本文整理汇总了C++中silk_SMULBB函数的典型用法代码示例。如果您正苦于以下问题：C++ silk_SMULBB函数的具体用法？C++ silk_SMULBB怎么用？C++ silk_SMULBB使用的例子？那么恭喜您, 这里精选的函数代码示例或许可以为您提供帮助。
在下文中一共展示了silk_SMULBB函数的20个代码示例，这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞，您的评价将有助于我们的系统推荐出更棒的C++代码示例。
示例1: silk_sigm_Q15
opus_int silk_sigm_Q15(
opus_int in_Q5 /* I */
)
{
opus_int ind;
if( in_Q5 < 0 ) {
/* Negative input */
in_Q5 = -in_Q5;
if( in_Q5 >= 6 * 32 ) {
return 0; /* Clip */
} else {
/* Linear interpolation of look up table */
ind = silk_RSHIFT( in_Q5, 5 );
return( sigm_LUT_neg_Q15[ ind ] - silk_SMULBB( sigm_LUT_slope_Q10[ ind ], in_Q5 & 0x1F ) );
}
} else {
/* Positive input */
if( in_Q5 >= 6 * 32 ) {
return 32767; /* clip */
} else {
/* Linear interpolation of look up table */
ind = silk_RSHIFT( in_Q5, 5 );
return( sigm_LUT_pos_Q15[ ind ] + silk_SMULBB( sigm_LUT_slope_Q10[ ind ], in_Q5 & 0x1F ) );
}
}
}
开发者ID:62gs8ha，项目名称:batphone，代码行数:27，代码来源:sigm_Q15.c
示例2: silk_int16_array_maxabs
/* Function that returns the maximum absolut value of the input vector */
opus_int16 silk_int16_array_maxabs( /* O Maximum absolute value, max: 2^15-1 */
const opus_int16 *vec, /* I Input vector [len] */
const opus_int32 len /* I Length of input vector */
)
{
opus_int32 max = 0, i, lvl = 0, ind;
if( len == 0 ) return 0;
ind = len - 1;
max = silk_SMULBB( vec[ ind ], vec[ ind ] );
for( i = len - 2; i >= 0; i-- ) {
lvl = silk_SMULBB( vec[ i ], vec[ i ] );
if( lvl > max ) {
max = lvl;
ind = i;
}
}
/* Do not return 32768, as it will not fit in an int16 so may lead to problems later on */
if( max >= 1073676289 ) { /* (2^15-1)^2 = 1073676289 */
return( silk_int16_MAX );
} else {
if( vec[ ind ] < 0 ) {
return( -vec[ ind ] );
} else {
return( vec[ ind ] );
}
}
}
开发者ID:oneman，项目名称:opus-oneman，代码行数:30，代码来源:vector_ops_FIX.c
示例3: silk_NLSF_unpack
/* Unpack predictor values and indices for entropy coding tables */
void silk_NLSF_unpack(int16_t ec_ix[], /* O Indices to entropy tables [ LPC_ORDER ] */
uint8_t pred_Q8[], /* O LSF predictor [ LPC_ORDER ] */
const silk_NLSF_CB_struct * psNLSF_CB, /* I Codebook object */
const int CB1_index /* I Index of vector in first LSF codebook */
)
{
int i;
uint8_t entry;
const uint8_t *ec_sel_ptr;
ec_sel_ptr = &psNLSF_CB->ec_sel[CB1_index * psNLSF_CB->order / 2];
for (i = 0; i < psNLSF_CB->order; i += 2) {
entry = *ec_sel_ptr++;
ec_ix[i] =
silk_SMULBB(silk_RSHIFT(entry, 1) & 7,
2 * NLSF_QUANT_MAX_AMPLITUDE + 1);
pred_Q8[i] =
psNLSF_CB->pred_Q8[i +
(entry & 1) * (psNLSF_CB->order - 1)];
ec_ix[i + 1] =
silk_SMULBB(silk_RSHIFT(entry, 5) & 7,
2 * NLSF_QUANT_MAX_AMPLITUDE + 1);
pred_Q8[i + 1] =
psNLSF_CB->pred_Q8[i +
(silk_RSHIFT(entry, 4) & 1) *
(psNLSF_CB->order - 1) + 1];
}
}
开发者ID:CEPBEP，项目名称:onion-phone，代码行数:29，代码来源:NLSF_unpack.c
示例4: silk_sum_sqr_shift
/* of int16s to make it fit in an int32 */
void silk_sum_sqr_shift(int32_t * energy, /* O Energy of x, after shifting to the right */
int * shift, /* O Number of bits right shift applied to energy */
const int16_t * x, /* I Input vector */
int len /* I Length of input vector */
)
{
int i, shft;
int32_t nrg_tmp, nrg;
nrg = 0;
shft = 0;
len--;
for (i = 0; i < len; i += 2) {
nrg = silk_SMLABB_ovflw(nrg, x[i], x[i]);
nrg = silk_SMLABB_ovflw(nrg, x[i + 1], x[i + 1]);
if (nrg < 0) {
/* Scale down */
nrg =
(int32_t) silk_RSHIFT_uint((uint32_t) nrg, 2);
shft = 2;
break;
}
}
for (; i < len; i += 2) {
nrg_tmp = silk_SMULBB(x[i], x[i]);
nrg_tmp = silk_SMLABB_ovflw(nrg_tmp, x[i + 1], x[i + 1]);
nrg =
(int32_t) silk_ADD_RSHIFT_uint(nrg,
(uint32_t) nrg_tmp,
shft);
if (nrg < 0) {
/* Scale down */
nrg =
(int32_t) silk_RSHIFT_uint((uint32_t) nrg, 2);
shft += 2;
}
}
if (i == len) {
/* One sample left to process */
nrg_tmp = silk_SMULBB(x[i], x[i]);
nrg = (int32_t) silk_ADD_RSHIFT_uint(nrg, nrg_tmp, shft);
}
/* Make sure to have at least one extra leading zero (two leading zeros in total) */
if (nrg & 0xC0000000) {
nrg = silk_RSHIFT_uint((uint32_t) nrg, 2);
shft += 2;
}
/* Output arguments */
*shift = shft;
*energy = nrg;
}
开发者ID:CEPBEP，项目名称:onion-phone，代码行数:54，代码来源:sum_sqr_shift.c
示例5: silk_stereo_MS_to_LR
/* 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 );
}
}
开发者ID:261117370，项目名称:godot，代码行数:52，代码来源:stereo_MS_to_LR.c
示例6: silk_QueryEncoder
static opus_int silk_QueryEncoder( /* O Returns error code */
const void *encState, /* I State */
silk_EncControlStruct *encStatus /* O Encoder Status */
)
{
opus_int ret = SILK_NO_ERROR;
silk_encoder_state_Fxx *state_Fxx;
silk_encoder *psEnc = (silk_encoder *)encState;
state_Fxx = psEnc->state_Fxx;
encStatus->nChannelsAPI = psEnc->nChannelsAPI;
encStatus->nChannelsInternal = psEnc->nChannelsInternal;
encStatus->API_sampleRate = state_Fxx[ 0 ].sCmn.API_fs_Hz;
encStatus->maxInternalSampleRate = state_Fxx[ 0 ].sCmn.maxInternal_fs_Hz;
encStatus->minInternalSampleRate = state_Fxx[ 0 ].sCmn.minInternal_fs_Hz;
encStatus->desiredInternalSampleRate = state_Fxx[ 0 ].sCmn.desiredInternal_fs_Hz;
encStatus->payloadSize_ms = state_Fxx[ 0 ].sCmn.PacketSize_ms;
encStatus->bitRate = state_Fxx[ 0 ].sCmn.TargetRate_bps;
encStatus->packetLossPercentage = state_Fxx[ 0 ].sCmn.PacketLoss_perc;
encStatus->complexity = state_Fxx[ 0 ].sCmn.Complexity;
encStatus->useInBandFEC = state_Fxx[ 0 ].sCmn.useInBandFEC;
encStatus->useDTX = state_Fxx[ 0 ].sCmn.useDTX;
encStatus->useCBR = state_Fxx[ 0 ].sCmn.useCBR;
encStatus->internalSampleRate = silk_SMULBB( state_Fxx[ 0 ].sCmn.fs_kHz, 1000 );
encStatus->allowBandwidthSwitch = state_Fxx[ 0 ].sCmn.allow_bandwidth_switch;
encStatus->inWBmodeWithoutVariableLP = state_Fxx[ 0 ].sCmn.fs_kHz == 16 && state_Fxx[ 0 ].sCmn.sLP.mode == 0;
return ret;
}
开发者ID:kode54，项目名称:Cog，代码行数:30，代码来源:enc_API.c
示例7: silk_NLSF_VQ
/* Compute quantization errors for an LPC_order element input vector for a VQ codebook */
void silk_NLSF_VQ(
opus_int32 err_Q26[], /* O Quantization errors [K] */
const opus_int16 in_Q15[], /* I Input vectors to be quantized [LPC_order] */
const opus_uint8 pCB_Q8[], /* I Codebook vectors [K*LPC_order] */
const opus_int K, /* I Number of codebook vectors */
const opus_int LPC_order /* I Number of LPCs */
)
{
opus_int i, m;
opus_int32 diff_Q15, sum_error_Q30, sum_error_Q26;
silk_assert( LPC_order <= 16 );
silk_assert( ( LPC_order & 1 ) == 0 );
/* Loop over codebook */
for( i = 0; i < K; i++ ) {
sum_error_Q26 = 0;
for( m = 0; m < LPC_order; m += 2 ) {
/* Compute weighted squared quantization error for index m */
diff_Q15 = silk_SUB_LSHIFT32( in_Q15[ m ], (opus_int32)*pCB_Q8++, 7 ); /* range: [ -32767 : 32767 ]*/
sum_error_Q30 = silk_SMULBB( diff_Q15, diff_Q15 );
/* Compute weighted squared quantization error for index m + 1 */
diff_Q15 = silk_SUB_LSHIFT32( in_Q15[m + 1], (opus_int32)*pCB_Q8++, 7 ); /* range: [ -32767 : 32767 ]*/
sum_error_Q30 = silk_SMLABB( sum_error_Q30, diff_Q15, diff_Q15 );
sum_error_Q26 = silk_ADD_RSHIFT32( sum_error_Q26, sum_error_Q30, 4 );
silk_assert( sum_error_Q26 >= 0 );
silk_assert( sum_error_Q30 >= 0 );
}
err_Q26[ i ] = sum_error_Q26;
}
}
开发者ID:wonktnodi，项目名称:webrtc_port，代码行数:35，代码来源:NLSF_VQ.c
示例8: silk_corrVector_FIX
/* Calculates correlation vector X'*t */
void silk_corrVector_FIX(
const opus_int16 *x, /* I x vector [L + order - 1] used to form data matrix X */
const opus_int16 *t, /* I Target vector [L] */
const opus_int L, /* I Length of vectors */
const opus_int order, /* I Max lag for correlation */
opus_int32 *Xt, /* O Pointer to X'*t correlation vector [order] */
const opus_int rshifts /* I Right shifts of correlations */
)
{
opus_int lag, i;
const opus_int16 *ptr1, *ptr2;
opus_int32 inner_prod;
ptr1 = &x[ order - 1 ]; /* Points to first sample of column 0 of X: X[:,0] */
ptr2 = t;
/* Calculate X'*t */
if( rshifts > 0 ) {
/* Right shifting used */
for( lag = 0; lag < order; lag++ ) {
inner_prod = 0;
for( i = 0; i < L; i++ ) {
inner_prod += silk_RSHIFT32( silk_SMULBB( ptr1[ i ], ptr2[i] ), rshifts );
}
Xt[ lag ] = inner_prod; /* X[:,lag]'*t */
ptr1--; /* Go to next column of X */
}
} else {
silk_assert( rshifts == 0 );
for( lag = 0; lag < order; lag++ ) {
Xt[ lag ] = silk_inner_prod_aligned( ptr1, ptr2, L ); /* X[:,lag]'*t */
ptr1--; /* Go to next column of X */
}
}
}
开发者ID:edd83，项目名称:Skype-like，代码行数:35，代码来源:corrMatrix_FIX.c
示例9: silk_HP_variable_cutoff
/* 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));
}
}
开发者ID:CEPBEP，项目名称:onion-phone，代码行数:61，代码来源:HP_variable_cutoff.c
示例10: silk_encode_signs
/* Encodes signs of excitation */
void silk_encode_signs(
ec_enc *psRangeEnc, /* I/O Compressor data structure */
const opus_int8 pulses[], /* I pulse signal */
opus_int length, /* I length of input */
const opus_int signalType, /* I Signal type */
const opus_int quantOffsetType, /* I Quantization offset type */
const opus_int sum_pulses[ MAX_NB_SHELL_BLOCKS ] /* I Sum of absolute pulses per block */
)
{
opus_int i, j, p;
opus_uint8 icdf[ 2 ];
const opus_int8 *q_ptr;
const opus_uint8 *icdf_ptr;
icdf[ 1 ] = 0;
q_ptr = pulses;
i = silk_SMULBB( 7, silk_ADD_LSHIFT( quantOffsetType, signalType, 1 ) );
icdf_ptr = &silk_sign_iCDF[ i ];
length = silk_RSHIFT( length + SHELL_CODEC_FRAME_LENGTH/2, LOG2_SHELL_CODEC_FRAME_LENGTH );
for( i = 0; i < length; i++ ) {
p = sum_pulses[ i ];
if( p > 0 ) {
icdf[ 0 ] = icdf_ptr[ silk_min( p & 0x1F, 6 ) ];
for( j = 0; j < SHELL_CODEC_FRAME_LENGTH; j++ ) {
if( q_ptr[ j ] != 0 ) {
ec_enc_icdf( psRangeEnc, silk_enc_map( q_ptr[ j ]), icdf, 8 );
}
}
}
q_ptr += SHELL_CODEC_FRAME_LENGTH;
}
}
开发者ID:4nykey，项目名称:rockbox，代码行数:33，代码来源:code_signs.c
示例11: silk_SMULWB
static OPUS_INLINE opus_int16 *silk_resampler_private_IIR_FIR_INTERPOL(
opus_int16 *out,
opus_int16 *buf,
opus_int32 max_index_Q16,
opus_int32 index_increment_Q16
)
{
opus_int32 index_Q16, res_Q15;
opus_int16 *buf_ptr;
opus_int32 table_index;
/* Interpolate upsampled signal and store in output array */
for( index_Q16 = 0; index_Q16 < max_index_Q16; index_Q16 += index_increment_Q16 ) {
table_index = silk_SMULWB( index_Q16 & 0xFFFF, 12 );
buf_ptr = &buf[ index_Q16 >> 16 ];
res_Q15 = silk_SMULBB( buf_ptr[ 0 ], silk_resampler_frac_FIR_12[ table_index ][ 0 ] );
res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 1 ], silk_resampler_frac_FIR_12[ table_index ][ 1 ] );
res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 2 ], silk_resampler_frac_FIR_12[ table_index ][ 2 ] );
res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 3 ], silk_resampler_frac_FIR_12[ table_index ][ 3 ] );
res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 4 ], silk_resampler_frac_FIR_12[ 11 - table_index ][ 3 ] );
res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 5 ], silk_resampler_frac_FIR_12[ 11 - table_index ][ 2 ] );
res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 6 ], silk_resampler_frac_FIR_12[ 11 - table_index ][ 1 ] );
res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 7 ], silk_resampler_frac_FIR_12[ 11 - table_index ][ 0 ] );
*out++ = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( res_Q15, 15 ) );
}
return out;
}
开发者ID:a12n，项目名称:godot，代码行数:28，代码来源:resampler_private_IIR_FIR.c
示例12: silk_prefilt_FIX
/* Prefilter for finding Quantizer input signal */
static OPUS_INLINE void silk_prefilt_FIX(
silk_prefilter_state_FIX *P, /* I/O state */
opus_int32 st_res_Q12[], /* I short term residual signal */
opus_int32 xw_Q3[], /* O prefiltered signal */
opus_int32 HarmShapeFIRPacked_Q12, /* I Harmonic shaping coeficients */
opus_int Tilt_Q14, /* I Tilt shaping coeficient */
opus_int32 LF_shp_Q14, /* I Low-frequancy shaping coeficients */
opus_int lag, /* I Lag for harmonic shaping */
opus_int length /* I Length of signals */
) {
opus_int i, idx, LTP_shp_buf_idx;
opus_int32 n_LTP_Q12, n_Tilt_Q10, n_LF_Q10;
opus_int32 sLF_MA_shp_Q12, sLF_AR_shp_Q12;
opus_int16 *LTP_shp_buf;
/* To speed up use temp variables instead of using the struct */
LTP_shp_buf = P->sLTP_shp;
LTP_shp_buf_idx = P->sLTP_shp_buf_idx;
sLF_AR_shp_Q12 = P->sLF_AR_shp_Q12;
sLF_MA_shp_Q12 = P->sLF_MA_shp_Q12;
for (i = 0; i < length; i++) {
if (lag > 0) {
/* unrolled loop */
silk_assert(HARM_SHAPE_FIR_TAPS == 3);
idx = lag + LTP_shp_buf_idx;
n_LTP_Q12 = silk_SMULBB(LTP_shp_buf[(idx - HARM_SHAPE_FIR_TAPS / 2 - 1) & LTP_MASK],
HarmShapeFIRPacked_Q12);
n_LTP_Q12 = silk_SMLABT(n_LTP_Q12,
LTP_shp_buf[(idx - HARM_SHAPE_FIR_TAPS / 2) & LTP_MASK],
HarmShapeFIRPacked_Q12);
n_LTP_Q12 = silk_SMLABB(n_LTP_Q12,
LTP_shp_buf[(idx - HARM_SHAPE_FIR_TAPS / 2 + 1) & LTP_MASK],
HarmShapeFIRPacked_Q12);
} else {
n_LTP_Q12 = 0;
}
n_Tilt_Q10 = silk_SMULWB(sLF_AR_shp_Q12, Tilt_Q14);
n_LF_Q10 = silk_SMLAWB(silk_SMULWT(sLF_AR_shp_Q12, LF_shp_Q14), sLF_MA_shp_Q12, LF_shp_Q14);
sLF_AR_shp_Q12 = silk_SUB32(st_res_Q12[i], silk_LSHIFT(n_Tilt_Q10, 2));
sLF_MA_shp_Q12 = silk_SUB32(sLF_AR_shp_Q12, silk_LSHIFT(n_LF_Q10, 2));
LTP_shp_buf_idx = (LTP_shp_buf_idx - 1) & LTP_MASK;
LTP_shp_buf[LTP_shp_buf_idx] = (opus_int16) silk_SAT16(
silk_RSHIFT_ROUND(sLF_MA_shp_Q12, 12));
xw_Q3[i] = silk_RSHIFT_ROUND(silk_SUB32(sLF_MA_shp_Q12, n_LTP_Q12), 9);
}
/* Copy temp variable back to state */
P->sLF_AR_shp_Q12 = sLF_AR_shp_Q12;
P->sLF_MA_shp_Q12 = sLF_MA_shp_Q12;
P->sLTP_shp_buf_idx = LTP_shp_buf_idx;
}
开发者ID:KeepYoung24，项目名称:Yahala-Messenger，代码行数:57，代码来源:prefilter_FIX.c
示例13: silk_log2lin
/* Convert input to a linear scale */
opus_int32 silk_log2lin(
const opus_int32 inLog_Q7 /* I input on log scale */
)
{
opus_int32 out, frac_Q7;
if( inLog_Q7 < 0 ) {
return 0;
}
out = silk_LSHIFT( 1, silk_RSHIFT( inLog_Q7, 7 ) );
frac_Q7 = inLog_Q7 & 0x7F;
if( inLog_Q7 < 2048 ) {
/* Piece-wise parabolic approximation */
out = silk_ADD_RSHIFT32( out, silk_MUL( out, silk_SMLAWB( frac_Q7, silk_SMULBB( frac_Q7, 128 - frac_Q7 ), -174 ) ), 7 );
} else {
/* Piece-wise parabolic approximation */
out = silk_MLA( out, silk_RSHIFT( out, 7 ), silk_SMLAWB( frac_Q7, silk_SMULBB( frac_Q7, 128 - frac_Q7 ), -174 ) );
}
return out;
}
开发者ID:AdaDoom3，项目名称:AdaDoom3Libraries，代码行数:22，代码来源:log2lin.c
示例14: silk_interpolate
/* Interpolate two vectors */
void silk_interpolate(
opus_int16 xi[ MAX_LPC_ORDER ], /* O interpolated vector */
const opus_int16 x0[ MAX_LPC_ORDER ], /* I first vector */
const opus_int16 x1[ MAX_LPC_ORDER ], /* I second vector */
const opus_int ifact_Q2, /* I interp. factor, weight on 2nd vector */
const opus_int d /* I number of parameters */
)
{
opus_int i;
silk_assert( ifact_Q2 >= 0 );
silk_assert( ifact_Q2 <= 4 );
for( i = 0; i < d; i++ ) {
xi[ i ] = (opus_int16)silk_ADD_RSHIFT( x0[ i ], silk_SMULBB( x1[ i ] - x0[ i ], ifact_Q2 ), 2 );
}
}
开发者ID:03050903，项目名称:godot，代码行数:18，代码来源:interpolate.c
示例15: silk_LPC_analysis_filter
void silk_LPC_analysis_filter(
opus_int16 *out, /* O Output signal */
const opus_int16 *in, /* I Input signal */
const opus_int16 *B, /* I MA prediction coefficients, Q12 [order] */
const opus_int32 len, /* I Signal length */
const opus_int32 d /* I Filter order */
)
{
opus_int ix, j;
opus_int32 out32_Q12, out32;
const opus_int16 *in_ptr;
silk_assert( d >= 6 );
silk_assert( (d & 1) == 0 );
silk_assert( d <= len );
for( ix = d; ix < len; ix++ ) {
in_ptr = &in[ ix - 1 ];
out32_Q12 = silk_SMULBB( in_ptr[ 0 ], B[ 0 ] );
/* Allowing wrap around so that two wraps can cancel each other. The rare
cases where the result wraps around can only be triggered by invalid streams*/
out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -1 ], B[ 1 ] );
out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -2 ], B[ 2 ] );
out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -3 ], B[ 3 ] );
out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -4 ], B[ 4 ] );
out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -5 ], B[ 5 ] );
for( j = 6; j < d; j += 2 ) {
out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -j ], B[ j ] );
out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -j - 1 ], B[ j + 1 ] );
}
/* Subtract prediction */
out32_Q12 = silk_SUB32_ovflw( silk_LSHIFT( (opus_int32)in_ptr[ 1 ], 12 ), out32_Q12 );
/* Scale to Q0 */
out32 = silk_RSHIFT_ROUND( out32_Q12, 12 );
/* Saturate output */
out[ ix ] = (opus_int16)silk_SAT16( out32 );
}
/* Set first d output samples to zero */
silk_memset( out, 0, d * sizeof( opus_int16 ) );
}
开发者ID:AshishNamdev，项目名称:mozilla-central，代码行数:45，代码来源:LPC_analysis_filter.c
示例16: silk_LTP_scale_ctrl_FIX
/* Calculation of LTP state scaling */
void silk_LTP_scale_ctrl_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 */
)
{
opus_int round_loss;
if( condCoding == CODE_INDEPENDENTLY ) {
/* Only scale if first frame in packet */
round_loss = psEnc->sCmn.PacketLoss_perc + psEnc->sCmn.nFramesPerPacket;
psEnc->sCmn.indices.LTP_scaleIndex = (opus_int8)silk_LIMIT(
silk_SMULWB( silk_SMULBB( round_loss, psEncCtrl->LTPredCodGain_Q7 ), SILK_FIX_CONST( 0.1, 9 ) ), 0, 2 );
} else {
/* Default is minimum scaling */
psEnc->sCmn.indices.LTP_scaleIndex = 0;
}
psEncCtrl->LTP_scale_Q14 = silk_LTPScales_table_Q14[ psEnc->sCmn.indices.LTP_scaleIndex ];
}
开发者ID:edd83，项目名称:Skype-like，代码行数:20，代码来源:LTP_scale_ctrl_FIX.c
示例17: silk_LS_SolveLast_FIX
/* Solve L^t*x = b, where L is lower triangular with ones on the diagonal */
static OPUS_INLINE void silk_LS_SolveLast_FIX(
const opus_int32 *L_Q16, /* I Pointer to Lower Triangular Matrix */
const opus_int M, /* I Dim of Matrix equation */
const opus_int32 *b, /* I b Vector */
opus_int32 *x_Q16 /* O x Vector */
)
{
opus_int i, j;
const opus_int32 *ptr32;
opus_int32 tmp_32;
for( i = M - 1; i >= 0; i-- ) {
ptr32 = matrix_adr( L_Q16, 0, i, M );
tmp_32 = 0;
for( j = M - 1; j > i; j-- ) {
tmp_32 = silk_SMLAWW( tmp_32, ptr32[ silk_SMULBB( j, M ) ], x_Q16[ j ] );
}
x_Q16[ i ] = silk_SUB32( b[ i ], tmp_32 );
}
}
开发者ID:skylersaleh，项目名称:ArgonEngine，代码行数:21，代码来源:solve_LS_FIX.c
示例18: silk_decode_signs
/* Decodes signs of excitation */
void silk_decode_signs(
ec_dec *psRangeDec, /* I/O Compressor data structure */
opus_int pulses[], /* I/O pulse signal */
opus_int length, /* I length of input */
const opus_int signalType, /* I Signal type */
const opus_int quantOffsetType, /* I Quantization offset type */
const opus_int sum_pulses[ MAX_NB_SHELL_BLOCKS ] /* I Sum of absolute pulses per block */
)
{
opus_int i, j, p;
opus_uint8 icdf[ 2 ];
opus_int *q_ptr;
const opus_uint8 *icdf_ptr;
icdf[ 1 ] = 0;
q_ptr = pulses;
i = silk_SMULBB( 7, silk_ADD_LSHIFT( quantOffsetType, signalType, 1 ) );
icdf_ptr = &silk_sign_iCDF[ i ];
length = silk_RSHIFT( length + SHELL_CODEC_FRAME_LENGTH/2, LOG2_SHELL_CODEC_FRAME_LENGTH );
for( i = 0; i < length; i++ ) {
p = sum_pulses[ i ];
if( p > 0 ) {
icdf[ 0 ] = icdf_ptr[ silk_min( p & 0x1F, 6 ) ];
for( j = 0; j < SHELL_CODEC_FRAME_LENGTH; j++ ) {
if( q_ptr[ j ] > 0 ) {
/* attach sign */
#if 0
/* conditional implementation */
if( ec_dec_icdf( psRangeDec, icdf, 8 ) == 0 ) {
q_ptr[ j ] = -q_ptr[ j ];
}
#else
/* implementation with shift, subtraction, multiplication */
q_ptr[ j ] *= silk_dec_map( ec_dec_icdf( psRangeDec, icdf, 8 ) );
#endif
}
}
}
q_ptr += SHELL_CODEC_FRAME_LENGTH;
}
}
开发者ID:4nykey，项目名称:rockbox，代码行数:42，代码来源:code_signs.c
示例19: silk_NLSF_residual_dequant
/* Predictive dequantizer for NLSF residuals */
static OPUS_INLINE void silk_NLSF_residual_dequant( /* O Returns RD value in Q30 */
opus_int16 x_Q10[], /* O Output [ order ] */
const opus_int8 indices[], /* I Quantization indices [ order ] */
const opus_uint8 pred_coef_Q8[], /* I Backward predictor coefs [ order ] */
const opus_int quant_step_size_Q16, /* I Quantization step size */
const opus_int16 order /* I Number of input values */
)
{
opus_int i, out_Q10, pred_Q10;
out_Q10 = 0;
for( i = order-1; i >= 0; i-- ) {
pred_Q10 = silk_RSHIFT( silk_SMULBB( out_Q10, (opus_int16)pred_coef_Q8[ i ] ), 8 );
out_Q10 = silk_LSHIFT( indices[ i ], 10 );
if( out_Q10 > 0 ) {
out_Q10 = silk_SUB16( out_Q10, SILK_FIX_CONST( NLSF_QUANT_LEVEL_ADJ, 10 ) );
} else if( out_Q10 < 0 ) {
out_Q10 = silk_ADD16( out_Q10, SILK_FIX_CONST( NLSF_QUANT_LEVEL_ADJ, 10 ) );
}
out_Q10 = silk_SMLAWB( pred_Q10, (opus_int32)out_Q10, quant_step_size_Q16 );
x_Q10[ i ] = out_Q10;
}
}
开发者ID:venkatarajasekhar，项目名称:Qt，代码行数:24，代码来源:NLSF_decode.c
示例20: silk_noise_shape_analysis_FLP
/* 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;
}
//.........这里部分代码省略.........
开发者ID:a12n，项目名称:godot，代码行数:101，代码来源:noise_shape_analysis_FLP.c
注：本文中的silk_SMULBB函数示例由纯净天空整理自Github/MSDocs等源码及文档管理平台，相关代码片段筛选自各路编程大神贡献的开源项目，源码版权归原作者所有，传播和使用请参考对应项目的License；未经允许，请勿转载。 |
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