/* 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));
}
}
static inline opus_int silk_setup_LBRR(
silk_encoder_state *psEncC, /* I/O */
const opus_int32 TargetRate_bps /* I */
)
{
opus_int ret = SILK_NO_ERROR;
opus_int32 LBRR_rate_thres_bps;
psEncC->LBRR_enabled = 0;
if( psEncC->useInBandFEC && psEncC->PacketLoss_perc > 0 ) {
if( psEncC->fs_kHz == 8 ) {
LBRR_rate_thres_bps = LBRR_NB_MIN_RATE_BPS;
} else if( psEncC->fs_kHz == 12 ) {
LBRR_rate_thres_bps = LBRR_MB_MIN_RATE_BPS;
} else {
LBRR_rate_thres_bps = LBRR_WB_MIN_RATE_BPS;
}
LBRR_rate_thres_bps = silk_SMULWB( silk_MUL( LBRR_rate_thres_bps, 125 - silk_min( psEncC->PacketLoss_perc, 25 ) ), SILK_FIX_CONST( 0.01, 16 ) );
if( TargetRate_bps > LBRR_rate_thres_bps ) {
/* Set gain increase for coding LBRR excitation */
psEncC->LBRR_enabled = 1;
psEncC->LBRR_GainIncreases = silk_max_int( 7 - silk_SMULWB( psEncC->PacketLoss_perc, SILK_FIX_CONST( 0.4, 16 ) ), 2 );
}
}
return ret;
}
static OPUS_INLINE opus_int silk_setup_LBRR(
silk_encoder_state *psEncC, /* I/O */
const opus_int32 TargetRate_bps /* I */
)
{
opus_int LBRR_in_previous_packet, ret = SILK_NO_ERROR;
opus_int32 LBRR_rate_thres_bps;
LBRR_in_previous_packet = psEncC->LBRR_enabled;
psEncC->LBRR_enabled = 0;
if( psEncC->useInBandFEC && psEncC->PacketLoss_perc > 0 ) {
if( psEncC->fs_kHz == 8 ) {
LBRR_rate_thres_bps = LBRR_NB_MIN_RATE_BPS;
} else if( psEncC->fs_kHz == 12 ) {
LBRR_rate_thres_bps = LBRR_MB_MIN_RATE_BPS;
} else {
LBRR_rate_thres_bps = LBRR_WB_MIN_RATE_BPS;
}
LBRR_rate_thres_bps = silk_SMULWB( silk_MUL( LBRR_rate_thres_bps, 125 - silk_min( psEncC->PacketLoss_perc, 25 ) ), SILK_FIX_CONST( 0.01, 16 ) );
if( TargetRate_bps > LBRR_rate_thres_bps ) {
/* Set gain increase for coding LBRR excitation */
if( LBRR_in_previous_packet == 0 ) {
/* Previous packet did not have LBRR, and was therefore coded at a higher bitrate */
psEncC->LBRR_GainIncreases = 7;
} else {
psEncC->LBRR_GainIncreases = silk_max_int( 7 - silk_SMULWB( (opus_int32)psEncC->PacketLoss_perc, SILK_FIX_CONST( 0.4, 16 ) ), 2 );
}
psEncC->LBRR_enabled = 1;
}
}
return ret;
}
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 );
}
/* Convert input to a log scale */
opus_int32 silk_lin2log(
const opus_int32 inLin /* I input in linear scale */
)
{
opus_int32 lz, frac_Q7;
silk_CLZ_FRAC( inLin, &lz, &frac_Q7 );
/* Piece-wise parabolic approximation */
return silk_LSHIFT( 31 - lz, 7 ) + silk_SMLAWB( frac_Q7, silk_MUL( frac_Q7, 128 - frac_Q7 ), 179 );
}
/* Chirp (bandwidth expand) LP AR filter */
void silk_bwexpander(int16_t * ar, /* I/O AR filter to be expanded (without leading 1) */
const int d, /* I Length of ar */
int32_t chirp_Q16 /* I Chirp factor (typically in the range 0 to 1) */
)
{
int i;
int32_t chirp_minus_one_Q16 = chirp_Q16 - 65536;
/* NB: Dont use silk_SMULWB, instead of silk_RSHIFT_ROUND( silk_MUL(), 16 ), below. */
/* Bias in silk_SMULWB can lead to unstable filters */
for (i = 0; i < d - 1; i++) {
ar[i] =
(int16_t) silk_RSHIFT_ROUND(silk_MUL(chirp_Q16, ar[i]),
16);
chirp_Q16 +=
silk_RSHIFT_ROUND(silk_MUL(chirp_Q16, chirp_minus_one_Q16),
16);
}
ar[d - 1] =
(int16_t) silk_RSHIFT_ROUND(silk_MUL(chirp_Q16, ar[d - 1]), 16);
}
/* Control SNR of redidual quantizer */
opus_int silk_control_SNR(
silk_encoder_state *psEncC, /* I/O Pointer to Silk encoder state */
opus_int32 TargetRate_bps /* I Target max bitrate (bps) */
) {
opus_int k, ret = SILK_NO_ERROR;
opus_int32 frac_Q6;
const opus_int32 *rateTable;
/* Set bitrate/coding quality */
TargetRate_bps = silk_LIMIT(TargetRate_bps, MIN_TARGET_RATE_BPS, MAX_TARGET_RATE_BPS);
if (TargetRate_bps != psEncC->TargetRate_bps) {
psEncC->TargetRate_bps = TargetRate_bps;
/* If new TargetRate_bps, translate to SNR_dB value */
if (psEncC->fs_kHz == 8) {
rateTable = silk_TargetRate_table_NB;
} else if (psEncC->fs_kHz == 12) {
rateTable = silk_TargetRate_table_MB;
} else {
rateTable = silk_TargetRate_table_WB;
}
/* Reduce bitrate for 10 ms modes in these calculations */
if (psEncC->nb_subfr == 2) {
TargetRate_bps -= REDUCE_BITRATE_10_MS_BPS;
}
/* Find bitrate interval in table and interpolate */
for (k = 1; k < TARGET_RATE_TAB_SZ; k++) {
if (TargetRate_bps <= rateTable[k]) {
frac_Q6 = silk_DIV32(silk_LSHIFT(TargetRate_bps - rateTable[k - 1], 6),
rateTable[k] - rateTable[k - 1]);
psEncC->SNR_dB_Q7 = silk_LSHIFT(silk_SNR_table_Q1[k - 1], 6) + silk_MUL(frac_Q6,
silk_SNR_table_Q1[k] -
silk_SNR_table_Q1[
k -
1]);
break;
}
}
/* Reduce coding quality whenever LBRR is enabled, to free up some bits */
if (psEncC->LBRR_enabled) {
psEncC->SNR_dB_Q7 = silk_SMLABB(psEncC->SNR_dB_Q7, 12 - psEncC->LBRR_GainIncreases,
SILK_FIX_CONST(-0.25, 7));
}
}
return ret;
}
/* Chirp (bandwidth expand) LP AR filter */
void silk_bwexpander_32(
opus_int32 *ar, /* I/O AR filter to be expanded (without leading 1) */
const opus_int d, /* I Length of ar */
opus_int32 chirp_Q16 /* I Chirp factor in Q16 */
)
{
opus_int i;
opus_int32 chirp_minus_one_Q16 = chirp_Q16 - 65536;
for( i = 0; i < d - 1; i++ ) {
ar[ i ] = silk_SMULWW( chirp_Q16, ar[ i ] );
chirp_Q16 += silk_RSHIFT_ROUND( silk_MUL( chirp_Q16, chirp_minus_one_Q16 ), 16 );
}
ar[ d - 1 ] = silk_SMULWW( chirp_Q16, ar[ d - 1 ] );
}
/* Convert int32 coefficients to int16 coefs and make sure there's no wrap-around */
void silk_LPC_fit(
opus_int16 *a_QOUT, /* O Output signal */
opus_int32 *a_QIN, /* I/O Input signal */
const opus_int QOUT, /* I Input Q domain */
const opus_int QIN, /* I Input Q domain */
const opus_int d /* I Filter order */
)
{
opus_int i, k, idx = 0;
opus_int32 maxabs, absval, chirp_Q16;
/* Limit the maximum absolute value of the prediction coefficients, so that they'll fit in int16 */
for( i = 0; i < 10; i++ ) {
/* Find maximum absolute value and its index */
maxabs = 0;
for( k = 0; k < d; k++ ) {
absval = silk_abs( a_QIN[k] );
if( absval > maxabs ) {
maxabs = absval;
idx = k;
}
}
maxabs = silk_RSHIFT_ROUND( maxabs, QIN - QOUT );
if( maxabs > silk_int16_MAX ) {
/* Reduce magnitude of prediction coefficients */
maxabs = silk_min( maxabs, 163838 ); /* ( silk_int32_MAX >> 14 ) + silk_int16_MAX = 163838 */
chirp_Q16 = SILK_FIX_CONST( 0.999, 16 ) - silk_DIV32( silk_LSHIFT( maxabs - silk_int16_MAX, 14 ),
silk_RSHIFT32( silk_MUL( maxabs, idx + 1), 2 ) );
silk_bwexpander_32( a_QIN, d, chirp_Q16 );
} else {
break;
}
}
if( i == 10 ) {
/* Reached the last iteration, clip the coefficients */
for( k = 0; k < d; k++ ) {
a_QOUT[ k ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( a_QIN[ k ], QIN - QOUT ) );
a_QIN[ k ] = silk_LSHIFT( (opus_int32)a_QOUT[ k ], QIN - QOUT );
}
} else {
for( k = 0; k < d; k++ ) {
a_QOUT[ k ] = (opus_int16)silk_RSHIFT_ROUND( a_QIN[ k ], QIN - QOUT );
}
}
}
/* 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;
}
void silk_find_LTP_FIX(
opus_int16 b_Q14[ MAX_NB_SUBFR * LTP_ORDER ], /* O LTP coefs */
opus_int32 WLTP[ MAX_NB_SUBFR * LTP_ORDER * LTP_ORDER ], /* O Weight for LTP quantization */
opus_int *LTPredCodGain_Q7, /* O LTP coding gain */
const opus_int16 r_lpc[], /* I residual signal after LPC signal + state for first 10 ms */
const opus_int lag[ MAX_NB_SUBFR ], /* I LTP lags */
const opus_int32 Wght_Q15[ MAX_NB_SUBFR ], /* I weights */
const opus_int subfr_length, /* I subframe length */
const opus_int nb_subfr, /* I number of subframes */
const opus_int mem_offset, /* I number of samples in LTP memory */
opus_int corr_rshifts[ MAX_NB_SUBFR ] /* O right shifts applied to correlations */
)
{
opus_int i, k, lshift;
const opus_int16 *r_ptr, *lag_ptr;
opus_int16 *b_Q14_ptr;
opus_int32 regu;
opus_int32 *WLTP_ptr;
opus_int32 b_Q16[ LTP_ORDER ], delta_b_Q14[ LTP_ORDER ], d_Q14[ MAX_NB_SUBFR ], nrg[ MAX_NB_SUBFR ], g_Q26;
opus_int32 w[ MAX_NB_SUBFR ], WLTP_max, max_abs_d_Q14, max_w_bits;
opus_int32 temp32, denom32;
opus_int extra_shifts;
opus_int rr_shifts, maxRshifts, maxRshifts_wxtra, LZs;
opus_int32 LPC_res_nrg, LPC_LTP_res_nrg, div_Q16;
opus_int32 Rr[ LTP_ORDER ], rr[ MAX_NB_SUBFR ];
opus_int32 wd, m_Q12;
b_Q14_ptr = b_Q14;
WLTP_ptr = WLTP;
r_ptr = &r_lpc[ mem_offset ];
for( k = 0; k < nb_subfr; k++ ) {
lag_ptr = r_ptr - ( lag[ k ] + LTP_ORDER / 2 );
silk_sum_sqr_shift( &rr[ k ], &rr_shifts, r_ptr, subfr_length ); /* rr[ k ] in Q( -rr_shifts ) */
/* Assure headroom */
LZs = silk_CLZ32( rr[k] );
if( LZs < LTP_CORRS_HEAD_ROOM ) {
rr[ k ] = silk_RSHIFT_ROUND( rr[ k ], LTP_CORRS_HEAD_ROOM - LZs );
rr_shifts += ( LTP_CORRS_HEAD_ROOM - LZs );
}
corr_rshifts[ k ] = rr_shifts;
silk_corrMatrix_FIX( lag_ptr, subfr_length, LTP_ORDER, LTP_CORRS_HEAD_ROOM, WLTP_ptr, &corr_rshifts[ k ] ); /* WLTP_fix_ptr in Q( -corr_rshifts[ k ] ) */
/* The correlation vector always has lower max abs value than rr and/or RR so head room is assured */
silk_corrVector_FIX( lag_ptr, r_ptr, subfr_length, LTP_ORDER, Rr, corr_rshifts[ k ] ); /* Rr_fix_ptr in Q( -corr_rshifts[ k ] ) */
if( corr_rshifts[ k ] > rr_shifts ) {
rr[ k ] = silk_RSHIFT( rr[ k ], corr_rshifts[ k ] - rr_shifts ); /* rr[ k ] in Q( -corr_rshifts[ k ] ) */
}
silk_assert( rr[ k ] >= 0 );
regu = 1;
regu = silk_SMLAWB( regu, rr[ k ], SILK_FIX_CONST( LTP_DAMPING/3, 16 ) );
regu = silk_SMLAWB( regu, matrix_ptr( WLTP_ptr, 0, 0, LTP_ORDER ), SILK_FIX_CONST( LTP_DAMPING/3, 16 ) );
regu = silk_SMLAWB( regu, matrix_ptr( WLTP_ptr, LTP_ORDER-1, LTP_ORDER-1, LTP_ORDER ), SILK_FIX_CONST( LTP_DAMPING/3, 16 ) );
silk_regularize_correlations_FIX( WLTP_ptr, &rr[k], regu, LTP_ORDER );
silk_solve_LDL_FIX( WLTP_ptr, LTP_ORDER, Rr, b_Q16 ); /* WLTP_fix_ptr and Rr_fix_ptr both in Q(-corr_rshifts[k]) */
/* Limit and store in Q14 */
silk_fit_LTP( b_Q16, b_Q14_ptr );
/* Calculate residual energy */
nrg[ k ] = silk_residual_energy16_covar_FIX( b_Q14_ptr, WLTP_ptr, Rr, rr[ k ], LTP_ORDER, 14 ); /* nrg_fix in Q( -corr_rshifts[ k ] ) */
/* temp = Wght[ k ] / ( nrg[ k ] * Wght[ k ] + 0.01f * subfr_length ); */
extra_shifts = silk_min_int( corr_rshifts[ k ], LTP_CORRS_HEAD_ROOM );
denom32 = silk_LSHIFT_SAT32( silk_SMULWB( nrg[ k ], Wght_Q15[ k ] ), 1 + extra_shifts ) + /* Q( -corr_rshifts[ k ] + extra_shifts ) */
silk_RSHIFT( silk_SMULWB( subfr_length, 655 ), corr_rshifts[ k ] - extra_shifts ); /* Q( -corr_rshifts[ k ] + extra_shifts ) */
denom32 = silk_max( denom32, 1 );
silk_assert( ((opus_int64)Wght_Q15[ k ] << 16 ) < silk_int32_MAX ); /* Wght always < 0.5 in Q0 */
temp32 = silk_DIV32( silk_LSHIFT( (opus_int32)Wght_Q15[ k ], 16 ), denom32 ); /* Q( 15 + 16 + corr_rshifts[k] - extra_shifts ) */
temp32 = silk_RSHIFT( temp32, 31 + corr_rshifts[ k ] - extra_shifts - 26 ); /* Q26 */
/* Limit temp such that the below scaling never wraps around */
WLTP_max = 0;
for( i = 0; i < LTP_ORDER * LTP_ORDER; i++ ) {
WLTP_max = silk_max( WLTP_ptr[ i ], WLTP_max );
}
lshift = silk_CLZ32( WLTP_max ) - 1 - 3; /* keep 3 bits free for vq_nearest_neighbor_fix */
silk_assert( 26 - 18 + lshift >= 0 );
if( 26 - 18 + lshift < 31 ) {
temp32 = silk_min_32( temp32, silk_LSHIFT( (opus_int32)1, 26 - 18 + lshift ) );
}
silk_scale_vector32_Q26_lshift_18( WLTP_ptr, temp32, LTP_ORDER * LTP_ORDER ); /* WLTP_ptr in Q( 18 - corr_rshifts[ k ] ) */
w[ k ] = matrix_ptr( WLTP_ptr, LTP_ORDER/2, LTP_ORDER/2, LTP_ORDER ); /* w in Q( 18 - corr_rshifts[ k ] ) */
silk_assert( w[k] >= 0 );
r_ptr += subfr_length;
b_Q14_ptr += LTP_ORDER;
WLTP_ptr += LTP_ORDER * LTP_ORDER;
}
maxRshifts = 0;
for( k = 0; k < nb_subfr; k++ ) {
maxRshifts = silk_max_int( corr_rshifts[ k ], maxRshifts );
}
/* Compute LTP coding gain */
if( LTPredCodGain_Q7 != NULL ) {
LPC_LTP_res_nrg = 0;
LPC_res_nrg = 0;
silk_assert( LTP_CORRS_HEAD_ROOM >= 2 ); /* Check that no overflow will happen when adding */
for( k = 0; k < nb_subfr; k++ ) {
LPC_res_nrg = silk_ADD32( LPC_res_nrg, silk_RSHIFT( silk_ADD32( silk_SMULWB( rr[ k ], Wght_Q15[ k ] ), 1 ), 1 + ( maxRshifts - corr_rshifts[ k ] ) ) ); /* Q( -maxRshifts ) */
LPC_LTP_res_nrg = silk_ADD32( LPC_LTP_res_nrg, silk_RSHIFT( silk_ADD32( silk_SMULWB( nrg[ k ], Wght_Q15[ k ] ), 1 ), 1 + ( maxRshifts - corr_rshifts[ k ] ) ) ); /* Q( -maxRshifts ) */
}
LPC_LTP_res_nrg = silk_max( LPC_LTP_res_nrg, 1 ); /* avoid division by zero */
div_Q16 = silk_DIV32_varQ( LPC_res_nrg, LPC_LTP_res_nrg, 16 );
*LTPredCodGain_Q7 = ( opus_int )silk_SMULBB( 3, silk_lin2log( div_Q16 ) - ( 16 << 7 ) );
silk_assert( *LTPredCodGain_Q7 == ( opus_int )silk_SAT16( silk_MUL( 3, silk_lin2log( div_Q16 ) - ( 16 << 7 ) ) ) );
}
/* smoothing */
/* d = sum( B, 1 ); */
b_Q14_ptr = b_Q14;
for( k = 0; k < nb_subfr; k++ ) {
d_Q14[ k ] = 0;
for( i = 0; i < LTP_ORDER; i++ ) {
d_Q14[ k ] += b_Q14_ptr[ i ];
}
b_Q14_ptr += LTP_ORDER;
}
/* m = ( w * d' ) / ( sum( w ) + 1e-3 ); */
/* Find maximum absolute value of d_Q14 and the bits used by w in Q0 */
max_abs_d_Q14 = 0;
max_w_bits = 0;
for( k = 0; k < nb_subfr; k++ ) {
max_abs_d_Q14 = silk_max_32( max_abs_d_Q14, silk_abs( d_Q14[ k ] ) );
/* w[ k ] is in Q( 18 - corr_rshifts[ k ] ) */
/* Find bits needed in Q( 18 - maxRshifts ) */
max_w_bits = silk_max_32( max_w_bits, 32 - silk_CLZ32( w[ k ] ) + corr_rshifts[ k ] - maxRshifts );
}
/* max_abs_d_Q14 = (5 << 15); worst case, i.e. LTP_ORDER * -silk_int16_MIN */
silk_assert( max_abs_d_Q14 <= ( 5 << 15 ) );
/* How many bits is needed for w*d' in Q( 18 - maxRshifts ) in the worst case, of all d_Q14's being equal to max_abs_d_Q14 */
extra_shifts = max_w_bits + 32 - silk_CLZ32( max_abs_d_Q14 ) - 14;
/* Subtract what we got available; bits in output var plus maxRshifts */
extra_shifts -= ( 32 - 1 - 2 + maxRshifts ); /* Keep sign bit free as well as 2 bits for accumulation */
extra_shifts = silk_max_int( extra_shifts, 0 );
maxRshifts_wxtra = maxRshifts + extra_shifts;
temp32 = silk_RSHIFT( 262, maxRshifts + extra_shifts ) + 1; /* 1e-3f in Q( 18 - (maxRshifts + extra_shifts) ) */
wd = 0;
for( k = 0; k < nb_subfr; k++ ) {
/* w has at least 2 bits of headroom so no overflow should happen */
temp32 = silk_ADD32( temp32, silk_RSHIFT( w[ k ], maxRshifts_wxtra - corr_rshifts[ k ] ) ); /* Q( 18 - maxRshifts_wxtra ) */
wd = silk_ADD32( wd, silk_LSHIFT( silk_SMULWW( silk_RSHIFT( w[ k ], maxRshifts_wxtra - corr_rshifts[ k ] ), d_Q14[ k ] ), 2 ) ); /* Q( 18 - maxRshifts_wxtra ) */
}
m_Q12 = silk_DIV32_varQ( wd, temp32, 12 );
b_Q14_ptr = b_Q14;
for( k = 0; k < nb_subfr; k++ ) {
/* w_fix[ k ] from Q( 18 - corr_rshifts[ k ] ) to Q( 16 ) */
if( 2 - corr_rshifts[k] > 0 ) {
temp32 = silk_RSHIFT( w[ k ], 2 - corr_rshifts[ k ] );
} else {
temp32 = silk_LSHIFT_SAT32( w[ k ], corr_rshifts[ k ] - 2 );
}
g_Q26 = silk_MUL(
silk_DIV32(
SILK_FIX_CONST( LTP_SMOOTHING, 26 ),
silk_RSHIFT( SILK_FIX_CONST( LTP_SMOOTHING, 26 ), 10 ) + temp32 ), /* Q10 */
silk_LSHIFT_SAT32( silk_SUB_SAT32( (opus_int32)m_Q12, silk_RSHIFT( d_Q14[ k ], 2 ) ), 4 ) ); /* Q16 */
temp32 = 0;
for( i = 0; i < LTP_ORDER; i++ ) {
delta_b_Q14[ i ] = silk_max_16( b_Q14_ptr[ i ], 1638 ); /* 1638_Q14 = 0.1_Q0 */
temp32 += delta_b_Q14[ i ]; /* Q14 */
}
temp32 = silk_DIV32( g_Q26, temp32 ); /* Q14 -> Q12 */
for( i = 0; i < LTP_ORDER; i++ ) {
b_Q14_ptr[ i ] = silk_LIMIT_32( (opus_int32)b_Q14_ptr[ i ] + silk_SMULWB( silk_LSHIFT_SAT32( temp32, 4 ), delta_b_Q14[ i ] ), -16000, 28000 );
}
b_Q14_ptr += LTP_ORDER;
}
}
/* Residual energy: nrg = wxx - 2 * wXx * c + c' * wXX * c */
opus_int32 silk_residual_energy16_covar_FIX(
const opus_int16 *c, /* I Prediction vector */
const opus_int32 *wXX, /* I Correlation matrix */
const opus_int32 *wXx, /* I Correlation vector */
opus_int32 wxx, /* I Signal energy */
opus_int D, /* I Dimension */
opus_int cQ /* I Q value for c vector 0 - 15 */
)
{
opus_int i, j, lshifts, Qxtra;
opus_int32 c_max, w_max, tmp, tmp2, nrg;
opus_int cn[ MAX_MATRIX_SIZE ];
const opus_int32 *pRow;
/* Safety checks */
silk_assert( D >= 0 );
silk_assert( D <= 16 );
silk_assert( cQ > 0 );
silk_assert( cQ < 16 );
lshifts = 16 - cQ;
Qxtra = lshifts;
c_max = 0;
for( i = 0; i < D; i++ ) {
c_max = silk_max_32( c_max, silk_abs( (opus_int32)c[ i ] ) );
}
Qxtra = silk_min_int( Qxtra, silk_CLZ32( c_max ) - 17 );
w_max = silk_max_32( wXX[ 0 ], wXX[ D * D - 1 ] );
Qxtra = silk_min_int( Qxtra, silk_CLZ32( silk_MUL( D, silk_RSHIFT( silk_SMULWB( w_max, c_max ), 4 ) ) ) - 5 );
Qxtra = silk_max_int( Qxtra, 0 );
for( i = 0; i < D; i++ ) {
cn[ i ] = silk_LSHIFT( ( opus_int )c[ i ], Qxtra );
silk_assert( silk_abs(cn[i]) <= ( silk_int16_MAX + 1 ) ); /* Check that silk_SMLAWB can be used */
}
lshifts -= Qxtra;
/* Compute wxx - 2 * wXx * c */
tmp = 0;
for( i = 0; i < D; i++ ) {
tmp = silk_SMLAWB( tmp, wXx[ i ], cn[ i ] );
}
nrg = silk_RSHIFT( wxx, 1 + lshifts ) - tmp; /* Q: -lshifts - 1 */
/* Add c' * wXX * c, assuming wXX is symmetric */
tmp2 = 0;
for( i = 0; i < D; i++ ) {
tmp = 0;
pRow = &wXX[ i * D ];
for( j = i + 1; j < D; j++ ) {
tmp = silk_SMLAWB( tmp, pRow[ j ], cn[ j ] );
}
tmp = silk_SMLAWB( tmp, silk_RSHIFT( pRow[ i ], 1 ), cn[ i ] );
tmp2 = silk_SMLAWB( tmp2, tmp, cn[ i ] );
}
nrg = silk_ADD_LSHIFT32( nrg, tmp2, lshifts ); /* Q: -lshifts - 1 */
/* Keep one bit free always, because we add them for LSF interpolation */
if( nrg < 1 ) {
nrg = 1;
} else if( nrg > silk_RSHIFT( silk_int32_MAX, lshifts + 2 ) ) {
nrg = silk_int32_MAX >> 1;
} else {
/* 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;
}
/* compute whitening filter coefficients from normalized line spectral frequencies */
void silk_NLSF2A(
opus_int16 *a_Q12, /* O monic whitening filter coefficients in Q12, [ d ] */
const opus_int16 *NLSF, /* I normalized line spectral frequencies in Q15, [ d ] */
const opus_int d /* I filter order (should be even) */
)
{
/* This ordering was found to maximize quality. It improves numerical accuracy of
silk_NLSF2A_find_poly() compared to "standard" ordering. */
static const unsigned char ordering16[16] = {
0, 15, 8, 7, 4, 11, 12, 3, 2, 13, 10, 5, 6, 9, 14, 1
};
static const unsigned char ordering10[10] = {
0, 9, 6, 3, 4, 5, 8, 1, 2, 7
};
const unsigned char *ordering;
opus_int k, i, dd;
opus_int32 cos_LSF_QA[ SILK_MAX_ORDER_LPC ];
opus_int32 P[ SILK_MAX_ORDER_LPC / 2 + 1 ], Q[ SILK_MAX_ORDER_LPC / 2 + 1 ];
opus_int32 Ptmp, Qtmp, f_int, f_frac, cos_val, delta;
opus_int32 a32_QA1[ SILK_MAX_ORDER_LPC ];
opus_int32 maxabs, absval, idx=0, sc_Q16;
silk_assert( LSF_COS_TAB_SZ_FIX == 128 );
silk_assert( d==10||d==16 );
/* convert LSFs to 2*cos(LSF), using piecewise linear curve from table */
ordering = d == 16 ? ordering16 : ordering10;
for( k = 0; k < d; k++ ) {
silk_assert(NLSF[k] >= 0 );
/* f_int on a scale 0-127 (rounded down) */
f_int = silk_RSHIFT( NLSF[k], 15 - 7 );
/* f_frac, range: 0..255 */
f_frac = NLSF[k] - silk_LSHIFT( f_int, 15 - 7 );
silk_assert(f_int >= 0);
silk_assert(f_int < LSF_COS_TAB_SZ_FIX );
/* Read start and end value from table */
cos_val = silk_LSFCosTab_FIX_Q12[ f_int ]; /* Q12 */
delta = silk_LSFCosTab_FIX_Q12[ f_int + 1 ] - cos_val; /* Q12, with a range of 0..200 */
/* Linear interpolation */
cos_LSF_QA[ordering[k]] = silk_RSHIFT_ROUND( silk_LSHIFT( cos_val, 8 ) + silk_MUL( delta, f_frac ), 20 - QA ); /* QA */
}
dd = silk_RSHIFT( d, 1 );
/* generate even and odd polynomials using convolution */
silk_NLSF2A_find_poly( P, &cos_LSF_QA[ 0 ], dd );
silk_NLSF2A_find_poly( Q, &cos_LSF_QA[ 1 ], dd );
/* convert even and odd polynomials to opus_int32 Q12 filter coefs */
for( k = 0; k < dd; k++ ) {
Ptmp = P[ k+1 ] + P[ k ];
Qtmp = Q[ k+1 ] - Q[ k ];
/* the Ptmp and Qtmp values at this stage need to fit in int32 */
a32_QA1[ k ] = -Qtmp - Ptmp; /* QA+1 */
a32_QA1[ d-k-1 ] = Qtmp - Ptmp; /* QA+1 */
}
/* Limit the maximum absolute value of the prediction coefficients, so that they'll fit in int16 */
for( i = 0; i < 10; i++ ) {
/* Find maximum absolute value and its index */
maxabs = 0;
for( k = 0; k < d; k++ ) {
absval = silk_abs( a32_QA1[k] );
if( absval > maxabs ) {
maxabs = absval;
idx = k;
}
}
maxabs = silk_RSHIFT_ROUND( maxabs, QA + 1 - 12 ); /* QA+1 -> Q12 */
if( maxabs > silk_int16_MAX ) {
/* Reduce magnitude of prediction coefficients */
maxabs = silk_min( maxabs, 163838 ); /* ( silk_int32_MAX >> 14 ) + silk_int16_MAX = 163838 */
sc_Q16 = SILK_FIX_CONST( 0.999, 16 ) - silk_DIV32( silk_LSHIFT( maxabs - silk_int16_MAX, 14 ),
silk_RSHIFT32( silk_MUL( maxabs, idx + 1), 2 ) );
silk_bwexpander_32( a32_QA1, d, sc_Q16 );
} else {
break;
}
}
if( i == 10 ) {
/* Reached the last iteration, clip the coefficients */
for( k = 0; k < d; k++ ) {
a_Q12[ k ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( a32_QA1[ k ], QA + 1 - 12 ) ); /* QA+1 -> Q12 */
a32_QA1[ k ] = silk_LSHIFT( (opus_int32)a_Q12[ k ], QA + 1 - 12 );
}
} else {
for( k = 0; k < d; k++ ) {
a_Q12[ k ] = (opus_int16)silk_RSHIFT_ROUND( a32_QA1[ k ], QA + 1 - 12 ); /* QA+1 -> Q12 */
}
}
for( i = 0; i < MAX_LPC_STABILIZE_ITERATIONS; i++ ) {
if( silk_LPC_inverse_pred_gain( a_Q12, d ) < SILK_FIX_CONST( 1.0 / MAX_PREDICTION_POWER_GAIN, 30 ) ) {
/* Prediction coefficients are (too close to) unstable; apply bandwidth expansion */
/* on the unscaled coefficients, convert to Q12 and measure again */
silk_bwexpander_32( a32_QA1, d, 65536 - silk_LSHIFT( 2, i ) );
for( k = 0; k < d; k++ ) {
a_Q12[ k ] = (opus_int16)silk_RSHIFT_ROUND( a32_QA1[ k ], QA + 1 - 12 ); /* QA+1 -> Q12 */
}
} else {
break;
}
}
}
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;
}
/* Decode parameters from payload */
void silk_decode_parameters(silk_decoder_state * psDec, /* I/O State */
silk_decoder_control * psDecCtrl, /* I/O Decoder control */
int condCoding /* I The type of conditional coding to use */
)
{
int i, k, Ix;
int16_t pNLSF_Q15[MAX_LPC_ORDER], pNLSF0_Q15[MAX_LPC_ORDER];
const int8_t *cbk_ptr_Q7;
/* Dequant Gains */
silk_gains_dequant(psDecCtrl->Gains_Q16, psDec->indices.GainsIndices,
&psDec->LastGainIndex,
condCoding == CODE_CONDITIONALLY, psDec->nb_subfr);
/****************/
/* Decode NLSFs */
/****************/
silk_NLSF_decode(pNLSF_Q15, psDec->indices.NLSFIndices,
psDec->psNLSF_CB);
/* Convert NLSF parameters to AR prediction filter coefficients */
silk_NLSF2A(psDecCtrl->PredCoef_Q12[1], pNLSF_Q15, psDec->LPC_order);
/* If just reset, e.g., because internal Fs changed, do not allow interpolation */
/* improves the case of packet loss in the first frame after a switch */
if (psDec->first_frame_after_reset == 1) {
psDec->indices.NLSFInterpCoef_Q2 = 4;
}
if (psDec->indices.NLSFInterpCoef_Q2 < 4) {
/* Calculation of the interpolated NLSF0 vector from the interpolation factor, */
/* the previous NLSF1, and the current NLSF1 */
for (i = 0; i < psDec->LPC_order; i++) {
pNLSF0_Q15[i] =
psDec->prevNLSF_Q15[i] +
silk_RSHIFT(silk_MUL
(psDec->indices.NLSFInterpCoef_Q2,
pNLSF_Q15[i] - psDec->prevNLSF_Q15[i]),
2);
}
/* Convert NLSF parameters to AR prediction filter coefficients */
silk_NLSF2A(psDecCtrl->PredCoef_Q12[0], pNLSF0_Q15,
psDec->LPC_order);
} else {
/* Copy LPC coefficients for first half from second half */
memcpy(psDecCtrl->PredCoef_Q12[0],
psDecCtrl->PredCoef_Q12[1],
psDec->LPC_order * sizeof(int16_t));
}
memcpy(psDec->prevNLSF_Q15, pNLSF_Q15,
psDec->LPC_order * sizeof(int16_t));
/* After a packet loss do BWE of LPC coefs */
if (psDec->lossCnt) {
silk_bwexpander(psDecCtrl->PredCoef_Q12[0], psDec->LPC_order,
BWE_AFTER_LOSS_Q16);
silk_bwexpander(psDecCtrl->PredCoef_Q12[1], psDec->LPC_order,
BWE_AFTER_LOSS_Q16);
}
if (psDec->indices.signalType == TYPE_VOICED) {
/*********************/
/* Decode pitch lags */
/*********************/
/* Decode pitch values */
silk_decode_pitch(psDec->indices.lagIndex,
psDec->indices.contourIndex,
psDecCtrl->pitchL, psDec->fs_kHz,
psDec->nb_subfr);
/* Decode Codebook Index */
cbk_ptr_Q7 = silk_LTP_vq_ptrs_Q7[psDec->indices.PERIndex]; /* set pointer to start of codebook */
for (k = 0; k < psDec->nb_subfr; k++) {
Ix = psDec->indices.LTPIndex[k];
for (i = 0; i < LTP_ORDER; i++) {
psDecCtrl->LTPCoef_Q14[k * LTP_ORDER + i] =
silk_LSHIFT(cbk_ptr_Q7[Ix * LTP_ORDER + i],
7);
}
}
/**********************/
/* Decode LTP scaling */
/**********************/
Ix = psDec->indices.LTP_scaleIndex;
psDecCtrl->LTP_scale_Q14 = silk_LTPScales_table_Q14[Ix];
} else {
memzero(psDecCtrl->pitchL, psDec->nb_subfr * sizeof(int));
memzero(psDecCtrl->LTPCoef_Q14,
LTP_ORDER * psDec->nb_subfr * sizeof(int16_t));
psDec->indices.PERIndex = 0;
psDecCtrl->LTP_scale_Q14 = 0;
}
}
void silk_prefilter_FIX(
silk_encoder_state_FIX *psEnc, /* I/O Encoder state */
const silk_encoder_control_FIX *psEncCtrl, /* I Encoder control */
opus_int32 xw_Q3[], /* O Weighted signal */
const opus_int16 x[] /* I Speech signal */
)
{
silk_prefilter_state_FIX *P = &psEnc->sPrefilt;
opus_int j, k, lag;
opus_int32 tmp_32;
const opus_int16 *AR1_shp_Q13;
const opus_int16 *px;
opus_int32 *pxw_Q3;
opus_int HarmShapeGain_Q12, Tilt_Q14;
opus_int32 HarmShapeFIRPacked_Q12, LF_shp_Q14;
VARDECL( opus_int32, x_filt_Q12 );
VARDECL( opus_int32, st_res_Q2 );
opus_int16 B_Q10[ 2 ];
SAVE_STACK;
/* Set up pointers */
px = x;
pxw_Q3 = xw_Q3;
lag = P->lagPrev;
ALLOC( x_filt_Q12, psEnc->sCmn.subfr_length, opus_int32 );
ALLOC( st_res_Q2, psEnc->sCmn.subfr_length, opus_int32 );
for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) {
/* Update Variables that change per sub frame */
if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) {
lag = psEncCtrl->pitchL[ k ];
}
/* Noise shape parameters */
HarmShapeGain_Q12 = silk_SMULWB( (opus_int32)psEncCtrl->HarmShapeGain_Q14[ k ], 16384 - psEncCtrl->HarmBoost_Q14[ k ] );
silk_assert( HarmShapeGain_Q12 >= 0 );
HarmShapeFIRPacked_Q12 = silk_RSHIFT( HarmShapeGain_Q12, 2 );
HarmShapeFIRPacked_Q12 |= silk_LSHIFT( (opus_int32)silk_RSHIFT( HarmShapeGain_Q12, 1 ), 16 );
Tilt_Q14 = psEncCtrl->Tilt_Q14[ k ];
LF_shp_Q14 = psEncCtrl->LF_shp_Q14[ k ];
AR1_shp_Q13 = &psEncCtrl->AR1_Q13[ k * MAX_SHAPE_LPC_ORDER ];
/* Short term FIR filtering*/
silk_warped_LPC_analysis_filter_FIX( P->sAR_shp, st_res_Q2, AR1_shp_Q13, px,
psEnc->sCmn.warping_Q16, psEnc->sCmn.subfr_length, psEnc->sCmn.shapingLPCOrder );
/* Reduce (mainly) low frequencies during harmonic emphasis */
B_Q10[ 0 ] = silk_RSHIFT_ROUND( psEncCtrl->GainsPre_Q14[ k ], 4 );
tmp_32 = silk_SMLABB( SILK_FIX_CONST( INPUT_TILT, 26 ), psEncCtrl->HarmBoost_Q14[ k ], HarmShapeGain_Q12 ); /* Q26 */
tmp_32 = silk_SMLABB( tmp_32, psEncCtrl->coding_quality_Q14, SILK_FIX_CONST( HIGH_RATE_INPUT_TILT, 12 ) ); /* Q26 */
tmp_32 = silk_SMULWB( tmp_32, -psEncCtrl->GainsPre_Q14[ k ] ); /* Q24 */
tmp_32 = silk_RSHIFT_ROUND( tmp_32, 14 ); /* Q10 */
B_Q10[ 1 ]= silk_SAT16( tmp_32 );
x_filt_Q12[ 0 ] = silk_MLA( silk_MUL( st_res_Q2[ 0 ], B_Q10[ 0 ] ), P->sHarmHP_Q2, B_Q10[ 1 ] );
for( j = 1; j < psEnc->sCmn.subfr_length; j++ ) {
x_filt_Q12[ j ] = silk_MLA( silk_MUL( st_res_Q2[ j ], B_Q10[ 0 ] ), st_res_Q2[ j - 1 ], B_Q10[ 1 ] );
}
P->sHarmHP_Q2 = st_res_Q2[ psEnc->sCmn.subfr_length - 1 ];
silk_prefilt_FIX( P, x_filt_Q12, pxw_Q3, HarmShapeFIRPacked_Q12, Tilt_Q14, LF_shp_Q14, lag, psEnc->sCmn.subfr_length );
px += psEnc->sCmn.subfr_length;
pxw_Q3 += psEnc->sCmn.subfr_length;
}
P->lagPrev = psEncCtrl->pitchL[ psEnc->sCmn.nb_subfr - 1 ];
RESTORE_STACK;
}
/* amplitude of monic warped coefficients by using bandwidth expansion on the true coefficients */
static inline void limit_warped_coefs(
opus_int32 *coefs_syn_Q24,
opus_int32 *coefs_ana_Q24,
opus_int lambda_Q16,
opus_int32 limit_Q24,
opus_int order
) {
opus_int i, iter, ind = 0;
opus_int32 tmp, maxabs_Q24, chirp_Q16, gain_syn_Q16, gain_ana_Q16;
opus_int32 nom_Q16, den_Q24;
/* Convert to monic coefficients */
lambda_Q16 = -lambda_Q16;
for( i = order - 1; i > 0; i-- ) {
coefs_syn_Q24[ i - 1 ] = silk_SMLAWB( coefs_syn_Q24[ i - 1 ], coefs_syn_Q24[ i ], lambda_Q16 );
coefs_ana_Q24[ i - 1 ] = silk_SMLAWB( coefs_ana_Q24[ i - 1 ], coefs_ana_Q24[ i ], lambda_Q16 );
}
lambda_Q16 = -lambda_Q16;
nom_Q16 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 16 ), -lambda_Q16, lambda_Q16 );
den_Q24 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 24 ), coefs_syn_Q24[ 0 ], lambda_Q16 );
gain_syn_Q16 = silk_DIV32_varQ( nom_Q16, den_Q24, 24 );
den_Q24 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 24 ), coefs_ana_Q24[ 0 ], lambda_Q16 );
gain_ana_Q16 = silk_DIV32_varQ( nom_Q16, den_Q24, 24 );
for( i = 0; i < order; i++ ) {
coefs_syn_Q24[ i ] = silk_SMULWW( gain_syn_Q16, coefs_syn_Q24[ i ] );
coefs_ana_Q24[ i ] = silk_SMULWW( gain_ana_Q16, coefs_ana_Q24[ i ] );
}
for( iter = 0; iter < 10; iter++ ) {
/* Find maximum absolute value */
maxabs_Q24 = -1;
for( i = 0; i < order; i++ ) {
tmp = silk_max( silk_abs_int32( coefs_syn_Q24[ i ] ), silk_abs_int32( coefs_ana_Q24[ i ] ) );
if( tmp > maxabs_Q24 ) {
maxabs_Q24 = tmp;
ind = i;
}
}
if( maxabs_Q24 <= limit_Q24 ) {
/* Coefficients are within range - done */
return;
}
/* Convert back to true warped coefficients */
for( i = 1; i < order; i++ ) {
coefs_syn_Q24[ i - 1 ] = silk_SMLAWB( coefs_syn_Q24[ i - 1 ], coefs_syn_Q24[ i ], lambda_Q16 );
coefs_ana_Q24[ i - 1 ] = silk_SMLAWB( coefs_ana_Q24[ i - 1 ], coefs_ana_Q24[ i ], lambda_Q16 );
}
gain_syn_Q16 = silk_INVERSE32_varQ( gain_syn_Q16, 32 );
gain_ana_Q16 = silk_INVERSE32_varQ( gain_ana_Q16, 32 );
for( i = 0; i < order; i++ ) {
coefs_syn_Q24[ i ] = silk_SMULWW( gain_syn_Q16, coefs_syn_Q24[ i ] );
coefs_ana_Q24[ i ] = silk_SMULWW( gain_ana_Q16, coefs_ana_Q24[ i ] );
}
/* Apply bandwidth expansion */
chirp_Q16 = SILK_FIX_CONST( 0.99, 16 ) - silk_DIV32_varQ(
silk_SMULWB( maxabs_Q24 - limit_Q24, silk_SMLABB( SILK_FIX_CONST( 0.8, 10 ), SILK_FIX_CONST( 0.1, 10 ), iter ) ),
silk_MUL( maxabs_Q24, ind + 1 ), 22 );
silk_bwexpander_32( coefs_syn_Q24, order, chirp_Q16 );
silk_bwexpander_32( coefs_ana_Q24, order, chirp_Q16 );
/* Convert to monic warped coefficients */
lambda_Q16 = -lambda_Q16;
for( i = order - 1; i > 0; i-- ) {
coefs_syn_Q24[ i - 1 ] = silk_SMLAWB( coefs_syn_Q24[ i - 1 ], coefs_syn_Q24[ i ], lambda_Q16 );
coefs_ana_Q24[ i - 1 ] = silk_SMLAWB( coefs_ana_Q24[ i - 1 ], coefs_ana_Q24[ i ], lambda_Q16 );
}
lambda_Q16 = -lambda_Q16;
nom_Q16 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 16 ), -lambda_Q16, lambda_Q16 );
den_Q24 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 24 ), coefs_syn_Q24[ 0 ], lambda_Q16 );
gain_syn_Q16 = silk_DIV32_varQ( nom_Q16, den_Q24, 24 );
den_Q24 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 24 ), coefs_ana_Q24[ 0 ], lambda_Q16 );
gain_ana_Q16 = silk_DIV32_varQ( nom_Q16, den_Q24, 24 );
for( i = 0; i < order; i++ ) {
coefs_syn_Q24[ i ] = silk_SMULWW( gain_syn_Q16, coefs_syn_Q24[ i ] );
coefs_ana_Q24[ i ] = silk_SMULWW( gain_ana_Q16, coefs_ana_Q24[ i ] );
}
}
silk_assert( 0 );
}
/* 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;
}
/* 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;
}
/* 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;
}
