void silk_find_pitch_lags_FLP(
silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */
silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */
silk_float res[], /* O Residual */
const silk_float x[], /* I Speech signal */
int arch /* I Run-time architecture */
)
{
opus_int buf_len;
silk_float thrhld, res_nrg;
const silk_float *x_buf_ptr, *x_buf;
silk_float auto_corr[ MAX_FIND_PITCH_LPC_ORDER + 1 ];
silk_float A[ MAX_FIND_PITCH_LPC_ORDER ];
silk_float refl_coef[ MAX_FIND_PITCH_LPC_ORDER ];
silk_float Wsig[ FIND_PITCH_LPC_WIN_MAX ];
silk_float *Wsig_ptr;
/******************************************/
/* Set up buffer lengths etc based on Fs */
/******************************************/
buf_len = psEnc->sCmn.la_pitch + psEnc->sCmn.frame_length + psEnc->sCmn.ltp_mem_length;
/* Safety check */
silk_assert( buf_len >= psEnc->sCmn.pitch_LPC_win_length );
x_buf = x - psEnc->sCmn.ltp_mem_length;
/******************************************/
/* Estimate LPC AR coeficients */
/******************************************/
/* Calculate windowed signal */
/* First LA_LTP samples */
x_buf_ptr = x_buf + buf_len - psEnc->sCmn.pitch_LPC_win_length;
Wsig_ptr = Wsig;
silk_apply_sine_window_FLP( Wsig_ptr, x_buf_ptr, 1, psEnc->sCmn.la_pitch );
/* Middle non-windowed samples */
Wsig_ptr += psEnc->sCmn.la_pitch;
x_buf_ptr += psEnc->sCmn.la_pitch;
silk_memcpy( Wsig_ptr, x_buf_ptr, ( psEnc->sCmn.pitch_LPC_win_length - ( psEnc->sCmn.la_pitch << 1 ) ) * sizeof( silk_float ) );
/* Last LA_LTP samples */
Wsig_ptr += psEnc->sCmn.pitch_LPC_win_length - ( psEnc->sCmn.la_pitch << 1 );
x_buf_ptr += psEnc->sCmn.pitch_LPC_win_length - ( psEnc->sCmn.la_pitch << 1 );
silk_apply_sine_window_FLP( Wsig_ptr, x_buf_ptr, 2, psEnc->sCmn.la_pitch );
/* Calculate autocorrelation sequence */
silk_autocorrelation_FLP( auto_corr, Wsig, psEnc->sCmn.pitch_LPC_win_length, psEnc->sCmn.pitchEstimationLPCOrder + 1 );
/* Add white noise, as a fraction of the energy */
auto_corr[ 0 ] += auto_corr[ 0 ] * FIND_PITCH_WHITE_NOISE_FRACTION + 1;
/* Calculate the reflection coefficients using Schur */
res_nrg = silk_schur_FLP( refl_coef, auto_corr, psEnc->sCmn.pitchEstimationLPCOrder );
/* Prediction gain */
psEncCtrl->predGain = auto_corr[ 0 ] / silk_max_float( res_nrg, 1.0f );
/* Convert reflection coefficients to prediction coefficients */
silk_k2a_FLP( A, refl_coef, psEnc->sCmn.pitchEstimationLPCOrder );
/* Bandwidth expansion */
silk_bwexpander_FLP( A, psEnc->sCmn.pitchEstimationLPCOrder, FIND_PITCH_BANDWIDTH_EXPANSION );
/*****************************************/
/* LPC analysis filtering */
/*****************************************/
silk_LPC_analysis_filter_FLP( res, A, x_buf, buf_len, psEnc->sCmn.pitchEstimationLPCOrder );
if( psEnc->sCmn.indices.signalType != TYPE_NO_VOICE_ACTIVITY && psEnc->sCmn.first_frame_after_reset == 0 ) {
/* Threshold for pitch estimator */
thrhld = 0.6f;
thrhld -= 0.004f * psEnc->sCmn.pitchEstimationLPCOrder;
thrhld -= 0.1f * psEnc->sCmn.speech_activity_Q8 * ( 1.0f / 256.0f );
thrhld -= 0.15f * (psEnc->sCmn.prevSignalType >> 1);
thrhld -= 0.1f * psEnc->sCmn.input_tilt_Q15 * ( 1.0f / 32768.0f );
/*****************************************/
/* Call Pitch estimator */
/*****************************************/
if( silk_pitch_analysis_core_FLP( res, psEncCtrl->pitchL, &psEnc->sCmn.indices.lagIndex,
&psEnc->sCmn.indices.contourIndex, &psEnc->LTPCorr, psEnc->sCmn.prevLag, psEnc->sCmn.pitchEstimationThreshold_Q16 / 65536.0f,
thrhld, psEnc->sCmn.fs_kHz, psEnc->sCmn.pitchEstimationComplexity, psEnc->sCmn.nb_subfr, arch ) == 0 )
{
psEnc->sCmn.indices.signalType = TYPE_VOICED;
} else {
psEnc->sCmn.indices.signalType = TYPE_UNVOICED;
}
} else {
/* Compute noise shaping coefficients and initial gain values */
void silk_noise_shape_analysis_FLP(
silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */
silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */
const silk_float *pitch_res, /* I LPC residual from pitch analysis */
const silk_float *x /* I Input signal [frame_length + la_shape] */
)
{
silk_shape_state_FLP *psShapeSt = &psEnc->sShape;
opus_int k, nSamples;
silk_float SNR_adj_dB, HarmBoost, HarmShapeGain, Tilt;
silk_float nrg, pre_nrg, log_energy, log_energy_prev, energy_variation;
silk_float delta, BWExp1, BWExp2, gain_mult, gain_add, strength, b, warping;
silk_float x_windowed[ SHAPE_LPC_WIN_MAX ];
silk_float auto_corr[ MAX_SHAPE_LPC_ORDER + 1 ];
const silk_float *x_ptr, *pitch_res_ptr;
/* Point to start of first LPC analysis block */
x_ptr = x - psEnc->sCmn.la_shape;
/****************/
/* GAIN CONTROL */
/****************/
SNR_adj_dB = psEnc->sCmn.SNR_dB_Q7 * ( 1 / 128.0f );
/* Input quality is the average of the quality in the lowest two VAD bands */
psEncCtrl->input_quality = 0.5f * ( psEnc->sCmn.input_quality_bands_Q15[ 0 ] + psEnc->sCmn.input_quality_bands_Q15[ 1 ] ) * ( 1.0f / 32768.0f );
/* Coding quality level, between 0.0 and 1.0 */
psEncCtrl->coding_quality = silk_sigmoid( 0.25f * ( SNR_adj_dB - 20.0f ) );
if( psEnc->sCmn.useCBR == 0 ) {
/* Reduce coding SNR during low speech activity */
b = 1.0f - psEnc->sCmn.speech_activity_Q8 * ( 1.0f / 256.0f );
SNR_adj_dB -= BG_SNR_DECR_dB * psEncCtrl->coding_quality * ( 0.5f + 0.5f * psEncCtrl->input_quality ) * b * b;
}
if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) {
/* Reduce gains for periodic signals */
SNR_adj_dB += HARM_SNR_INCR_dB * psEnc->LTPCorr;
} else {
/* For unvoiced signals and low-quality input, adjust the quality slower than SNR_dB setting */
SNR_adj_dB += ( -0.4f * psEnc->sCmn.SNR_dB_Q7 * ( 1 / 128.0f ) + 6.0f ) * ( 1.0f - psEncCtrl->input_quality );
}
/*************************/
/* SPARSENESS PROCESSING */
/*************************/
/* Set quantizer offset */
if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) {
/* Initially set to 0; may be overruled in process_gains(..) */
psEnc->sCmn.indices.quantOffsetType = 0;
psEncCtrl->sparseness = 0.0f;
} else {
/* Sparseness measure, based on relative fluctuations of energy per 2 milliseconds */
nSamples = 2 * psEnc->sCmn.fs_kHz;
energy_variation = 0.0f;
log_energy_prev = 0.0f;
pitch_res_ptr = pitch_res;
for( k = 0; k < silk_SMULBB( SUB_FRAME_LENGTH_MS, psEnc->sCmn.nb_subfr ) / 2; k++ ) {
nrg = ( silk_float )nSamples + ( silk_float )silk_energy_FLP( pitch_res_ptr, nSamples );
log_energy = silk_log2( nrg );
if( k > 0 ) {
energy_variation += silk_abs_float( log_energy - log_energy_prev );
}
log_energy_prev = log_energy;
pitch_res_ptr += nSamples;
}
psEncCtrl->sparseness = silk_sigmoid( 0.4f * ( energy_variation - 5.0f ) );
/* Set quantization offset depending on sparseness measure */
if( psEncCtrl->sparseness > SPARSENESS_THRESHOLD_QNT_OFFSET ) {
psEnc->sCmn.indices.quantOffsetType = 0;
} else {
psEnc->sCmn.indices.quantOffsetType = 1;
}
/* Increase coding SNR for sparse signals */
SNR_adj_dB += SPARSE_SNR_INCR_dB * ( psEncCtrl->sparseness - 0.5f );
}
/*******************************/
/* Control bandwidth expansion */
/*******************************/
/* More BWE for signals with high prediction gain */
strength = FIND_PITCH_WHITE_NOISE_FRACTION * psEncCtrl->predGain; /* between 0.0 and 1.0 */
BWExp1 = BWExp2 = BANDWIDTH_EXPANSION / ( 1.0f + strength * strength );
delta = LOW_RATE_BANDWIDTH_EXPANSION_DELTA * ( 1.0f - 0.75f * psEncCtrl->coding_quality );
BWExp1 -= delta;
BWExp2 += delta;
/* BWExp1 will be applied after BWExp2, so make it relative */
BWExp1 /= BWExp2;
if( psEnc->sCmn.warping_Q16 > 0 ) {
/* Slightly more warping in analysis will move quantization noise up in frequency, where it's better masked */
warping = (silk_float)psEnc->sCmn.warping_Q16 / 65536.0f + 0.01f * psEncCtrl->coding_quality;
} else {
warping = 0.0f;
}
/********************************************/
/* Compute noise shaping AR coefs and gains */
/********************************************/
for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) {
/* Apply window: sine slope followed by flat part followed by cosine slope */
opus_int shift, slope_part, flat_part;
flat_part = psEnc->sCmn.fs_kHz * 3;
slope_part = ( psEnc->sCmn.shapeWinLength - flat_part ) / 2;
silk_apply_sine_window_FLP( x_windowed, x_ptr, 1, slope_part );
shift = slope_part;
silk_memcpy( x_windowed + shift, x_ptr + shift, flat_part * sizeof(silk_float) );
shift += flat_part;
silk_apply_sine_window_FLP( x_windowed + shift, x_ptr + shift, 2, slope_part );
/* Update pointer: next LPC analysis block */
x_ptr += psEnc->sCmn.subfr_length;
if( psEnc->sCmn.warping_Q16 > 0 ) {
/* Calculate warped auto correlation */
silk_warped_autocorrelation_FLP( auto_corr, x_windowed, warping,
psEnc->sCmn.shapeWinLength, psEnc->sCmn.shapingLPCOrder );
} else {
/* Calculate regular auto correlation */
silk_autocorrelation_FLP( auto_corr, x_windowed, psEnc->sCmn.shapeWinLength, psEnc->sCmn.shapingLPCOrder + 1 );
}
/* Add white noise, as a fraction of energy */
auto_corr[ 0 ] += auto_corr[ 0 ] * SHAPE_WHITE_NOISE_FRACTION;
/* Convert correlations to prediction coefficients, and compute residual energy */
nrg = silk_levinsondurbin_FLP( &psEncCtrl->AR2[ k * MAX_SHAPE_LPC_ORDER ], auto_corr, psEnc->sCmn.shapingLPCOrder );
psEncCtrl->Gains[ k ] = ( silk_float )sqrt( nrg );
if( psEnc->sCmn.warping_Q16 > 0 ) {
/* Adjust gain for warping */
psEncCtrl->Gains[ k ] *= warped_gain( &psEncCtrl->AR2[ k * MAX_SHAPE_LPC_ORDER ], warping, psEnc->sCmn.shapingLPCOrder );
}
/* Bandwidth expansion for synthesis filter shaping */
silk_bwexpander_FLP( &psEncCtrl->AR2[ k * MAX_SHAPE_LPC_ORDER ], psEnc->sCmn.shapingLPCOrder, BWExp2 );
/* Compute noise shaping filter coefficients */
silk_memcpy(
&psEncCtrl->AR1[ k * MAX_SHAPE_LPC_ORDER ],
&psEncCtrl->AR2[ k * MAX_SHAPE_LPC_ORDER ],
psEnc->sCmn.shapingLPCOrder * sizeof( silk_float ) );
/* Bandwidth expansion for analysis filter shaping */
silk_bwexpander_FLP( &psEncCtrl->AR1[ k * MAX_SHAPE_LPC_ORDER ], psEnc->sCmn.shapingLPCOrder, BWExp1 );
/* Ratio of prediction gains, in energy domain */
pre_nrg = silk_LPC_inverse_pred_gain_FLP( &psEncCtrl->AR2[ k * MAX_SHAPE_LPC_ORDER ], psEnc->sCmn.shapingLPCOrder );
nrg = silk_LPC_inverse_pred_gain_FLP( &psEncCtrl->AR1[ k * MAX_SHAPE_LPC_ORDER ], psEnc->sCmn.shapingLPCOrder );
psEncCtrl->GainsPre[ k ] = 1.0f - 0.7f * ( 1.0f - pre_nrg / nrg );
/* Convert to monic warped prediction coefficients and limit absolute values */
warped_true2monic_coefs( &psEncCtrl->AR2[ k * MAX_SHAPE_LPC_ORDER ], &psEncCtrl->AR1[ k * MAX_SHAPE_LPC_ORDER ],
warping, 3.999f, psEnc->sCmn.shapingLPCOrder );
}
/*****************/
/* Gain tweaking */
/*****************/
/* Increase gains during low speech activity */
gain_mult = (silk_float)pow( 2.0f, -0.16f * SNR_adj_dB );
gain_add = (silk_float)pow( 2.0f, 0.16f * MIN_QGAIN_DB );
for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) {
psEncCtrl->Gains[ k ] *= gain_mult;
psEncCtrl->Gains[ k ] += gain_add;
}
gain_mult = 1.0f + INPUT_TILT + psEncCtrl->coding_quality * HIGH_RATE_INPUT_TILT;
for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) {
psEncCtrl->GainsPre[ k ] *= gain_mult;
}
/************************************************/
/* Control low-frequency shaping and noise tilt */
/************************************************/
/* Less low frequency shaping for noisy inputs */
strength = LOW_FREQ_SHAPING * ( 1.0f + LOW_QUALITY_LOW_FREQ_SHAPING_DECR * ( psEnc->sCmn.input_quality_bands_Q15[ 0 ] * ( 1.0f / 32768.0f ) - 1.0f ) );
strength *= psEnc->sCmn.speech_activity_Q8 * ( 1.0f / 256.0f );
if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) {
/* Reduce low frequencies quantization noise for periodic signals, depending on pitch lag */
/*f = 400; freqz([1, -0.98 + 2e-4 * f], [1, -0.97 + 7e-4 * f], 2^12, Fs); axis([0, 1000, -10, 1])*/
for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) {
b = 0.2f / psEnc->sCmn.fs_kHz + 3.0f / psEncCtrl->pitchL[ k ];
psEncCtrl->LF_MA_shp[ k ] = -1.0f + b;
psEncCtrl->LF_AR_shp[ k ] = 1.0f - b - b * strength;
}
Tilt = - HP_NOISE_COEF -
(1 - HP_NOISE_COEF) * HARM_HP_NOISE_COEF * psEnc->sCmn.speech_activity_Q8 * ( 1.0f / 256.0f );
} else {
b = 1.3f / psEnc->sCmn.fs_kHz;
psEncCtrl->LF_MA_shp[ 0 ] = -1.0f + b;
psEncCtrl->LF_AR_shp[ 0 ] = 1.0f - b - b * strength * 0.6f;
for( k = 1; k < psEnc->sCmn.nb_subfr; k++ ) {
psEncCtrl->LF_MA_shp[ k ] = psEncCtrl->LF_MA_shp[ 0 ];
psEncCtrl->LF_AR_shp[ k ] = psEncCtrl->LF_AR_shp[ 0 ];
}
Tilt = -HP_NOISE_COEF;
}
/****************************/
/* HARMONIC SHAPING CONTROL */
/****************************/
/* Control boosting of harmonic frequencies */
HarmBoost = LOW_RATE_HARMONIC_BOOST * ( 1.0f - psEncCtrl->coding_quality ) * psEnc->LTPCorr;
/* More harmonic boost for noisy input signals */
HarmBoost += LOW_INPUT_QUALITY_HARMONIC_BOOST * ( 1.0f - psEncCtrl->input_quality );
if( USE_HARM_SHAPING && psEnc->sCmn.indices.signalType == TYPE_VOICED ) {
/* Harmonic noise shaping */
HarmShapeGain = HARMONIC_SHAPING;
/* More harmonic noise shaping for high bitrates or noisy input */
HarmShapeGain += HIGH_RATE_OR_LOW_QUALITY_HARMONIC_SHAPING *
( 1.0f - ( 1.0f - psEncCtrl->coding_quality ) * psEncCtrl->input_quality );
/* Less harmonic noise shaping for less periodic signals */
HarmShapeGain *= ( silk_float )sqrt( psEnc->LTPCorr );
} else {
HarmShapeGain = 0.0f;
}
/*************************/
/* Smooth over subframes */
/*************************/
for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) {
psShapeSt->HarmBoost_smth += SUBFR_SMTH_COEF * ( HarmBoost - psShapeSt->HarmBoost_smth );
psEncCtrl->HarmBoost[ k ] = psShapeSt->HarmBoost_smth;
psShapeSt->HarmShapeGain_smth += SUBFR_SMTH_COEF * ( HarmShapeGain - psShapeSt->HarmShapeGain_smth );
psEncCtrl->HarmShapeGain[ k ] = psShapeSt->HarmShapeGain_smth;
psShapeSt->Tilt_smth += SUBFR_SMTH_COEF * ( Tilt - psShapeSt->Tilt_smth );
psEncCtrl->Tilt[ k ] = psShapeSt->Tilt_smth;
}
}
