHIFT( psEnc->sCmn.frame_length, 1 ) + LA_SHAPE_MS * psEnc->sCmn.fs_kHz;
#ifndef FIXED_POINT
silk_float2short_array( x_bufFIX, psEnc->x_buf, nSamples_temp );
#endif
/* Initialize resampler for temporary resampling of x_buf data to API_fs_Hz */
ret += silk_resampler_init( &temp_resampler_state, silk_SMULBB( psEnc->sCmn.fs_kHz, 1000 ), psEnc->sCmn.API_fs_Hz, 0 );
/* 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 = silk_DIV32_16( nSamples_temp * psEnc->sCmn.API_fs_Hz, silk_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, silk_SMULBB( fs_kHz, 1000 ), 1 );
/* Correct resampler state 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
silk_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;
}
/* Laroia low complexity NLSF weights */
void silk_NLSF_VQ_weights_laroia(
opus_int16 *pNLSFW_Q_OUT, /* O Pointer to input vector weights [D] */
const opus_int16 *pNLSF_Q15, /* I Pointer to input vector [D] */
const opus_int D /* I Input vector dimension (even) */
)
{
opus_int k;
opus_int32 tmp1_int, tmp2_int;
silk_assert( D > 0 );
silk_assert( ( D & 1 ) == 0 );
/* First value */
tmp1_int = silk_max_int( pNLSF_Q15[ 0 ], 1 );
tmp1_int = silk_DIV32_16( 1 << ( 15 + NLSF_W_Q ), tmp1_int );
tmp2_int = silk_max_int( pNLSF_Q15[ 1 ] - pNLSF_Q15[ 0 ], 1 );
tmp2_int = silk_DIV32_16( 1 << ( 15 + NLSF_W_Q ), tmp2_int );
pNLSFW_Q_OUT[ 0 ] = (opus_int16)silk_min_int( tmp1_int + tmp2_int, silk_int16_MAX );
silk_assert( pNLSFW_Q_OUT[ 0 ] > 0 );
/* Main loop */
for( k = 1; k < D - 1; k += 2 ) {
tmp1_int = silk_max_int( pNLSF_Q15[ k + 1 ] - pNLSF_Q15[ k ], 1 );
tmp1_int = silk_DIV32_16( 1 << ( 15 + NLSF_W_Q ), tmp1_int );
pNLSFW_Q_OUT[ k ] = (opus_int16)silk_min_int( tmp1_int + tmp2_int, silk_int16_MAX );
silk_assert( pNLSFW_Q_OUT[ k ] > 0 );
tmp2_int = silk_max_int( pNLSF_Q15[ k + 2 ] - pNLSF_Q15[ k + 1 ], 1 );
tmp2_int = silk_DIV32_16( 1 << ( 15 + NLSF_W_Q ), tmp2_int );
pNLSFW_Q_OUT[ k + 1 ] = (opus_int16)silk_min_int( tmp1_int + tmp2_int, silk_int16_MAX );
silk_assert( pNLSFW_Q_OUT[ k + 1 ] > 0 );
}
/* Last value */
tmp1_int = silk_max_int( ( 1 << 15 ) - pNLSF_Q15[ D - 1 ], 1 );
tmp1_int = silk_DIV32_16( 1 << ( 15 + NLSF_W_Q ), tmp1_int );
pNLSFW_Q_OUT[ D - 1 ] = (opus_int16)silk_min_int( tmp1_int + tmp2_int, silk_int16_MAX );
silk_assert( pNLSFW_Q_OUT[ 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 */
silk_memcpy( x1, state->sMid, 2 * sizeof( opus_int16 ) );
silk_memcpy( x2, state->sSide, 2 * sizeof( opus_int16 ) );
silk_memcpy( state->sMid, &x1[ frame_length ], 2 * sizeof( opus_int16 ) );
silk_memcpy( state->sSide, &x2[ frame_length ], 2 * sizeof( opus_int16 ) );
/* Interpolate predictors and add prediction to side channel */
pred0_Q13 = state->pred_prev_Q13[ 0 ];
pred1_Q13 = state->pred_prev_Q13[ 1 ];
denom_Q16 = silk_DIV32_16( (opus_int32)1 << 16, STEREO_INTERP_LEN_MS * fs_kHz );
delta0_Q13 = silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 0 ] - state->pred_prev_Q13[ 0 ], denom_Q16 ), 16 );
delta1_Q13 = silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 1 ] - state->pred_prev_Q13[ 1 ], denom_Q16 ), 16 );
for( n = 0; n < STEREO_INTERP_LEN_MS * fs_kHz; n++ ) {
pred0_Q13 += delta0_Q13;
pred1_Q13 += delta1_Q13;
sum = silk_LSHIFT( silk_ADD_LSHIFT( x1[ n ] + x1[ n + 2 ], x1[ n + 1 ], 1 ), 9 ); /* Q11 */
sum = silk_SMLAWB( silk_LSHIFT( (opus_int32)x2[ n + 1 ], 8 ), sum, pred0_Q13 ); /* Q8 */
sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)x1[ n + 1 ], 11 ), pred1_Q13 ); /* Q8 */
x2[ n + 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) );
}
pred0_Q13 = pred_Q13[ 0 ];
pred1_Q13 = pred_Q13[ 1 ];
for( n = STEREO_INTERP_LEN_MS * fs_kHz; n < frame_length; n++ ) {
sum = silk_LSHIFT( silk_ADD_LSHIFT( x1[ n ] + x1[ n + 2 ], x1[ n + 1 ], 1 ), 9 ); /* Q11 */
sum = silk_SMLAWB( silk_LSHIFT( (opus_int32)x2[ n + 1 ], 8 ), sum, pred0_Q13 ); /* Q8 */
sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)x1[ n + 1 ], 11 ), pred1_Q13 ); /* Q8 */
x2[ n + 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) );
}
state->pred_prev_Q13[ 0 ] = pred_Q13[ 0 ];
state->pred_prev_Q13[ 1 ] = pred_Q13[ 1 ];
/* Convert to left/right signals */
for( n = 0; n < frame_length; n++ ) {
sum = x1[ n + 1 ] + (opus_int32)x2[ n + 1 ];
diff = x1[ n + 1 ] - (opus_int32)x2[ n + 1 ];
x1[ n + 1 ] = (opus_int16)silk_SAT16( sum );
x2[ n + 1 ] = (opus_int16)silk_SAT16( diff );
}
}
/* Processing of gains */
void silk_process_gains_FIX(
silk_encoder_state_FIX *psEnc, /* I/O Encoder state */
silk_encoder_control_FIX *psEncCtrl, /* I/O Encoder control */
opus_int condCoding /* I The type of conditional coding to use */
)
{
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 * silk_sigmoid( 0.25f * ( psEncCtrl->LTPredCodGain - 12.0f ) ); */
s_Q16 = -silk_sigm_Q15( silk_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 ] = silk_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 = silk_DIV32_16( silk_log2lin(
silk_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 = silk_SMULWW( ResNrg, InvMaxSqrVal_Q16 );
if( psEncCtrl->ResNrgQ[ k ] > 0 ) {
ResNrgPart = silk_RSHIFT_ROUND( ResNrgPart, psEncCtrl->ResNrgQ[ k ] );
} else {
if( ResNrgPart >= silk_RSHIFT( silk_int32_MAX, -psEncCtrl->ResNrgQ[ k ] ) ) {
ResNrgPart = silk_int32_MAX;
} else {
ResNrgPart = silk_LSHIFT( ResNrgPart, -psEncCtrl->ResNrgQ[ k ] );
}
}
gain = psEncCtrl->Gains_Q16[ k ];
gain_squared = silk_ADD_SAT32( ResNrgPart, silk_SMMUL( gain, gain ) );
if( gain_squared < silk_int16_MAX ) {
/* recalculate with higher precision */
gain_squared = silk_SMLAWW( silk_LSHIFT( ResNrgPart, 16 ), gain, gain );
silk_assert( gain_squared > 0 );
gain = silk_SQRT_APPROX( gain_squared ); /* Q8 */
gain = silk_min( gain, silk_int32_MAX >> 8 );
psEncCtrl->Gains_Q16[ k ] = silk_LSHIFT_SAT32( gain, 8 ); /* Q16 */
} else {
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 ] = silk_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 ] );
/* Predictive residual 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( silk_LSHIFT( (opus_int32)W_tmp_QW[ i ], 18 - NLSF_W_Q ) );
NLSF_Q15_tmp = silk_ADD32( pNLSF_Q15[ i ], silk_DIV32_16( silk_LSHIFT( (opus_int32)res_Q10[ i ], 14 ), W_tmp_Q9 ) );
pNLSF_Q15[ i ] = (opus_int16)silk_LIMIT( NLSF_Q15_tmp, 0, 32767 );
}
/* NLSF stabilization */
silk_NLSF_stabilize( pNLSF_Q15, psNLSF_CB->deltaMin_Q15, psNLSF_CB->order );
}
/* Quantize mid/side predictors */
void silk_stereo_quant_pred(
opus_int32 pred_Q13[], /* I/O Predictors (out: quantized) */
opus_int8 ix[ 2 ][ 3 ] /* O Quantization indices */
)
{
opus_int i, j, n;
opus_int32 low_Q13, step_Q13, lvl_Q13, err_min_Q13, err_Q13, quant_pred_Q13 = 0;
/* Quantize */
for( n = 0; n < 2; n++ ) {
/* Brute-force search over quantization levels */
err_min_Q13 = silk_int32_MAX;
for( i = 0; i < STEREO_QUANT_TAB_SIZE - 1; i++ ) {
low_Q13 = silk_stereo_pred_quant_Q13[ i ];
step_Q13 = silk_SMULWB( silk_stereo_pred_quant_Q13[ i + 1 ] - low_Q13,
SILK_FIX_CONST( 0.5 / STEREO_QUANT_SUB_STEPS, 16 ) );
for( j = 0; j < STEREO_QUANT_SUB_STEPS; j++ ) {
lvl_Q13 = silk_SMLABB( low_Q13, step_Q13, 2 * j + 1 );
err_Q13 = silk_abs( pred_Q13[ n ] - lvl_Q13 );
if( err_Q13 < err_min_Q13 ) {
err_min_Q13 = err_Q13;
quant_pred_Q13 = lvl_Q13;
ix[ n ][ 0 ] = i;
ix[ n ][ 1 ] = j;
} else {
/* Error increasing, so we're past the optimum */
goto done;
}
}
}
done:
ix[ n ][ 2 ] = silk_DIV32_16( ix[ n ][ 0 ], 3 );
ix[ n ][ 0 ] -= ix[ n ][ 2 ] * 3;
pred_Q13[ n ] = quant_pred_Q13;
}
/* Subtract second from first predictor (helps when actually applying these) */
pred_Q13[ 0 ] -= pred_Q13[ 1 ];
}
/* Deactivate by setting psEncC->mode = 0; */
void silk_LP_variable_cutoff(
silk_LP_state *psLP, /* I/O LP filter state */
opus_int16 *frame, /* I/O Low-pass filtered output signal */
const opus_int frame_length /* I Frame length */
)
{
opus_int32 B_Q28[ TRANSITION_NB ], A_Q28[ TRANSITION_NA ], fac_Q16 = 0;
opus_int ind = 0;
silk_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 = silk_LSHIFT( TRANSITION_FRAMES - psLP->transition_frame_no, 16 - 6 );
#else
fac_Q16 = silk_DIV32_16( silk_LSHIFT( TRANSITION_FRAMES - psLP->transition_frame_no, 16 ), TRANSITION_FRAMES );
#endif
ind = silk_RSHIFT( fac_Q16, 16 );
fac_Q16 -= silk_LSHIFT( ind, 16 );
silk_assert( ind >= 0 );
silk_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 = silk_LIMIT( psLP->transition_frame_no + psLP->mode, 0, TRANSITION_FRAMES );
/* ARMA low-pass filtering */
silk_assert( TRANSITION_NB == 3 && TRANSITION_NA == 2 );
silk_biquad_alt( frame, B_Q28, A_Q28, psLP->In_LP_State, frame, frame_length, 1);
}
}
opus_int silk_VAD_GetSA_Q8( /* O Return value, 0 if success */
silk_encoder_state *psEncC, /* I/O Encoder state */
const opus_int16 pIn[] /* I PCM input */
)
{
opus_int SA_Q15, pSNR_dB_Q7, input_tilt;
opus_int decimated_framelength1, decimated_framelength2;
opus_int decimated_framelength;
opus_int dec_subframe_length, dec_subframe_offset, SNR_Q7, i, b, s;
opus_int32 sumSquared, smooth_coef_Q16;
opus_int16 HPstateTmp;
VARDECL( opus_int16, X );
opus_int32 Xnrg[ VAD_N_BANDS ];
opus_int32 NrgToNoiseRatio_Q8[ VAD_N_BANDS ];
opus_int32 speech_nrg, x_tmp;
opus_int X_offset[ VAD_N_BANDS ];
opus_int ret = 0;
silk_VAD_state *psSilk_VAD = &psEncC->sVAD;
SAVE_STACK;
/* Safety checks */
silk_assert( VAD_N_BANDS == 4 );
silk_assert( MAX_FRAME_LENGTH >= psEncC->frame_length );
silk_assert( psEncC->frame_length <= 512 );
silk_assert( psEncC->frame_length == 8 * silk_RSHIFT( psEncC->frame_length, 3 ) );
/***********************/
/* Filter and Decimate */
/***********************/
decimated_framelength1 = silk_RSHIFT( psEncC->frame_length, 1 );
decimated_framelength2 = silk_RSHIFT( psEncC->frame_length, 2 );
decimated_framelength = silk_RSHIFT( psEncC->frame_length, 3 );
/* Decimate into 4 bands:
0 L 3L L 3L 5L
- -- - -- --
8 8 2 4 4
[0-1 kHz| temp. |1-2 kHz| 2-4 kHz | 4-8 kHz |
They're arranged to allow the minimal ( frame_length / 4 ) extra
scratch space during the downsampling process */
X_offset[ 0 ] = 0;
X_offset[ 1 ] = decimated_framelength + decimated_framelength2;
X_offset[ 2 ] = X_offset[ 1 ] + decimated_framelength;
X_offset[ 3 ] = X_offset[ 2 ] + decimated_framelength2;
ALLOC( X, X_offset[ 3 ] + decimated_framelength1, opus_int16 );
/* 0-8 kHz to 0-4 kHz and 4-8 kHz */
silk_ana_filt_bank_1( pIn, &psSilk_VAD->AnaState[ 0 ],
X, &X[ X_offset[ 3 ] ], psEncC->frame_length );
/* 0-4 kHz to 0-2 kHz and 2-4 kHz */
silk_ana_filt_bank_1( X, &psSilk_VAD->AnaState1[ 0 ],
X, &X[ X_offset[ 2 ] ], decimated_framelength1 );
/* 0-2 kHz to 0-1 kHz and 1-2 kHz */
silk_ana_filt_bank_1( X, &psSilk_VAD->AnaState2[ 0 ],
X, &X[ X_offset[ 1 ] ], decimated_framelength2 );
/*********************************************/
/* HP filter on lowest band (differentiator) */
/*********************************************/
X[ decimated_framelength - 1 ] = silk_RSHIFT( X[ decimated_framelength - 1 ], 1 );
HPstateTmp = X[ decimated_framelength - 1 ];
for( i = decimated_framelength - 1; i > 0; i-- ) {
X[ i - 1 ] = silk_RSHIFT( X[ i - 1 ], 1 );
X[ i ] -= X[ i - 1 ];
}
X[ 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 = silk_RSHIFT( psEncC->frame_length, silk_min_int( VAD_N_BANDS - b, VAD_N_BANDS - 1 ) );
/* Split length into subframe lengths */
dec_subframe_length = silk_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 * ( silk_int16_MIN / 8 ) ^ 2. */
/* Therefore we can accumulate with no risk of overflow (unless dec_subframe_length > 128) */
x_tmp = silk_RSHIFT(
X[ X_offset[ b ] + i + dec_subframe_offset ], 3 );
sumSquared = silk_SMLABB( sumSquared, x_tmp, x_tmp );
/* Safety check */
silk_assert( sumSquared >= 0 );
}
/* Add/saturate summed energy of current subframe */
if( s < VAD_INTERNAL_SUBFRAMES - 1 ) {
Xnrg[ b ] = silk_ADD_POS_SAT32( Xnrg[ b ], sumSquared );
} else {
/* Look-ahead subframe */
Xnrg[ b ] = silk_ADD_POS_SAT32( Xnrg[ b ], silk_RSHIFT( sumSquared, 1 ) );
}
dec_subframe_offset += dec_subframe_length;
}
psSilk_VAD->XnrgSubfr[ b ] = sumSquared;
}
/********************/
/* Noise estimation */
/********************/
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 ] = silk_DIV32( silk_LSHIFT( Xnrg[ b ], 8 ), psSilk_VAD->NL[ b ] + 1 );
} else {
NrgToNoiseRatio_Q8[ b ] = silk_DIV32( Xnrg[ b ], silk_RSHIFT( psSilk_VAD->NL[ b ], 8 ) + 1 );
}
/* Convert to log domain */
SNR_Q7 = silk_lin2log( NrgToNoiseRatio_Q8[ b ] ) - 8 * 128;
/* Sum-of-squares */
sumSquared = silk_SMLABB( sumSquared, SNR_Q7, SNR_Q7 ); /* Q14 */
/* Tilt measure */
if( speech_nrg < ( (opus_int32)1 << 20 ) ) {
/* Scale down SNR value for small subband speech energies */
SNR_Q7 = silk_SMULWB( silk_LSHIFT( silk_SQRT_APPROX( speech_nrg ), 6 ), SNR_Q7 );
}
input_tilt = silk_SMLAWB( input_tilt, tiltWeights[ b ], SNR_Q7 );
} else {
NrgToNoiseRatio_Q8[ b ] = 256;
}
}
/* Mean-of-squares */
sumSquared = silk_DIV32_16( sumSquared, VAD_N_BANDS ); /* Q14 */
/* Root-mean-square approximation, scale to dBs, and write to output pointer */
pSNR_dB_Q7 = (opus_int16)( 3 * silk_SQRT_APPROX( sumSquared ) ); /* Q7 */
/*********************************/
/* Speech Probability Estimation */
/*********************************/
SA_Q15 = silk_sigm_Q15( silk_SMULWB( VAD_SNR_FACTOR_Q16, pSNR_dB_Q7 ) - VAD_NEGATIVE_OFFSET_Q5 );
/**************************/
/* Frequency Tilt Measure */
/**************************/
psEncC->input_tilt_Q15 = silk_LSHIFT( 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 ) * silk_RSHIFT( Xnrg[ b ] - psSilk_VAD->NL[ b ], 4 );
}
/* Power scaling */
if( speech_nrg <= 0 ) {
SA_Q15 = silk_RSHIFT( SA_Q15, 1 );
} else if( speech_nrg < 32768 ) {
if( psEncC->frame_length == 10 * psEncC->fs_kHz ) {
speech_nrg = silk_LSHIFT_SAT32( speech_nrg, 16 );
} else {
speech_nrg = silk_LSHIFT_SAT32( speech_nrg, 15 );
}
/* square-root */
speech_nrg = silk_SQRT_APPROX( speech_nrg );
SA_Q15 = silk_SMULWB( 32768 + speech_nrg, SA_Q15 );
}
/* Copy the resulting speech activity in Q8 */
psEncC->speech_activity_Q8 = silk_min_int( silk_RSHIFT( SA_Q15, 7 ), silk_uint8_MAX );
/***********************************/
/* Energy Level and SNR estimation */
/***********************************/
/* Smoothing coefficient */
smooth_coef_Q16 = silk_SMULWB( VAD_SNR_SMOOTH_COEF_Q18, silk_SMULWB( (opus_int32)SA_Q15, SA_Q15 ) );
if( psEncC->frame_length == 10 * psEncC->fs_kHz ) {
smooth_coef_Q16 >>= 1;
}
/* Initialize/reset the resampler state for a given pair of input/output sampling rates */
opus_int silk_resampler_init(
silk_resampler_state_struct *S, /* I/O Resampler state */
opus_int32 Fs_Hz_in, /* I Input sampling rate (Hz) */
opus_int32 Fs_Hz_out /* I Output sampling rate (Hz) */
)
{
opus_int32 up2 = 0, down2 = 0;
/* Clear state */
silk_memset( S, 0, sizeof( silk_resampler_state_struct ) );
/* Input checking */
if( ( Fs_Hz_in != 8000 && Fs_Hz_in != 12000 && Fs_Hz_in != 16000 && Fs_Hz_in != 24000 && Fs_Hz_in != 48000 ) ||
( Fs_Hz_out != 8000 && Fs_Hz_out != 12000 && Fs_Hz_out != 16000 && Fs_Hz_out != 24000 && Fs_Hz_out != 48000 ) ) {
silk_assert( 0 );
return -1;
}
/* Number of samples processed per batch */
S->batchSize = silk_DIV32_16( Fs_Hz_in, 100 );
/* Find resampler with the right sampling ratio */
if( Fs_Hz_out > Fs_Hz_in ) {
/* Upsample */
if( Fs_Hz_out == silk_MUL( Fs_Hz_in, 2 ) ) { /* Fs_out : Fs_in = 2 : 1 */
/* Special case: directly use 2x upsampler */
S->resampler_function = USE_silk_resampler_private_up2_HQ_wrapper;
} else {
/* Default resampler */
S->resampler_function = USE_silk_resampler_private_IIR_FIR;
up2 = 1;
}
} else if ( Fs_Hz_out < Fs_Hz_in ) {
/* Downsample */
if( silk_MUL( Fs_Hz_out, 4 ) == silk_MUL( Fs_Hz_in, 3 ) ) { /* Fs_out : Fs_in = 3 : 4 */
S->FIR_Fracs = 3;
S->Coefs = silk_Resampler_3_4_COEFS;
S->resampler_function = USE_silk_resampler_private_down_FIR;
} else if( silk_MUL( Fs_Hz_out, 3 ) == silk_MUL( Fs_Hz_in, 2 ) ) { /* Fs_out : Fs_in = 2 : 3 */
S->FIR_Fracs = 2;
S->Coefs = silk_Resampler_2_3_COEFS;
S->resampler_function = USE_silk_resampler_private_down_FIR;
} else if( silk_MUL( Fs_Hz_out, 2 ) == Fs_Hz_in ) { /* Fs_out : Fs_in = 1 : 2 */
S->FIR_Fracs = 1;
S->Coefs = silk_Resampler_1_2_COEFS;
S->resampler_function = USE_silk_resampler_private_down_FIR;
} else if( silk_MUL( Fs_Hz_out, 3 ) == Fs_Hz_in ) { /* Fs_out : Fs_in = 1 : 3 */
S->FIR_Fracs = 1;
S->Coefs = silk_Resampler_1_3_COEFS;
S->resampler_function = USE_silk_resampler_private_down_FIR;
} else if( silk_MUL( Fs_Hz_out, 4 ) == Fs_Hz_in ) { /* Fs_out : Fs_in = 1 : 4 */
S->FIR_Fracs = 1;
down2 = 1;
S->Coefs = silk_Resampler_1_2_COEFS;
S->resampler_function = USE_silk_resampler_private_down_FIR;
} else if( silk_MUL( Fs_Hz_out, 6 ) == Fs_Hz_in ) { /* Fs_out : Fs_in = 1 : 6 */
S->FIR_Fracs = 1;
down2 = 1;
S->Coefs = silk_Resampler_1_3_COEFS;
S->resampler_function = USE_silk_resampler_private_down_FIR;
} else {
/* None available */
silk_assert( 0 );
return -1;
}
} else {
/* Input and output sampling rates are equal: copy */
S->resampler_function = USE_silk_resampler_copy;
}
S->input2x = up2 | down2;
/* Ratio of input/output samples */
S->invRatio_Q16 = silk_LSHIFT32( silk_DIV32( silk_LSHIFT32( Fs_Hz_in, 14 + up2 - down2 ), Fs_Hz_out ), 2 );
/* Make sure the ratio is rounded up */
while( silk_SMULWW( S->invRatio_Q16, silk_LSHIFT32( Fs_Hz_out, down2 ) ) < silk_LSHIFT32( Fs_Hz_in, up2 ) ) {
S->invRatio_Q16++;
}
return 0;
}
/* 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 condCoding /* I The type of conditional coding to use */
)
{
opus_int i, k, 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( encode_LBRR ) {
psIndices = &psEncC->indices_LBRR[ FrameIndex ];
} else {
psIndices = &psEncC->indices;
}
/*******************************************/
/* Encode signal type and quantizer offset */
/*******************************************/
typeOffset = 2 * psIndices->signalType + psIndices->quantOffsetType;
silk_assert( typeOffset >= 0 && typeOffset < 6 );
silk_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 */
/****************/
/* first subframe */
if( condCoding == CODE_CONDITIONALLY ) {
/* conditional coding */
silk_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 */
silk_assert( psIndices->GainsIndices[ 0 ] >= 0 && psIndices->GainsIndices[ 0 ] < N_LEVELS_QGAIN );
ec_enc_icdf( psRangeEnc, silk_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++ ) {
silk_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 );
}
/****************/
/* Encode NLSFs */
/****************/
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 ] );
silk_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 ) {
silk_assert( psIndices->NLSFInterpCoef_Q2 >= 0 && psIndices->NLSFInterpCoef_Q2 < 5 );
ec_enc_icdf( psRangeEnc, psIndices->NLSFInterpCoef_Q2, silk_NLSF_interpolation_factor_iCDF, 8 );
}
if( psIndices->signalType == TYPE_VOICED )
{
/*********************/
/* Encode pitch lags */
/*********************/
/* lag index */
encode_absolute_lagIndex = 1;
if( condCoding == CODE_CONDITIONALLY && 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 */
}
silk_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 = silk_DIV32_16( psIndices->lagIndex, silk_RSHIFT( psEncC->fs_kHz, 1 ) );
pitch_low_bits = psIndices->lagIndex - silk_SMULBB( pitch_high_bits, silk_RSHIFT( psEncC->fs_kHz, 1 ) );
silk_assert( pitch_low_bits < psEncC->fs_kHz / 2 );
silk_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;
/* Countour index */
silk_assert( psIndices->contourIndex >= 0 );
silk_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 );
/********************/
/* Encode LTP gains */
/********************/
/* PERIndex value */
silk_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++ ) {
silk_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 == CODE_INDEPENDENTLY ) {
silk_assert( psIndices->LTP_scaleIndex >= 0 && psIndices->LTP_scaleIndex < 3 );
ec_enc_icdf( psRangeEnc, psIndices->LTP_scaleIndex, silk_LTPscale_iCDF, 8 );
}
silk_assert( !condCoding || psIndices->LTP_scaleIndex == 0 );
}
psEncC->ec_prevSignalType = psIndices->signalType;
/***************/
/* Encode seed */
/***************/
silk_assert( psIndices->Seed >= 0 && psIndices->Seed < 4 );
ec_enc_icdf( psRangeEnc, psIndices->Seed, silk_uniform4_iCDF, 8 );
}
/* Control internal sampling rate */
int silk_control_audio_bandwidth(silk_encoder_state * psEncC, /* I/O Pointer to Silk encoder state */
silk_EncControlStruct * encControl /* I Control structure */
)
{
int fs_kHz;
int32_t fs_Hz;
fs_kHz = psEncC->fs_kHz;
fs_Hz = silk_SMULBB(fs_kHz, 1000);
if (fs_Hz == 0) {
/* Encoder has just been initialized */
fs_Hz =
silk_min(psEncC->desiredInternal_fs_Hz, psEncC->API_fs_Hz);
fs_kHz = silk_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 = silk_min(fs_Hz, psEncC->maxInternal_fs_Hz);
fs_Hz = silk_max(fs_Hz, psEncC->minInternal_fs_Hz);
fs_kHz = silk_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 || encControl->opusCanSwitch) {
/* Check if we should switch down */
if (silk_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 */
memzero(psEncC->sLP.In_LP_State,
sizeof(psEncC->sLP.In_LP_State));
}
if (encControl->opusCanSwitch) {
/* Stop transition phase */
psEncC->sLP.mode = 0;
/* Switch to a lower sample frequency */
fs_kHz = psEncC->fs_kHz == 16 ? 12 : 8;
} else {
if (psEncC->sLP.transition_frame_no <=
0) {
encControl->switchReady = 1;
/* Make room for redundancy */
encControl->maxBits -=
encControl->maxBits * 5 /
(encControl->
payloadSize_ms + 5);
} else {
/* Direction: down (at double speed) */
psEncC->sLP.mode = -2;
}
}
} else
/* Check if we should switch up */
if (silk_SMULBB(psEncC->fs_kHz, 1000) <
psEncC->desiredInternal_fs_Hz) {
/* Switch up */
if (encControl->opusCanSwitch) {
/* 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 */
memzero(psEncC->sLP.In_LP_State,
sizeof(psEncC->sLP.In_LP_State));
/* Direction: up */
psEncC->sLP.mode = 1;
} else {
if (psEncC->sLP.mode == 0) {
encControl->switchReady = 1;
/* Make room for redundancy */
encControl->maxBits -=
encControl->maxBits * 5 /
(encControl->
payloadSize_ms + 5);
} else {
/* Direction: up */
psEncC->sLP.mode = 1;
}
}
} else {
if (psEncC->sLP.mode < 0)
psEncC->sLP.mode = 1;
}
}
}
return fs_kHz;
}
/* If not all roots are found, the a_Q16 coefficients are bandwidth expanded until convergence. */
void silk_A2NLSF(
opus_int16 *NLSF, /* O Normalized Line Spectral Frequencies in Q15 (0..2^15-1) [d] */
opus_int32 *a_Q16, /* I/O Monic whitening filter coefficients in Q16 [d] */
const opus_int d /* I Filter order (must be even) */
)
{
opus_int i, k, m, dd, root_ix, ffrac;
opus_int32 xlo, xhi, xmid;
opus_int32 ylo, yhi, ymid, thr;
opus_int32 nom, den;
opus_int32 P[ SILK_MAX_ORDER_LPC / 2 + 1 ];
opus_int32 Q[ SILK_MAX_ORDER_LPC / 2 + 1 ];
opus_int32 *PQ[ 2 ];
opus_int32 *p;
/* Store pointers to array */
PQ[ 0 ] = P;
PQ[ 1 ] = Q;
dd = silk_RSHIFT( d, 1 );
silk_A2NLSF_init( a_Q16, P, Q, dd );
/* Find roots, alternating between P and Q */
p = P; /* Pointer to polynomial */
xlo = silk_LSFCosTab_FIX_Q12[ 0 ]; /* Q12*/
ylo = silk_A2NLSF_eval_poly( p, xlo, dd );
if( ylo < 0 ) {
/* Set the first NLSF to zero and move on to the next */
NLSF[ 0 ] = 0;
p = Q; /* Pointer to polynomial */
ylo = silk_A2NLSF_eval_poly( p, xlo, dd );
root_ix = 1; /* Index of current root */
} else {
root_ix = 0; /* Index of current root */
}
k = 1; /* Loop counter */
i = 0; /* Counter for bandwidth expansions applied */
thr = 0;
while( 1 ) {
/* Evaluate polynomial */
xhi = silk_LSFCosTab_FIX_Q12[ k ]; /* Q12 */
yhi = silk_A2NLSF_eval_poly( p, xhi, dd );
/* Detect zero crossing */
if( ( ylo <= 0 && yhi >= thr ) || ( ylo >= 0 && yhi <= -thr ) ) {
if( yhi == 0 ) {
/* If the root lies exactly at the end of the current */
/* interval, look for the next root in the next interval */
thr = 1;
} else {
thr = 0;
}
/* Binary division */
ffrac = -256;
for( m = 0; m < BIN_DIV_STEPS_A2NLSF_FIX; m++ ) {
/* Evaluate polynomial */
xmid = silk_RSHIFT_ROUND( xlo + xhi, 1 );
ymid = silk_A2NLSF_eval_poly( p, xmid, dd );
/* Detect zero crossing */
if( ( ylo <= 0 && ymid >= 0 ) || ( ylo >= 0 && ymid <= 0 ) ) {
/* Reduce frequency */
xhi = xmid;
yhi = ymid;
} else {
/* Increase frequency */
xlo = xmid;
ylo = ymid;
ffrac = silk_ADD_RSHIFT( ffrac, 128, m );
}
}
/* Interpolate */
if( silk_abs( ylo ) < 65536 ) {
/* Avoid dividing by zero */
den = ylo - yhi;
nom = silk_LSHIFT( ylo, 8 - BIN_DIV_STEPS_A2NLSF_FIX ) + silk_RSHIFT( den, 1 );
if( den != 0 ) {
ffrac += silk_DIV32( nom, den );
}
} else {
/* No risk of dividing by zero because abs(ylo - yhi) >= abs(ylo) >= 65536 */
ffrac += silk_DIV32( ylo, silk_RSHIFT( ylo - yhi, 8 - BIN_DIV_STEPS_A2NLSF_FIX ) );
}
NLSF[ root_ix ] = (opus_int16)silk_min_32( silk_LSHIFT( (opus_int32)k, 8 ) + ffrac, silk_int16_MAX );
silk_assert( NLSF[ root_ix ] >= 0 );
root_ix++; /* Next root */
if( root_ix >= d ) {
/* Found all roots */
break;
}
/* Alternate pointer to polynomial */
p = PQ[ root_ix & 1 ];
/* Evaluate polynomial */
xlo = silk_LSFCosTab_FIX_Q12[ k - 1 ]; /* Q12*/
ylo = silk_LSHIFT( 1 - ( root_ix & 2 ), 12 );
} else {
/* Increment loop counter */
k++;
xlo = xhi;
ylo = yhi;
thr = 0;
if( k > LSF_COS_TAB_SZ_FIX ) {
i++;
if( i > MAX_ITERATIONS_A2NLSF_FIX ) {
/* Set NLSFs to white spectrum and exit */
NLSF[ 0 ] = (opus_int16)silk_DIV32_16( 1 << 15, d + 1 );
for( k = 1; k < d; k++ ) {
NLSF[ k ] = (opus_int16)silk_SMULBB( k + 1, NLSF[ 0 ] );
}
return;
}
/* Error: Apply progressively more bandwidth expansion and run again */
silk_bwexpander_32( a_Q16, d, 65536 - silk_SMULBB( 10 + i, i ) ); /* 10_Q16 = 0.00015*/
silk_A2NLSF_init( a_Q16, P, Q, dd );
p = P; /* Pointer to polynomial */
xlo = silk_LSFCosTab_FIX_Q12[ 0 ]; /* Q12*/
ylo = silk_A2NLSF_eval_poly( p, xlo, dd );
if( ylo < 0 ) {
/* Set the first NLSF to zero and move on to the next */
NLSF[ 0 ] = 0;
p = Q; /* Pointer to polynomial */
ylo = silk_A2NLSF_eval_poly( p, xlo, dd );
root_ix = 1; /* Index of current root */
} else {
root_ix = 0; /* Index of current root */
}
k = 1; /* Reset loop counter */
}
}
}
}
static opus_int silk_setup_resamplers(
silk_encoder_state_Fxx *psEnc, /* I/O */
opus_int fs_kHz /* I */
)
{
opus_int ret = SILK_NO_ERROR;
SAVE_STACK;
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, 1 );
} else {
VARDECL( opus_int16, x_buf_API_fs_Hz );
VARDECL( silk_resampler_state_struct, temp_resampler_state );
#ifdef OPUS_FIXED_POINT
opus_int16 *x_bufFIX = psEnc->x_buf;
#else
VARDECL( opus_int16, x_bufFIX );
opus_int32 new_buf_samples;
#endif
opus_int32 api_buf_samples;
opus_int32 old_buf_samples;
opus_int32 buf_length_ms;
buf_length_ms = silk_LSHIFT( psEnc->sCmn.nb_subfr * 5, 1 ) + LA_SHAPE_MS;
old_buf_samples = buf_length_ms * psEnc->sCmn.fs_kHz;
#ifndef OPUS_FIXED_POINT
new_buf_samples = buf_length_ms * fs_kHz;
ALLOC( x_bufFIX, silk_max( old_buf_samples, new_buf_samples ),
opus_int16 );
silk_float2short_array( x_bufFIX, psEnc->x_buf, old_buf_samples );
#endif
/* Initialize resampler for temporary resampling of x_buf data to API_fs_Hz */
ALLOC( temp_resampler_state, 1, silk_resampler_state_struct );
ret += silk_resampler_init( temp_resampler_state, silk_SMULBB( psEnc->sCmn.fs_kHz, 1000 ), psEnc->sCmn.API_fs_Hz, 0 );
/* Calculate number of samples to temporarily upsample */
api_buf_samples = buf_length_ms * silk_DIV32_16( psEnc->sCmn.API_fs_Hz, 1000 );
/* Temporary resampling of x_buf data to API_fs_Hz */
ALLOC( x_buf_API_fs_Hz, api_buf_samples, opus_int16 );
ret += silk_resampler( temp_resampler_state, x_buf_API_fs_Hz, x_bufFIX, old_buf_samples );
/* 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, silk_SMULBB( fs_kHz, 1000 ), 1 );
/* Correct resampler state 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, api_buf_samples );
#ifndef OPUS_FIXED_POINT
silk_short2float_array( psEnc->x_buf, x_bufFIX, new_buf_samples);
#endif
}
}
psEnc->sCmn.prev_API_fs_Hz = psEnc->sCmn.API_fs_Hz;
RESTORE_STACK;
return ret;
}
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 = silk_SMULBB( PacketSize_ms, fs_kHz );
psEnc->sCmn.pitch_LPC_win_length = silk_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 = silk_DIV32_16( PacketSize_ms, MAX_FRAME_LENGTH_MS );
psEnc->sCmn.nb_subfr = MAX_NB_SUBFR;
psEnc->sCmn.frame_length = silk_SMULBB( 20, fs_kHz );
psEnc->sCmn.pitch_LPC_win_length = silk_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 */
silk_assert( fs_kHz == 8 || fs_kHz == 12 || fs_kHz == 16 );
silk_assert( psEnc->sCmn.nb_subfr == 2 || psEnc->sCmn.nb_subfr == 4 );
if( psEnc->sCmn.fs_kHz != fs_kHz ) {
/* reset part of the state */
silk_memset( &psEnc->sShape, 0, sizeof( psEnc->sShape ) );
silk_memset( &psEnc->sPrefilt, 0, sizeof( psEnc->sPrefilt ) );
silk_memset( &psEnc->sCmn.sNSQ, 0, sizeof( psEnc->sCmn.sNSQ ) );
silk_memset( psEnc->sCmn.prev_NLSFq_Q15, 0, sizeof( psEnc->sCmn.prev_NLSFq_Q15 ) );
silk_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_gain_Q16 = 65536;
psEnc->sCmn.prevSignalType = TYPE_NO_VOICE_ACTIVITY;
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 = silk_SMULBB( psEnc->sCmn.subfr_length, psEnc->sCmn.nb_subfr );
psEnc->sCmn.ltp_mem_length = silk_SMULBB( LTP_MEM_LENGTH_MS, fs_kHz );
psEnc->sCmn.la_pitch = silk_SMULBB( LA_PITCH_MS, fs_kHz );
psEnc->sCmn.max_pitch_lag = silk_SMULBB( 18, fs_kHz );
if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) {
psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS, fs_kHz );
} else {
psEnc->sCmn.pitch_LPC_win_length = silk_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 */
silk_assert( ( psEnc->sCmn.subfr_length * psEnc->sCmn.nb_subfr ) == psEnc->sCmn.frame_length );
return ret;
}
/* Convert Left/Right stereo signal to adaptive Mid/Side representation */
void silk_stereo_LR_to_MS(
stereo_enc_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 */
opus_int8 ix[ 2 ][ 3 ], /* O Quantization indices */
opus_int8 *mid_only_flag, /* O Flag: only mid signal coded */
opus_int32 mid_side_rates_bps[], /* O Bitrates for mid and side signals */
opus_int32 total_rate_bps, /* I Total bitrate */
opus_int prev_speech_act_Q8, /* I Speech activity level in previous frame */
opus_int toMono, /* I Last frame before a stereo->mono transition */
opus_int fs_kHz, /* I Sample rate (kHz) */
opus_int frame_length /* I Number of samples */
)
{
opus_int n, is10msFrame, denom_Q16, delta0_Q13, delta1_Q13;
opus_int32 sum, diff, smooth_coef_Q16, pred_Q13[ 2 ], pred0_Q13, pred1_Q13;
opus_int32 LP_ratio_Q14, HP_ratio_Q14, frac_Q16, frac_3_Q16, min_mid_rate_bps, width_Q14, w_Q24, deltaw_Q24;
VARDECL( opus_int16, side );
VARDECL( opus_int16, LP_mid );
VARDECL( opus_int16, HP_mid );
VARDECL( opus_int16, LP_side );
VARDECL( opus_int16, HP_side );
opus_int16 *mid = &x1[ -2 ];
SAVE_STACK;
ALLOC( side, frame_length + 2, opus_int16 );
/* Convert to basic mid/side signals */
for( n = 0; n < frame_length + 2; n++ ) {
sum = x1[ n - 2 ] + (opus_int32)x2[ n - 2 ];
diff = x1[ n - 2 ] - (opus_int32)x2[ n - 2 ];
mid[ n ] = (opus_int16)silk_RSHIFT_ROUND( sum, 1 );
side[ n ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( diff, 1 ) );
}
/* Buffering */
silk_memcpy( mid, state->sMid, 2 * sizeof( opus_int16 ) );
silk_memcpy( side, state->sSide, 2 * sizeof( opus_int16 ) );
silk_memcpy( state->sMid, &mid[ frame_length ], 2 * sizeof( opus_int16 ) );
silk_memcpy( state->sSide, &side[ frame_length ], 2 * sizeof( opus_int16 ) );
/* LP and HP filter mid signal */
ALLOC( LP_mid, frame_length, opus_int16 );
ALLOC( HP_mid, frame_length, opus_int16 );
for( n = 0; n < frame_length; n++ ) {
sum = silk_RSHIFT_ROUND( silk_ADD_LSHIFT( mid[ n ] + mid[ n + 2 ], mid[ n + 1 ], 1 ), 2 );
LP_mid[ n ] = sum;
HP_mid[ n ] = mid[ n + 1 ] - sum;
}
/* LP and HP filter side signal */
ALLOC( LP_side, frame_length, opus_int16 );
ALLOC( HP_side, frame_length, opus_int16 );
for( n = 0; n < frame_length; n++ ) {
sum = silk_RSHIFT_ROUND( silk_ADD_LSHIFT( side[ n ] + side[ n + 2 ], side[ n + 1 ], 1 ), 2 );
LP_side[ n ] = sum;
HP_side[ n ] = side[ n + 1 ] - sum;
}
/* Find energies and predictors */
is10msFrame = frame_length == 10 * fs_kHz;
smooth_coef_Q16 = is10msFrame ?
SILK_FIX_CONST( STEREO_RATIO_SMOOTH_COEF / 2, 16 ) :
SILK_FIX_CONST( STEREO_RATIO_SMOOTH_COEF, 16 );
smooth_coef_Q16 = silk_SMULWB( silk_SMULBB( prev_speech_act_Q8, prev_speech_act_Q8 ), smooth_coef_Q16 );
pred_Q13[ 0 ] = silk_stereo_find_predictor( &LP_ratio_Q14, LP_mid, LP_side, &state->mid_side_amp_Q0[ 0 ], frame_length, smooth_coef_Q16 );
pred_Q13[ 1 ] = silk_stereo_find_predictor( &HP_ratio_Q14, HP_mid, HP_side, &state->mid_side_amp_Q0[ 2 ], frame_length, smooth_coef_Q16 );
/* Ratio of the norms of residual and mid signals */
frac_Q16 = silk_SMLABB( HP_ratio_Q14, LP_ratio_Q14, 3 );
frac_Q16 = silk_min( frac_Q16, SILK_FIX_CONST( 1, 16 ) );
/* Determine bitrate distribution between mid and side, and possibly reduce stereo width */
total_rate_bps -= is10msFrame ? 1200 : 600; /* Subtract approximate bitrate for coding stereo parameters */
if( total_rate_bps < 1 ) {
total_rate_bps = 1;
}
min_mid_rate_bps = silk_SMLABB( 2000, fs_kHz, 900 );
silk_assert( min_mid_rate_bps < 32767 );
/* Default bitrate distribution: 8 parts for Mid and (5+3*frac) parts for Side. so: mid_rate = ( 8 / ( 13 + 3 * frac ) ) * total_ rate */
frac_3_Q16 = silk_MUL( 3, frac_Q16 );
mid_side_rates_bps[ 0 ] = silk_DIV32_varQ( total_rate_bps, SILK_FIX_CONST( 8 + 5, 16 ) + frac_3_Q16, 16+3 );
/* If Mid bitrate below minimum, reduce stereo width */
if( mid_side_rates_bps[ 0 ] < min_mid_rate_bps ) {
mid_side_rates_bps[ 0 ] = min_mid_rate_bps;
mid_side_rates_bps[ 1 ] = total_rate_bps - mid_side_rates_bps[ 0 ];
/* width = 4 * ( 2 * side_rate - min_rate ) / ( ( 1 + 3 * frac ) * min_rate ) */
width_Q14 = silk_DIV32_varQ( silk_LSHIFT( mid_side_rates_bps[ 1 ], 1 ) - min_mid_rate_bps,
silk_SMULWB( SILK_FIX_CONST( 1, 16 ) + frac_3_Q16, min_mid_rate_bps ), 14+2 );
width_Q14 = silk_LIMIT( width_Q14, 0, SILK_FIX_CONST( 1, 14 ) );
} else {
mid_side_rates_bps[ 1 ] = total_rate_bps - mid_side_rates_bps[ 0 ];
width_Q14 = SILK_FIX_CONST( 1, 14 );
}
/* Smoother */
state->smth_width_Q14 = (opus_int16)silk_SMLAWB( state->smth_width_Q14, width_Q14 - state->smth_width_Q14, smooth_coef_Q16 );
/* At very low bitrates or for inputs that are nearly amplitude panned, switch to panned-mono coding */
*mid_only_flag = 0;
if( toMono ) {
/* Last frame before stereo->mono transition; collapse stereo width */
width_Q14 = 0;
pred_Q13[ 0 ] = 0;
pred_Q13[ 1 ] = 0;
silk_stereo_quant_pred( pred_Q13, ix );
} else if( state->width_prev_Q14 == 0 &&
( 8 * total_rate_bps < 13 * min_mid_rate_bps || silk_SMULWB( frac_Q16, state->smth_width_Q14 ) < SILK_FIX_CONST( 0.05, 14 ) ) )
{
/* Code as panned-mono; previous frame already had zero width */
/* Scale down and quantize predictors */
pred_Q13[ 0 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 0 ] ), 14 );
pred_Q13[ 1 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 1 ] ), 14 );
silk_stereo_quant_pred( pred_Q13, ix );
/* Collapse stereo width */
width_Q14 = 0;
pred_Q13[ 0 ] = 0;
pred_Q13[ 1 ] = 0;
mid_side_rates_bps[ 0 ] = total_rate_bps;
mid_side_rates_bps[ 1 ] = 0;
*mid_only_flag = 1;
} else if( state->width_prev_Q14 != 0 &&
( 8 * total_rate_bps < 11 * min_mid_rate_bps || silk_SMULWB( frac_Q16, state->smth_width_Q14 ) < SILK_FIX_CONST( 0.02, 14 ) ) )
{
/* Transition to zero-width stereo */
/* Scale down and quantize predictors */
pred_Q13[ 0 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 0 ] ), 14 );
pred_Q13[ 1 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 1 ] ), 14 );
silk_stereo_quant_pred( pred_Q13, ix );
/* Collapse stereo width */
width_Q14 = 0;
pred_Q13[ 0 ] = 0;
pred_Q13[ 1 ] = 0;
} else if( state->smth_width_Q14 > SILK_FIX_CONST( 0.95, 14 ) ) {
/* Full-width stereo coding */
silk_stereo_quant_pred( pred_Q13, ix );
width_Q14 = SILK_FIX_CONST( 1, 14 );
} else {
/* Reduced-width stereo coding; scale down and quantize predictors */
pred_Q13[ 0 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 0 ] ), 14 );
pred_Q13[ 1 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 1 ] ), 14 );
silk_stereo_quant_pred( pred_Q13, ix );
width_Q14 = state->smth_width_Q14;
}
/* Make sure to keep on encoding until the tapered output has been transmitted */
if( *mid_only_flag == 1 ) {
state->silent_side_len += frame_length - STEREO_INTERP_LEN_MS * fs_kHz;
if( state->silent_side_len < LA_SHAPE_MS * fs_kHz ) {
*mid_only_flag = 0;
} else {
/* Limit to avoid wrapping around */
state->silent_side_len = 10000;
}
} else {
state->silent_side_len = 0;
}
if( *mid_only_flag == 0 && mid_side_rates_bps[ 1 ] < 1 ) {
mid_side_rates_bps[ 1 ] = 1;
mid_side_rates_bps[ 0 ] = silk_max_int( 1, total_rate_bps - mid_side_rates_bps[ 1 ]);
}
/* Interpolate predictors and subtract prediction from side channel */
pred0_Q13 = -state->pred_prev_Q13[ 0 ];
pred1_Q13 = -state->pred_prev_Q13[ 1 ];
w_Q24 = silk_LSHIFT( state->width_prev_Q14, 10 );
denom_Q16 = silk_DIV32_16( (opus_int32)1 << 16, STEREO_INTERP_LEN_MS * fs_kHz );
delta0_Q13 = -silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 0 ] - state->pred_prev_Q13[ 0 ], denom_Q16 ), 16 );
delta1_Q13 = -silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 1 ] - state->pred_prev_Q13[ 1 ], denom_Q16 ), 16 );
deltaw_Q24 = silk_LSHIFT( silk_SMULWB( width_Q14 - state->width_prev_Q14, denom_Q16 ), 10 );
for( n = 0; n < STEREO_INTERP_LEN_MS * fs_kHz; n++ ) {
pred0_Q13 += delta0_Q13;
pred1_Q13 += delta1_Q13;
w_Q24 += deltaw_Q24;
sum = silk_LSHIFT( silk_ADD_LSHIFT( mid[ n ] + mid[ n + 2 ], mid[ n + 1 ], 1 ), 9 ); /* Q11 */
sum = silk_SMLAWB( silk_SMULWB( w_Q24, side[ n + 1 ] ), sum, pred0_Q13 ); /* Q8 */
sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)mid[ n + 1 ], 11 ), pred1_Q13 ); /* Q8 */
x2[ n - 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) );
}
pred0_Q13 = -pred_Q13[ 0 ];
pred1_Q13 = -pred_Q13[ 1 ];
w_Q24 = silk_LSHIFT( width_Q14, 10 );
for( n = STEREO_INTERP_LEN_MS * fs_kHz; n < frame_length; n++ ) {
sum = silk_LSHIFT( silk_ADD_LSHIFT( mid[ n ] + mid[ n + 2 ], mid[ n + 1 ], 1 ), 9 ); /* Q11 */
sum = silk_SMLAWB( silk_SMULWB( w_Q24, side[ n + 1 ] ), sum, pred0_Q13 ); /* Q8 */
sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)mid[ n + 1 ], 11 ), pred1_Q13 ); /* Q8 */
x2[ n - 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) );
}
state->pred_prev_Q13[ 0 ] = (opus_int16)pred_Q13[ 0 ];
state->pred_prev_Q13[ 1 ] = (opus_int16)pred_Q13[ 1 ];
state->width_prev_Q14 = (opus_int16)width_Q14;
RESTORE_STACK;
}
int silk_encode_frame_FIX(silk_encoder_state_FIX * psEnc, /* I/O Pointer to Silk FIX encoder state */
int32_t * pnBytesOut, /* O Pointer to number of payload bytes; */
ec_enc * psRangeEnc, /* I/O compressor data structure */
int condCoding, /* I The type of conditional coding to use */
int maxBits, /* I If > 0: maximum number of output bits */
int useCBR /* I Flag to force constant-bitrate operation */
) {
silk_encoder_control_FIX sEncCtrl;
int i, iter, maxIter, found_upper, found_lower, ret = 0;
int16_t *x_frame;
ec_enc sRangeEnc_copy, sRangeEnc_copy2;
silk_nsq_state sNSQ_copy, sNSQ_copy2;
int32_t seed_copy, nBits, nBits_lower, nBits_upper, gainMult_lower,
gainMult_upper;
int32_t gainsID, gainsID_lower, gainsID_upper;
int16_t gainMult_Q8;
int16_t ec_prevLagIndex_copy;
int ec_prevSignalType_copy;
int8_t LastGainIndex_copy2;
/* This is totally unnecessary but many compilers (including gcc) are too dumb to realise it */
LastGainIndex_copy2 = nBits_lower = nBits_upper = gainMult_lower =
gainMult_upper = 0;
psEnc->sCmn.indices.Seed = psEnc->sCmn.frameCounter++ & 3;
/**************************************************************/
/* Set up Input Pointers, and insert frame in input buffer */
/*************************************************************/
/* start of frame to encode */
x_frame = psEnc->x_buf + psEnc->sCmn.ltp_mem_length;
/***************************************/
/* Ensure smooth bandwidth transitions */
/***************************************/
silk_LP_variable_cutoff(&psEnc->sCmn.sLP, psEnc->sCmn.inputBuf + 1,
psEnc->sCmn.frame_length);
/*******************************************/
/* Copy new frame to front of input buffer */
/*******************************************/
memcpy(x_frame + LA_SHAPE_MS * psEnc->sCmn.fs_kHz,
psEnc->sCmn.inputBuf + 1,
psEnc->sCmn.frame_length * sizeof(int16_t));
if (!psEnc->sCmn.prefillFlag) {
int16_t *res_pitch_frame;
int16_t res_pitch[psEnc->sCmn.la_pitch +
psEnc->sCmn.frame_length +
psEnc->sCmn.ltp_mem_length];
/* start of pitch LPC residual frame */
res_pitch_frame = res_pitch + psEnc->sCmn.ltp_mem_length;
/*****************************************/
/* Find pitch lags, initial LPC analysis */
/*****************************************/
silk_find_pitch_lags_FIX(psEnc, &sEncCtrl, res_pitch, x_frame,
psEnc->sCmn.arch);
/************************/
/* Noise shape analysis */
/************************/
silk_noise_shape_analysis_FIX(psEnc, &sEncCtrl, res_pitch_frame,
x_frame, psEnc->sCmn.arch);
/***************************************************/
/* Find linear prediction coefficients (LPC + LTP) */
/***************************************************/
silk_find_pred_coefs_FIX(psEnc, &sEncCtrl, res_pitch, x_frame,
condCoding);
/****************************************/
/* Process gains */
/****************************************/
silk_process_gains_FIX(psEnc, &sEncCtrl, condCoding);
/*****************************************/
/* Prefiltering for noise shaper */
/*****************************************/
int32_t xfw_Q3[psEnc->sCmn.frame_length];
silk_prefilter_FIX(psEnc, &sEncCtrl, xfw_Q3, x_frame);
/****************************************/
/* Low Bitrate Redundant Encoding */
/****************************************/
silk_LBRR_encode_FIX(psEnc, &sEncCtrl, xfw_Q3, condCoding);
/* Loop over quantizer and entropy coding to control bitrate */
maxIter = 6;
gainMult_Q8 = SILK_FIX_CONST(1, 8);
found_lower = 0;
found_upper = 0;
gainsID =
silk_gains_ID(psEnc->sCmn.indices.GainsIndices,
psEnc->sCmn.nb_subfr);
gainsID_lower = -1;
gainsID_upper = -1;
/* Copy part of the input state */
memcpy(&sRangeEnc_copy, psRangeEnc, sizeof(ec_enc));
memcpy(&sNSQ_copy, &psEnc->sCmn.sNSQ,
sizeof(silk_nsq_state));
seed_copy = psEnc->sCmn.indices.Seed;
ec_prevLagIndex_copy = psEnc->sCmn.ec_prevLagIndex;
ec_prevSignalType_copy = psEnc->sCmn.ec_prevSignalType;
uint8_t ec_buf_copy[1275];
for (iter = 0;; iter++) {
if (gainsID == gainsID_lower) {
nBits = nBits_lower;
} else if (gainsID == gainsID_upper) {
nBits = nBits_upper;
} else {
/* Restore part of the input state */
if (iter > 0) {
memcpy(psRangeEnc, &sRangeEnc_copy,
sizeof(ec_enc));
memcpy(&psEnc->sCmn.sNSQ,
&sNSQ_copy,
sizeof(silk_nsq_state));
psEnc->sCmn.indices.Seed = seed_copy;
psEnc->sCmn.ec_prevLagIndex =
ec_prevLagIndex_copy;
psEnc->sCmn.ec_prevSignalType =
ec_prevSignalType_copy;
}
/*****************************************/
/* Noise shaping quantization */
/*****************************************/
if (psEnc->sCmn.nStatesDelayedDecision > 1
|| psEnc->sCmn.warping_Q16 > 0) {
silk_NSQ_del_dec(&psEnc->sCmn,
&psEnc->sCmn.sNSQ,
&psEnc->sCmn.indices,
xfw_Q3,
psEnc->sCmn.pulses,
sEncCtrl.
PredCoef_Q12[0],
sEncCtrl.LTPCoef_Q14,
sEncCtrl.AR2_Q13,
sEncCtrl.
HarmShapeGain_Q14,
sEncCtrl.Tilt_Q14,
sEncCtrl.LF_shp_Q14,
sEncCtrl.Gains_Q16,
sEncCtrl.pitchL,
sEncCtrl.Lambda_Q10,
sEncCtrl.
LTP_scale_Q14);
} else {
silk_NSQ(&psEnc->sCmn,
&psEnc->sCmn.sNSQ,
&psEnc->sCmn.indices, xfw_Q3,
psEnc->sCmn.pulses,
sEncCtrl.PredCoef_Q12[0],
sEncCtrl.LTPCoef_Q14,
sEncCtrl.AR2_Q13,
sEncCtrl.HarmShapeGain_Q14,
sEncCtrl.Tilt_Q14,
sEncCtrl.LF_shp_Q14,
sEncCtrl.Gains_Q16,
sEncCtrl.pitchL,
sEncCtrl.Lambda_Q10,
sEncCtrl.LTP_scale_Q14);
}
/****************************************/
/* Encode Parameters */
/****************************************/
silk_encode_indices(&psEnc->sCmn, psRangeEnc,
psEnc->sCmn.nFramesEncoded,
0, condCoding);
/****************************************/
/* Encode Excitation Signal */
/****************************************/
silk_encode_pulses(psRangeEnc,
psEnc->sCmn.indices.
signalType,
psEnc->sCmn.indices.
quantOffsetType,
psEnc->sCmn.pulses,
psEnc->sCmn.frame_length);
nBits = ec_tell(psRangeEnc);
if (useCBR == 0 && iter == 0
&& nBits <= maxBits) {
break;
}
}
if (iter == maxIter) {
if (found_lower
&& (gainsID == gainsID_lower
|| nBits > maxBits)) {
/* Restore output state from earlier iteration that did meet the bitrate budget */
memcpy(psRangeEnc,
&sRangeEnc_copy2,
sizeof(ec_enc));
assert(sRangeEnc_copy2.offs <=
1275);
memcpy(psRangeEnc->buf,
ec_buf_copy,
sRangeEnc_copy2.offs);
memcpy(&psEnc->sCmn.sNSQ,
&sNSQ_copy2,
sizeof(silk_nsq_state));
psEnc->sShape.LastGainIndex =
LastGainIndex_copy2;
}
break;
}
if (nBits > maxBits) {
if (found_lower == 0 && iter >= 2) {
/* Adjust the quantizer's rate/distortion tradeoff and discard previous "upper" results */
sEncCtrl.Lambda_Q10 =
silk_ADD_RSHIFT32(sEncCtrl.
Lambda_Q10,
sEncCtrl.
Lambda_Q10, 1);
found_upper = 0;
gainsID_upper = -1;
} else {
found_upper = 1;
nBits_upper = nBits;
gainMult_upper = gainMult_Q8;
gainsID_upper = gainsID;
}
} else if (nBits < maxBits - 5) {
found_lower = 1;
nBits_lower = nBits;
gainMult_lower = gainMult_Q8;
if (gainsID != gainsID_lower) {
gainsID_lower = gainsID;
/* Copy part of the output state */
memcpy(&sRangeEnc_copy2,
psRangeEnc, sizeof(ec_enc));
assert(psRangeEnc->offs <= 1275);
memcpy(ec_buf_copy,
psRangeEnc->buf,
psRangeEnc->offs);
memcpy(&sNSQ_copy2,
&psEnc->sCmn.sNSQ,
sizeof(silk_nsq_state));
LastGainIndex_copy2 =
psEnc->sShape.LastGainIndex;
}
} else {
/* Within 5 bits of budget: close enough */
break;
}
if ((found_lower & found_upper) == 0) {
/* Adjust gain according to high-rate rate/distortion curve */
int32_t gain_factor_Q16;
gain_factor_Q16 =
silk_log2lin(silk_LSHIFT(nBits - maxBits, 7)
/ psEnc->sCmn.frame_length +
SILK_FIX_CONST(16, 7));
gain_factor_Q16 =
silk_min_32(gain_factor_Q16,
SILK_FIX_CONST(2, 16));
if (nBits > maxBits) {
gain_factor_Q16 =
silk_max_32(gain_factor_Q16,
SILK_FIX_CONST(1.3,
16));
}
gainMult_Q8 =
silk_SMULWB(gain_factor_Q16, gainMult_Q8);
} else {
/* Adjust gain by interpolating */
assert(nBits_upper != nBits_lower);
gainMult_Q8 =
gainMult_lower +
silk_DIV32_16(silk_MUL
(gainMult_upper -
gainMult_lower,
maxBits - nBits_lower),
nBits_upper - nBits_lower);
/* New gain multplier must be between 25% and 75% of old range (note that gainMult_upper < gainMult_lower) */
if (gainMult_Q8 >
silk_ADD_RSHIFT32(gainMult_lower,
gainMult_upper -
gainMult_lower, 2)) {
gainMult_Q8 =
silk_ADD_RSHIFT32(gainMult_lower,
gainMult_upper -
gainMult_lower,
2);
} else if (gainMult_Q8 <
silk_SUB_RSHIFT32(gainMult_upper,
gainMult_upper -
gainMult_lower,
2)) {
gainMult_Q8 =
silk_SUB_RSHIFT32(gainMult_upper,
gainMult_upper -
gainMult_lower,
2);
}
}
for (i = 0; i < psEnc->sCmn.nb_subfr; i++) {
sEncCtrl.Gains_Q16[i] =
silk_LSHIFT_SAT32(silk_SMULWB
(sEncCtrl.GainsUnq_Q16[i],
gainMult_Q8), 8);
}
/* Quantize gains */
psEnc->sShape.LastGainIndex =
sEncCtrl.lastGainIndexPrev;
silk_gains_quant(psEnc->sCmn.indices.GainsIndices,
sEncCtrl.Gains_Q16,
&psEnc->sShape.LastGainIndex,
condCoding == CODE_CONDITIONALLY,
psEnc->sCmn.nb_subfr);
/* Unique identifier of gains vector */
gainsID =
silk_gains_ID(psEnc->sCmn.indices.GainsIndices,
psEnc->sCmn.nb_subfr);
}
}
/* Update input buffer */
memmove(psEnc->x_buf, &psEnc->x_buf[psEnc->sCmn.frame_length],
(psEnc->sCmn.ltp_mem_length + LA_SHAPE_MS * psEnc->sCmn.fs_kHz) * sizeof(int16_t));
/* Exit without entropy coding */
if (psEnc->sCmn.prefillFlag) {
/* No payload */
*pnBytesOut = 0;
return ret;
}
/* Parameters needed for next frame */
psEnc->sCmn.prevLag = sEncCtrl.pitchL[psEnc->sCmn.nb_subfr - 1];
psEnc->sCmn.prevSignalType = psEnc->sCmn.indices.signalType;
/****************************************/
/* Finalize payload */
/****************************************/
psEnc->sCmn.first_frame_after_reset = 0;
/* Payload size */
*pnBytesOut = silk_RSHIFT(ec_tell(psRangeEnc) + 7, 3);
return ret;
}
/* Initialize/reset the resampler state for a given pair of input/output sampling rates */
opus_int silk_resampler_init(
silk_resampler_state_struct *S, /* I/O Resampler state */
opus_int32 Fs_Hz_in, /* I Input sampling rate (Hz) */
opus_int32 Fs_Hz_out, /* I Output sampling rate (Hz) */
opus_int forEnc /* I If 1: encoder; if 0: decoder */
)
{
opus_int up2x;
/* Clear state */
silk_memset( S, 0, sizeof( silk_resampler_state_struct ) );
/* Input checking */
if( forEnc ) {
if( ( Fs_Hz_in != 8000 && Fs_Hz_in != 12000 && Fs_Hz_in != 16000 && Fs_Hz_in != 24000 && Fs_Hz_in != 48000 ) ||
( Fs_Hz_out != 8000 && Fs_Hz_out != 12000 && Fs_Hz_out != 16000 ) ) {
silk_assert( 0 );
return -1;
}
S->inputDelay = delay_matrix_enc[ rateID( Fs_Hz_in ) ][ rateID( Fs_Hz_out ) ];
} else {
if( ( Fs_Hz_in != 8000 && Fs_Hz_in != 12000 && Fs_Hz_in != 16000 ) ||
( Fs_Hz_out != 8000 && Fs_Hz_out != 12000 && Fs_Hz_out != 16000 && Fs_Hz_out != 24000 && Fs_Hz_out != 48000 ) ) {
silk_assert( 0 );
return -1;
}
S->inputDelay = delay_matrix_dec[ rateID( Fs_Hz_in ) ][ rateID( Fs_Hz_out ) ];
}
S->Fs_in_kHz = silk_DIV32_16( Fs_Hz_in, 1000 );
S->Fs_out_kHz = silk_DIV32_16( Fs_Hz_out, 1000 );
/* Number of samples processed per batch */
S->batchSize = S->Fs_in_kHz * RESAMPLER_MAX_BATCH_SIZE_MS;
/* Find resampler with the right sampling ratio */
up2x = 0;
if( Fs_Hz_out > Fs_Hz_in ) {
/* Upsample */
if( Fs_Hz_out == silk_MUL( Fs_Hz_in, 2 ) ) { /* Fs_out : Fs_in = 2 : 1 */
/* Special case: directly use 2x upsampler */
S->resampler_function = USE_silk_resampler_private_up2_HQ_wrapper;
} else {
/* Default resampler */
S->resampler_function = USE_silk_resampler_private_IIR_FIR;
up2x = 1;
}
} else if ( Fs_Hz_out < Fs_Hz_in ) {
/* Downsample */
S->resampler_function = USE_silk_resampler_private_down_FIR;
if( silk_MUL( Fs_Hz_out, 4 ) == silk_MUL( Fs_Hz_in, 3 ) ) { /* Fs_out : Fs_in = 3 : 4 */
S->FIR_Fracs = 3;
S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR0;
S->Coefs = silk_Resampler_3_4_COEFS;
} else if( silk_MUL( Fs_Hz_out, 3 ) == silk_MUL( Fs_Hz_in, 2 ) ) { /* Fs_out : Fs_in = 2 : 3 */
S->FIR_Fracs = 2;
S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR0;
S->Coefs = silk_Resampler_2_3_COEFS;
} else if( silk_MUL( Fs_Hz_out, 2 ) == Fs_Hz_in ) { /* Fs_out : Fs_in = 1 : 2 */
S->FIR_Fracs = 1;
S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR1;
S->Coefs = silk_Resampler_1_2_COEFS;
} else if( silk_MUL( Fs_Hz_out, 3 ) == Fs_Hz_in ) { /* Fs_out : Fs_in = 1 : 3 */
S->FIR_Fracs = 1;
S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR2;
S->Coefs = silk_Resampler_1_3_COEFS;
} else if( silk_MUL( Fs_Hz_out, 4 ) == Fs_Hz_in ) { /* Fs_out : Fs_in = 1 : 4 */
S->FIR_Fracs = 1;
S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR2;
S->Coefs = silk_Resampler_1_4_COEFS;
} else if( silk_MUL( Fs_Hz_out, 6 ) == Fs_Hz_in ) { /* Fs_out : Fs_in = 1 : 6 */
S->FIR_Fracs = 1;
S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR2;
S->Coefs = silk_Resampler_1_6_COEFS;
} else {
/* None available */
silk_assert( 0 );
return -1;
}
} else {
/* Input and output sampling rates are equal: copy */
S->resampler_function = USE_silk_resampler_copy;
}
/* Ratio of input/output samples */
S->invRatio_Q16 = silk_LSHIFT32( silk_DIV32( silk_LSHIFT32( Fs_Hz_in, 14 + up2x ), Fs_Hz_out ), 2 );
/* Make sure the ratio is rounded up */
while( silk_SMULWW( S->invRatio_Q16, Fs_Hz_out ) < silk_LSHIFT32( Fs_Hz_in, up2x ) ) {
S->invRatio_Q16++;
}
return 0;
}
/* encControl->payloadSize_ms is set to */
opus_int silk_Encode( /* O Returns error code */
void *encState, /* I/O State */
silk_EncControlStruct *encControl, /* I Control status */
const opus_int16 *samplesIn, /* I Speech sample input vector */
opus_int nSamplesIn, /* I Number of samples in input vector */
ec_enc *psRangeEnc, /* I/O Compressor data structure */
opus_int32 *nBytesOut, /* I/O Number of bytes in payload (input: Max bytes) */
const opus_int prefillFlag /* I Flag to indicate prefilling buffers no coding */
)
{
opus_int n, i, nBits, flags, tmp_payloadSize_ms = 0, tmp_complexity = 0, ret = 0;
opus_int nSamplesToBuffer, nSamplesToBufferMax, nBlocksOf10ms;
opus_int nSamplesFromInput = 0, nSamplesFromInputMax;
opus_int speech_act_thr_for_switch_Q8;
opus_int32 TargetRate_bps, MStargetRates_bps[ 2 ], channelRate_bps, LBRR_symbol, sum;
silk_encoder *psEnc = ( silk_encoder * )encState;
VARDECL( opus_int16, buf );
opus_int transition, curr_block, tot_blocks;
SAVE_STACK;
if (encControl->reducedDependency)
{
psEnc->state_Fxx[0].sCmn.first_frame_after_reset = 1;
psEnc->state_Fxx[1].sCmn.first_frame_after_reset = 1;
}
psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded = psEnc->state_Fxx[ 1 ].sCmn.nFramesEncoded = 0;
/* Check values in encoder control structure */
if( ( ret = check_control_input( encControl ) ) != 0 ) {
silk_assert( 0 );
RESTORE_STACK;
return ret;
}
encControl->switchReady = 0;
if( encControl->nChannelsInternal > psEnc->nChannelsInternal ) {
/* Mono -> Stereo transition: init state of second channel and stereo state */
ret += silk_init_encoder( &psEnc->state_Fxx[ 1 ], psEnc->state_Fxx[ 0 ].sCmn.arch );
silk_memset( psEnc->sStereo.pred_prev_Q13, 0, sizeof( psEnc->sStereo.pred_prev_Q13 ) );
silk_memset( psEnc->sStereo.sSide, 0, sizeof( psEnc->sStereo.sSide ) );
psEnc->sStereo.mid_side_amp_Q0[ 0 ] = 0;
psEnc->sStereo.mid_side_amp_Q0[ 1 ] = 1;
psEnc->sStereo.mid_side_amp_Q0[ 2 ] = 0;
psEnc->sStereo.mid_side_amp_Q0[ 3 ] = 1;
psEnc->sStereo.width_prev_Q14 = 0;
psEnc->sStereo.smth_width_Q14 = SILK_FIX_CONST( 1, 14 );
if( psEnc->nChannelsAPI == 2 ) {
silk_memcpy( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state, &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, sizeof( silk_resampler_state_struct ) );
silk_memcpy( &psEnc->state_Fxx[ 1 ].sCmn.In_HP_State, &psEnc->state_Fxx[ 0 ].sCmn.In_HP_State, sizeof( psEnc->state_Fxx[ 1 ].sCmn.In_HP_State ) );
}
}
transition = (encControl->payloadSize_ms != psEnc->state_Fxx[ 0 ].sCmn.PacketSize_ms) || (psEnc->nChannelsInternal != encControl->nChannelsInternal);
psEnc->nChannelsAPI = encControl->nChannelsAPI;
psEnc->nChannelsInternal = encControl->nChannelsInternal;
nBlocksOf10ms = silk_DIV32( 100 * nSamplesIn, encControl->API_sampleRate );
tot_blocks = ( nBlocksOf10ms > 1 ) ? nBlocksOf10ms >> 1 : 1;
curr_block = 0;
if( prefillFlag ) {
/* Only accept input length of 10 ms */
if( nBlocksOf10ms != 1 ) {
silk_assert( 0 );
RESTORE_STACK;
return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
}
/* Reset Encoder */
for( n = 0; n < encControl->nChannelsInternal; n++ ) {
ret = silk_init_encoder( &psEnc->state_Fxx[ n ], psEnc->state_Fxx[ n ].sCmn.arch );
silk_assert( !ret );
}
tmp_payloadSize_ms = encControl->payloadSize_ms;
encControl->payloadSize_ms = 10;
tmp_complexity = encControl->complexity;
encControl->complexity = 0;
for( n = 0; n < encControl->nChannelsInternal; n++ ) {
psEnc->state_Fxx[ n ].sCmn.controlled_since_last_payload = 0;
psEnc->state_Fxx[ n ].sCmn.prefillFlag = 1;
}
} else {
/* Only accept input lengths that are a multiple of 10 ms */
if( nBlocksOf10ms * encControl->API_sampleRate != 100 * nSamplesIn || nSamplesIn < 0 ) {
silk_assert( 0 );
RESTORE_STACK;
return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
}
/* Make sure no more than one packet can be produced */
if( 1000 * (opus_int32)nSamplesIn > encControl->payloadSize_ms * encControl->API_sampleRate ) {
silk_assert( 0 );
RESTORE_STACK;
return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
}
}
TargetRate_bps = silk_RSHIFT32( encControl->bitRate, encControl->nChannelsInternal - 1 );
for( n = 0; n < encControl->nChannelsInternal; n++ ) {
/* Force the side channel to the same rate as the mid */
opus_int force_fs_kHz = (n==1) ? psEnc->state_Fxx[0].sCmn.fs_kHz : 0;
if( ( ret = silk_control_encoder( &psEnc->state_Fxx[ n ], encControl, TargetRate_bps, psEnc->allowBandwidthSwitch, n, force_fs_kHz ) ) != 0 ) {
silk_assert( 0 );
RESTORE_STACK;
return ret;
}
if( psEnc->state_Fxx[n].sCmn.first_frame_after_reset || transition ) {
for( i = 0; i < psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket; i++ ) {
psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i ] = 0;
}
}
psEnc->state_Fxx[ n ].sCmn.inDTX = psEnc->state_Fxx[ n ].sCmn.useDTX;
}
silk_assert( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 0 ].sCmn.fs_kHz == psEnc->state_Fxx[ 1 ].sCmn.fs_kHz );
/* Input buffering/resampling and encoding */
nSamplesToBufferMax =
10 * nBlocksOf10ms * psEnc->state_Fxx[ 0 ].sCmn.fs_kHz;
nSamplesFromInputMax =
silk_DIV32_16( nSamplesToBufferMax *
psEnc->state_Fxx[ 0 ].sCmn.API_fs_Hz,
psEnc->state_Fxx[ 0 ].sCmn.fs_kHz * 1000 );
ALLOC( buf, nSamplesFromInputMax, opus_int16 );
while( 1 ) {
nSamplesToBuffer = psEnc->state_Fxx[ 0 ].sCmn.frame_length - psEnc->state_Fxx[ 0 ].sCmn.inputBufIx;
nSamplesToBuffer = silk_min( nSamplesToBuffer, nSamplesToBufferMax );
nSamplesFromInput = silk_DIV32_16( nSamplesToBuffer * psEnc->state_Fxx[ 0 ].sCmn.API_fs_Hz, psEnc->state_Fxx[ 0 ].sCmn.fs_kHz * 1000 );
/* Resample and write to buffer */
if( encControl->nChannelsAPI == 2 && encControl->nChannelsInternal == 2 ) {
opus_int id = psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded;
for( n = 0; n < nSamplesFromInput; n++ ) {
buf[ n ] = samplesIn[ 2 * n ];
}
/* Making sure to start both resamplers from the same state when switching from mono to stereo */
if( psEnc->nPrevChannelsInternal == 1 && id==0 ) {
silk_memcpy( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state, &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, sizeof(psEnc->state_Fxx[ 1 ].sCmn.resampler_state));
}
ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state,
&psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput );
psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer;
nSamplesToBuffer = psEnc->state_Fxx[ 1 ].sCmn.frame_length - psEnc->state_Fxx[ 1 ].sCmn.inputBufIx;
nSamplesToBuffer = silk_min( nSamplesToBuffer, 10 * nBlocksOf10ms * psEnc->state_Fxx[ 1 ].sCmn.fs_kHz );
for( n = 0; n < nSamplesFromInput; n++ ) {
buf[ n ] = samplesIn[ 2 * n + 1 ];
}
ret += silk_resampler( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state,
&psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ psEnc->state_Fxx[ 1 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput );
psEnc->state_Fxx[ 1 ].sCmn.inputBufIx += nSamplesToBuffer;
} else if( encControl->nChannelsAPI == 2 && encControl->nChannelsInternal == 1 ) {
/* Combine left and right channels before resampling */
for( n = 0; n < nSamplesFromInput; n++ ) {
sum = samplesIn[ 2 * n ] + samplesIn[ 2 * n + 1 ];
buf[ n ] = (opus_int16)silk_RSHIFT_ROUND( sum, 1 );
}
ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state,
&psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput );
/* On the first mono frame, average the results for the two resampler states */
if( psEnc->nPrevChannelsInternal == 2 && psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded == 0 ) {
ret += silk_resampler( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state,
&psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ psEnc->state_Fxx[ 1 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput );
for( n = 0; n < psEnc->state_Fxx[ 0 ].sCmn.frame_length; n++ ) {
psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx+n+2 ] =
silk_RSHIFT(psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx+n+2 ]
+ psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ psEnc->state_Fxx[ 1 ].sCmn.inputBufIx+n+2 ], 1);
}
}
psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer;
} else {
silk_assert( encControl->nChannelsAPI == 1 && encControl->nChannelsInternal == 1 );
silk_memcpy(buf, samplesIn, nSamplesFromInput*sizeof(opus_int16));
ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state,
&psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput );
psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer;
}
samplesIn += nSamplesFromInput * encControl->nChannelsAPI;
nSamplesIn -= nSamplesFromInput;
/* Default */
psEnc->allowBandwidthSwitch = 0;
/* Silk encoder */
if( psEnc->state_Fxx[ 0 ].sCmn.inputBufIx >= psEnc->state_Fxx[ 0 ].sCmn.frame_length ) {
/* Enough data in input buffer, so encode */
silk_assert( psEnc->state_Fxx[ 0 ].sCmn.inputBufIx == psEnc->state_Fxx[ 0 ].sCmn.frame_length );
silk_assert( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 1 ].sCmn.inputBufIx == psEnc->state_Fxx[ 1 ].sCmn.frame_length );
/* Deal with LBRR data */
if( psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded == 0 && !prefillFlag ) {
/* Create space at start of payload for VAD and FEC flags */
opus_uint8 iCDF[ 2 ] = { 0, 0 };
iCDF[ 0 ] = 256 - silk_RSHIFT( 256, ( psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket + 1 ) * encControl->nChannelsInternal );
ec_enc_icdf( psRangeEnc, 0, iCDF, 8 );
/* Encode any LBRR data from previous packet */
/* Encode LBRR flags */
for( n = 0; n < encControl->nChannelsInternal; n++ ) {
LBRR_symbol = 0;
for( i = 0; i < psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket; i++ ) {
LBRR_symbol |= silk_LSHIFT( psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i ], i );
}
psEnc->state_Fxx[ n ].sCmn.LBRR_flag = LBRR_symbol > 0 ? 1 : 0;
if( LBRR_symbol && psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket > 1 ) {
ec_enc_icdf( psRangeEnc, LBRR_symbol - 1, silk_LBRR_flags_iCDF_ptr[ psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket - 2 ], 8 );
}
}
/* Code LBRR indices and excitation signals */
for( i = 0; i < psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket; i++ ) {
for( n = 0; n < encControl->nChannelsInternal; n++ ) {
if( psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i ] ) {
opus_int condCoding;
if( encControl->nChannelsInternal == 2 && n == 0 ) {
silk_stereo_encode_pred( psRangeEnc, psEnc->sStereo.predIx[ i ] );
/* For LBRR data there's no need to code the mid-only flag if the side-channel LBRR flag is set */
if( psEnc->state_Fxx[ 1 ].sCmn.LBRR_flags[ i ] == 0 ) {
silk_stereo_encode_mid_only( psRangeEnc, psEnc->sStereo.mid_only_flags[ i ] );
}
}
/* Use conditional coding if previous frame available */
if( i > 0 && psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i - 1 ] ) {
condCoding = CODE_CONDITIONALLY;
} else {
condCoding = CODE_INDEPENDENTLY;
}
silk_encode_indices( &psEnc->state_Fxx[ n ].sCmn, psRangeEnc, i, 1, condCoding );
silk_encode_pulses( psRangeEnc, psEnc->state_Fxx[ n ].sCmn.indices_LBRR[i].signalType, psEnc->state_Fxx[ n ].sCmn.indices_LBRR[i].quantOffsetType,
psEnc->state_Fxx[ n ].sCmn.pulses_LBRR[ i ], psEnc->state_Fxx[ n ].sCmn.frame_length );
}
}
}
/* Reset LBRR flags */
for( n = 0; n < encControl->nChannelsInternal; n++ ) {
silk_memset( psEnc->state_Fxx[ n ].sCmn.LBRR_flags, 0, sizeof( psEnc->state_Fxx[ n ].sCmn.LBRR_flags ) );
}
psEnc->nBitsUsedLBRR = ec_tell( psRangeEnc );
}
silk_HP_variable_cutoff( psEnc->state_Fxx );
/* Total target bits for packet */
nBits = silk_DIV32_16( silk_MUL( encControl->bitRate, encControl->payloadSize_ms ), 1000 );
/* Subtract bits used for LBRR */
if( !prefillFlag ) {
nBits -= psEnc->nBitsUsedLBRR;
}
/* Divide by number of uncoded frames left in packet */
nBits = silk_DIV32_16( nBits, psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket );
/* Convert to bits/second */
if( encControl->payloadSize_ms == 10 ) {
TargetRate_bps = silk_SMULBB( nBits, 100 );
} else {
TargetRate_bps = silk_SMULBB( nBits, 50 );
}
/* Subtract fraction of bits in excess of target in previous frames and packets */
TargetRate_bps -= silk_DIV32_16( silk_MUL( psEnc->nBitsExceeded, 1000 ), BITRESERVOIR_DECAY_TIME_MS );
if( !prefillFlag && psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded > 0 ) {
/* Compare actual vs target bits so far in this packet */
opus_int32 bitsBalance = ec_tell( psRangeEnc ) - psEnc->nBitsUsedLBRR - nBits * psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded;
TargetRate_bps -= silk_DIV32_16( silk_MUL( bitsBalance, 1000 ), BITRESERVOIR_DECAY_TIME_MS );
}
/* Never exceed input bitrate */
TargetRate_bps = silk_LIMIT( TargetRate_bps, encControl->bitRate, 5000 );
/* Convert Left/Right to Mid/Side */
if( encControl->nChannelsInternal == 2 ) {
silk_stereo_LR_to_MS( &psEnc->sStereo, &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ 2 ], &psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ 2 ],
psEnc->sStereo.predIx[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ], &psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ],
MStargetRates_bps, TargetRate_bps, psEnc->state_Fxx[ 0 ].sCmn.speech_activity_Q8, encControl->toMono,
psEnc->state_Fxx[ 0 ].sCmn.fs_kHz, psEnc->state_Fxx[ 0 ].sCmn.frame_length );
if( psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] == 0 ) {
/* Reset side channel encoder memory for first frame with side coding */
if( psEnc->prev_decode_only_middle == 1 ) {
silk_memset( &psEnc->state_Fxx[ 1 ].sShape, 0, sizeof( psEnc->state_Fxx[ 1 ].sShape ) );
silk_memset( &psEnc->state_Fxx[ 1 ].sPrefilt, 0, sizeof( psEnc->state_Fxx[ 1 ].sPrefilt ) );
silk_memset( &psEnc->state_Fxx[ 1 ].sCmn.sNSQ, 0, sizeof( psEnc->state_Fxx[ 1 ].sCmn.sNSQ ) );
silk_memset( psEnc->state_Fxx[ 1 ].sCmn.prev_NLSFq_Q15, 0, sizeof( psEnc->state_Fxx[ 1 ].sCmn.prev_NLSFq_Q15 ) );
silk_memset( &psEnc->state_Fxx[ 1 ].sCmn.sLP.In_LP_State, 0, sizeof( psEnc->state_Fxx[ 1 ].sCmn.sLP.In_LP_State ) );
psEnc->state_Fxx[ 1 ].sCmn.prevLag = 100;
psEnc->state_Fxx[ 1 ].sCmn.sNSQ.lagPrev = 100;
psEnc->state_Fxx[ 1 ].sShape.LastGainIndex = 10;
psEnc->state_Fxx[ 1 ].sCmn.prevSignalType = TYPE_NO_VOICE_ACTIVITY;
psEnc->state_Fxx[ 1 ].sCmn.sNSQ.prev_gain_Q16 = 65536;
psEnc->state_Fxx[ 1 ].sCmn.first_frame_after_reset = 1;
}
silk_encode_do_VAD_Fxx( &psEnc->state_Fxx[ 1 ] );
} else {
psEnc->state_Fxx[ 1 ].sCmn.VAD_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] = 0;
}
if( !prefillFlag ) {
silk_stereo_encode_pred( psRangeEnc, psEnc->sStereo.predIx[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] );
if( psEnc->state_Fxx[ 1 ].sCmn.VAD_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] == 0 ) {
silk_stereo_encode_mid_only( psRangeEnc, psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] );
}
}
} else {
/* Buffering */
silk_memcpy( psEnc->state_Fxx[ 0 ].sCmn.inputBuf, psEnc->sStereo.sMid, 2 * sizeof( opus_int16 ) );
silk_memcpy( psEnc->sStereo.sMid, &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.frame_length ], 2 * sizeof( opus_int16 ) );
}
silk_encode_do_VAD_Fxx( &psEnc->state_Fxx[ 0 ] );
/* Encode */
for( n = 0; n < encControl->nChannelsInternal; n++ ) {
opus_int maxBits, useCBR;
/* Handling rate constraints */
maxBits = encControl->maxBits;
if( tot_blocks == 2 && curr_block == 0 ) {
maxBits = maxBits * 3 / 5;
} else if( tot_blocks == 3 ) {
if( curr_block == 0 ) {
maxBits = maxBits * 2 / 5;
} else if( curr_block == 1 ) {
maxBits = maxBits * 3 / 4;
}
}
useCBR = encControl->useCBR && curr_block == tot_blocks - 1;
if( encControl->nChannelsInternal == 1 ) {
channelRate_bps = TargetRate_bps;
} else {
channelRate_bps = MStargetRates_bps[ n ];
if( n == 0 && MStargetRates_bps[ 1 ] > 0 ) {
useCBR = 0;
/* Give mid up to 1/2 of the max bits for that frame */
maxBits -= encControl->maxBits / ( tot_blocks * 2 );
}
}
if( channelRate_bps > 0 ) {
opus_int condCoding;
silk_control_SNR( &psEnc->state_Fxx[ n ].sCmn, channelRate_bps );
/* Use independent coding if no previous frame available */
if( psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded - n <= 0 ) {
condCoding = CODE_INDEPENDENTLY;
} else if( n > 0 && psEnc->prev_decode_only_middle ) {
/* If we skipped a side frame in this packet, we don't
need LTP scaling; the LTP state is well-defined. */
condCoding = CODE_INDEPENDENTLY_NO_LTP_SCALING;
} else {
condCoding = CODE_CONDITIONALLY;
}
if( ( ret = silk_encode_frame_Fxx( &psEnc->state_Fxx[ n ], nBytesOut, psRangeEnc, condCoding, maxBits, useCBR ) ) != 0 ) {
silk_assert( 0 );
}
}
psEnc->state_Fxx[ n ].sCmn.controlled_since_last_payload = 0;
psEnc->state_Fxx[ n ].sCmn.inputBufIx = 0;
psEnc->state_Fxx[ n ].sCmn.nFramesEncoded++;
}
psEnc->prev_decode_only_middle = psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded - 1 ];
/* Insert VAD and FEC flags at beginning of bitstream */
if( *nBytesOut > 0 && psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded == psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket) {
flags = 0;
for( n = 0; n < encControl->nChannelsInternal; n++ ) {
for( i = 0; i < psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket; i++ ) {
flags = silk_LSHIFT( flags, 1 );
flags |= psEnc->state_Fxx[ n ].sCmn.VAD_flags[ i ];
}
flags = silk_LSHIFT( flags, 1 );
flags |= psEnc->state_Fxx[ n ].sCmn.LBRR_flag;
}
if( !prefillFlag ) {
ec_enc_patch_initial_bits( psRangeEnc, flags, ( psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket + 1 ) * encControl->nChannelsInternal );
}
/* Return zero bytes if all channels DTXed */
if( psEnc->state_Fxx[ 0 ].sCmn.inDTX && ( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 1 ].sCmn.inDTX ) ) {
*nBytesOut = 0;
}
psEnc->nBitsExceeded += *nBytesOut * 8;
psEnc->nBitsExceeded -= silk_DIV32_16( silk_MUL( encControl->bitRate, encControl->payloadSize_ms ), 1000 );
psEnc->nBitsExceeded = silk_LIMIT( psEnc->nBitsExceeded, 0, 10000 );
/* Update flag indicating if bandwidth switching is allowed */
speech_act_thr_for_switch_Q8 = (opus_int) silk_SMLAWB( SILK_FIX_CONST( SPEECH_ACTIVITY_DTX_THRES, 8 ),
SILK_FIX_CONST( ( 1 - SPEECH_ACTIVITY_DTX_THRES ) / MAX_BANDWIDTH_SWITCH_DELAY_MS, 16 + 8 ), psEnc->timeSinceSwitchAllowed_ms );
if( psEnc->state_Fxx[ 0 ].sCmn.speech_activity_Q8 < speech_act_thr_for_switch_Q8 ) {
psEnc->allowBandwidthSwitch = 1;
psEnc->timeSinceSwitchAllowed_ms = 0;
} else {
psEnc->allowBandwidthSwitch = 0;
psEnc->timeSinceSwitchAllowed_ms += encControl->payloadSize_ms;
}
}
if( nSamplesIn == 0 ) {
break;
}
} else {
break;
}
curr_block++;
}
psEnc->nPrevChannelsInternal = encControl->nChannelsInternal;
encControl->allowBandwidthSwitch = psEnc->allowBandwidthSwitch;
encControl->inWBmodeWithoutVariableLP = psEnc->state_Fxx[ 0 ].sCmn.fs_kHz == 16 && psEnc->state_Fxx[ 0 ].sCmn.sLP.mode == 0;
encControl->internalSampleRate = silk_SMULBB( psEnc->state_Fxx[ 0 ].sCmn.fs_kHz, 1000 );
encControl->stereoWidth_Q14 = encControl->toMono ? 0 : psEnc->sStereo.smth_width_Q14;
if( prefillFlag ) {
encControl->payloadSize_ms = tmp_payloadSize_ms;
encControl->complexity = tmp_complexity;
for( n = 0; n < encControl->nChannelsInternal; n++ ) {
psEnc->state_Fxx[ n ].sCmn.controlled_since_last_payload = 0;
psEnc->state_Fxx[ n ].sCmn.prefillFlag = 0;
}
}
RESTORE_STACK;
return ret;
}