void Lsp_get_quant(
Word16 lspcb1[][M], /* (i) Q13 : first stage LSP codebook */
Word16 lspcb2[][M], /* (i) Q13 : Second stage LSP codebook */
Word16 code0, /* (i) : selected code of first stage */
Word16 code1, /* (i) : selected code of second stage */
Word16 code2, /* (i) : selected code of second stage */
Word16 fg[][M], /* (i) Q15 : MA prediction coef. */
Word16 freq_prev[][M], /* (i) Q13 : previous LSP vector */
Word16 lspq[], /* (o) Q13 : quantized LSP parameters */
Word16 fg_sum[] /* (i) Q15 : present MA prediction coef. */
)
{
Word16 j;
Word16 buf[M]; /* Q13 */
for ( j = 0 ; j < NC ; j++ )
buf[j] = add( lspcb1[code0][j], lspcb2[code1][j] );
for ( j = NC ; j < M ; j++ )
buf[j] = add( lspcb1[code0][j], lspcb2[code2][j] );
Lsp_expand_1_2(buf, GAP1);
Lsp_expand_1_2(buf, GAP2);
Lsp_prev_compose(buf, lspq, fg, freq_prev, fg_sum);
Lsp_prev_update(buf, freq_prev);
Lsp_stability( lspq );
return;
}
void sid_lsfq_decode(Word16 *index, /* (i) : quantized indices */
Word16 *lspq, /* (o) : quantized lsp vector */
Word16 freq_prev[MA_NP][M] /* (i) : memory of predictor */
)
{
Word32 acc0;
Word16 i, j, k, lsfq[M], tmpbuf[M];
/* get the lsf error vector */
Copy(lspcb1[PtrTab_1[index[1]]], tmpbuf, M);
for (i=0; i<M/2; i++)
tmpbuf[i] = add(tmpbuf[i], lspcb2[PtrTab_2[0][index[2]]][i]);
for (i=M/2; i<M; i++)
tmpbuf[i] = add(tmpbuf[i], lspcb2[PtrTab_2[1][index[2]]][i]);
/* guarantee minimum distance of 0.0012 (~10 in Q13) between tmpbuf[j]
and tmpbuf[j+1] */
for (j=1; j<M; j++){
acc0 = L_mult(tmpbuf[j-1], 16384);
acc0 = L_mac(acc0, tmpbuf[j], -16384);
acc0 = L_mac(acc0, 10, 16384);
k = extract_h(acc0);
if (k > 0){
tmpbuf[j-1] = sub(tmpbuf[j-1], k);
tmpbuf[j] = add(tmpbuf[j], k);
}
}
/* compute the quantized lsf vector */
Lsp_prev_compose(tmpbuf, lsfq, noise_fg[index[0]], freq_prev,
noise_fg_sum[index[0]]);
/* update the prediction memory */
Lsp_prev_update(tmpbuf, freq_prev);
/* lsf stability check */
Lsp_stability(lsfq);
/* convert lsf to lsp */
Lsf_lsp2(lsfq, lspq, M);
}
/*----------------------------------------------------------------------------
* Lsp_iqua_cs - LSP main quantization routine
*----------------------------------------------------------------------------
*/
void Lsp_iqua_cs(
Word16 prm[], /* (i) : indexes of the selected LSP */
Word16 lsp_q[], /* (o) Q13 : Quantized LSP parameters */
Word16 erase /* (i) : frame erase information */
)
{
Word16 mode_index;
Word16 code0;
Word16 code1;
Word16 code2;
Word16 buf[M]; /* Q13 */
if( erase==0 ) { /* Not frame erasure */
mode_index = shr(prm[0] ,NC0_B) & (Word16)1;
code0 = prm[0] & (Word16)(NC0 - 1);
code1 = shr(prm[1] ,NC1_B) & (Word16)(NC1 - 1);
code2 = prm[1] & (Word16)(NC1 - 1);
/* compose quantized LSP (lsp_q) from indexes */
Lsp_get_quant(lspcb1, lspcb2, code0, code1, code2,
fg[mode_index], freq_prev, lsp_q, fg_sum[mode_index]);
/* save parameters to use in case of the frame erased situation */
Copy(lsp_q, prev_lsp, M);
prev_ma = mode_index;
}
else { /* Frame erased */
/* use revious LSP */
Copy(prev_lsp, lsp_q, M);
/* update freq_prev */
Lsp_prev_extract(prev_lsp, buf,
fg[prev_ma], freq_prev, fg_sum_inv[prev_ma]);
Lsp_prev_update(buf, freq_prev);
}
return;
}
void Coder_ld8h(
Word16 ana[], /* (o) : analysis parameters */
Word16 rate /* input : rate selector/frame =0 6.4kbps , =1 8kbps,= 2 11.8 kbps*/
)
{
/* LPC analysis */
Word16 r_l_fwd[MP1], r_h_fwd[MP1]; /* Autocorrelations low and hi (forward) */
Word32 r_bwd[M_BWDP1]; /* Autocorrelations (backward) */
Word16 r_l_bwd[M_BWDP1]; /* Autocorrelations low (backward) */
Word16 r_h_bwd[M_BWDP1]; /* Autocorrelations high (backward) */
Word16 rc_fwd[M]; /* Reflection coefficients : forward analysis */
Word16 rc_bwd[M_BWD]; /* Reflection coefficients : backward analysis */
Word16 A_t_fwd[MP1*2]; /* A(z) forward unquantized for the 2 subframes */
Word16 A_t_fwd_q[MP1*2]; /* A(z) forward quantized for the 2 subframes */
Word16 A_t_bwd[2*M_BWDP1]; /* A(z) backward for the 2 subframes */
Word16 *Aq; /* A(z) "quantized" for the 2 subframes */
Word16 *Ap; /* A(z) "unquantized" for the 2 subframes */
Word16 *pAp, *pAq;
Word16 Ap1[M_BWDP1]; /* A(z) with spectral expansion */
Word16 Ap2[M_BWDP1]; /* A(z) with spectral expansion */
Word16 lsp_new[M], lsp_new_q[M]; /* LSPs at 2th subframe */
Word16 lsf_int[M]; /* Interpolated LSF 1st subframe. */
Word16 lsf_new[M];
Word16 lp_mode; /* Backward / Forward Indication mode */
Word16 m_ap, m_aq, i_gamma;
Word16 code_lsp[2];
/* Other vectors */
Word16 h1[L_SUBFR]; /* Impulse response h1[] */
Word16 xn[L_SUBFR]; /* Target vector for pitch search */
Word16 xn2[L_SUBFR]; /* Target vector for codebook search */
Word16 code[L_SUBFR]; /* Fixed codebook excitation */
Word16 y1[L_SUBFR]; /* Filtered adaptive excitation */
Word16 y2[L_SUBFR]; /* Filtered fixed codebook excitation */
Word16 g_coeff[4]; /* Correlations between xn & y1 */
Word16 res2[L_SUBFR]; /* residual after long term prediction*/
Word16 g_coeff_cs[5];
Word16 exp_g_coeff_cs[5]; /* Correlations between xn, y1, & y2
<y1,y1>, -2<xn,y1>,
<y2,y2>, -2<xn,y2>, 2<y1,y2> */
/* Scalars */
Word16 i, j, k, i_subfr;
Word16 T_op, T0, T0_min, T0_max, T0_frac;
Word16 gain_pit, gain_code, index;
Word16 taming, pit_sharp;
Word16 sat_filter;
Word32 L_temp;
Word16 freq_cur[M];
Word16 temp;
/*------------------------------------------------------------------------*
* - Perform LPC analysis: *
* * autocorrelation + lag windowing *
* * Levinson-durbin algorithm to find a[] *
* * convert a[] to lsp[] *
* * quantize and code the LSPs *
* * find the interpolated LSPs and convert to a[] for the 2 *
* subframes (both quantized and unquantized) *
*------------------------------------------------------------------------*/
/* ------------------- */
/* LP Forward analysis */
/* ------------------- */
Autocorr(p_window, M, r_h_fwd, r_l_fwd); /* Autocorrelations */
Lag_window(M, r_h_fwd, r_l_fwd); /* Lag windowing */
Levinsone(M, r_h_fwd, r_l_fwd, &A_t_fwd[MP1], rc_fwd, old_A_fwd, old_rc_fwd); /* Levinson Durbin */
Az_lsp(&A_t_fwd[MP1], lsp_new, lsp_old); /* From A(z) to lsp */
/* -------------------- */
/* LP Backward analysis */
/* -------------------- */
/* -------------------- */
/* LP Backward analysis */
/* -------------------- */
if ( rate== G729E) {
/* LPC recursive Window as in G728 */
autocorr_hyb_window(synth, r_bwd, rexp); /* Autocorrelations */
Lag_window_bwd(r_bwd, r_h_bwd, r_l_bwd); /* Lag windowing */
/* Fixed Point Levinson (as in G729) */
Levinsone(M_BWD, r_h_bwd, r_l_bwd, &A_t_bwd[M_BWDP1], rc_bwd,
old_A_bwd, old_rc_bwd);
/* Tests saturation of A_t_bwd */
sat_filter = 0;
for (i=M_BWDP1; i<2*M_BWDP1; i++) if (A_t_bwd[i] >= 32767) sat_filter = 1;
if (sat_filter == 1) Copy(A_t_bwd_mem, &A_t_bwd[M_BWDP1], M_BWDP1);
else Copy(&A_t_bwd[M_BWDP1], A_t_bwd_mem, M_BWDP1);
/* Additional bandwidth expansion on backward filter */
Weight_Az(&A_t_bwd[M_BWDP1], GAMMA_BWD, M_BWD, &A_t_bwd[M_BWDP1]);
}
/*--------------------------------------------------*
* Update synthesis signal for next frame. *
*--------------------------------------------------*/
Copy(&synth[L_FRAME], &synth[0], MEM_SYN_BWD);
/*--------------------------------------------------------------------*
* Find interpolated LPC parameters in all subframes (unquantized). *
* The interpolated parameters are in array A_t[] of size (M+1)*4 *
*--------------------------------------------------------------------*/
if( prev_lp_mode == 0) {
Int_lpc(lsp_old, lsp_new, lsf_int, lsf_new, A_t_fwd);
}
else {
/* no interpolation */
/* unquantized */
Lsp_Az(lsp_new, A_t_fwd); /* Subframe 1 */
Lsp_lsf(lsp_new, lsf_new, M); /* transformation from LSP to LSF (freq.domain) */
Copy(lsf_new, lsf_int, M); /* Subframe 1 */
}
/* ---------------- */
/* LSP quantization */
/* ---------------- */
Qua_lspe(lsp_new, lsp_new_q, code_lsp, freq_prev, freq_cur);
/*--------------------------------------------------------------------*
* Find interpolated LPC parameters in all subframes (quantized) *
* the quantized interpolated parameters are in array Aq_t[] *
*--------------------------------------------------------------------*/
if( prev_lp_mode == 0) {
Int_qlpc(lsp_old_q, lsp_new_q, A_t_fwd_q);
}
else {
/* no interpolation */
Lsp_Az(lsp_new_q, &A_t_fwd_q[MP1]); /* Subframe 2 */
Copy(&A_t_fwd_q[MP1], A_t_fwd_q, MP1); /* Subframe 1 */
}
/*---------------------------------------------------------------------*
* - Decision for the switch Forward / Backward *
*---------------------------------------------------------------------*/
if(rate == G729E) {
set_lpc_modeg(speech, A_t_fwd_q, A_t_bwd, &lp_mode,
lsp_new, lsp_old, &bwd_dominant, prev_lp_mode, prev_filter,
&C_int, &glob_stat, &stat_bwd, &val_stat_bwd);
}
else {
update_bwd( &lp_mode, &bwd_dominant, &C_int, &glob_stat);
}
/* ---------------------------------- */
/* update the LSPs for the next frame */
/* ---------------------------------- */
Copy(lsp_new, lsp_old, M);
/*----------------------------------------------------------------------*
* - Find the weighted input speech w_sp[] for the whole speech frame *
*----------------------------------------------------------------------*/
if(lp_mode == 0) {
m_ap = M;
if (bwd_dominant == 0) Ap = A_t_fwd;
else Ap = A_t_fwd_q;
perc_var(gamma1, gamma2, lsf_int, lsf_new, rc_fwd);
}
else {
if (bwd_dominant == 0) {
m_ap = M;
Ap = A_t_fwd;
}
else {
m_ap = M_BWD;
Ap = A_t_bwd;
}
perc_vare(gamma1, gamma2, bwd_dominant);
}
pAp = Ap;
for (i=0; i<2; i++) {
Weight_Az(pAp, gamma1[i], m_ap, Ap1);
Weight_Az(pAp, gamma2[i], m_ap, Ap2);
Residue(m_ap, Ap1, &speech[i*L_SUBFR], &wsp[i*L_SUBFR], L_SUBFR);
Syn_filte(m_ap, Ap2, &wsp[i*L_SUBFR], &wsp[i*L_SUBFR], L_SUBFR,
&mem_w[M_BWD-m_ap], 0);
for(j=0; j<M_BWD; j++) mem_w[j] = wsp[i*L_SUBFR+L_SUBFR-M_BWD+j];
pAp += m_ap+1;
}
*ana++ = rate+ (Word16)2; /* bit rate mode */
if(lp_mode == 0) {
m_aq = M;
Aq = A_t_fwd_q;
/* update previous filter for next frame */
Copy(&Aq[MP1], prev_filter, MP1);
for(i=MP1; i <M_BWDP1; i++) prev_filter[i] = 0;
for(j=MP1; j<M_BWDP1; j++) ai_zero[j] = 0;
}
else {
m_aq = M_BWD;
Aq = A_t_bwd;
if (bwd_dominant == 0) {
for(j=MP1; j<M_BWDP1; j++) ai_zero[j] = 0;
}
/* update previous filter for next frame */
Copy(&Aq[M_BWDP1], prev_filter, M_BWDP1);
}
if (rate == G729E) *ana++ = lp_mode;
/*----------------------------------------------------------------------*
* - Find the weighted input speech w_sp[] for the whole speech frame *
* - Find the open-loop pitch delay *
*----------------------------------------------------------------------*/
if( lp_mode == 0) {
Copy(lsp_new_q, lsp_old_q, M);
Lsp_prev_update(freq_cur, freq_prev);
*ana++ = code_lsp[0];
*ana++ = code_lsp[1];
}
/* Find open loop pitch lag */
T_op = Pitch_ol(wsp, PIT_MIN, PIT_MAX, L_FRAME);
/* Range for closed loop pitch search in 1st subframe */
T0_min = sub(T_op, 3);
if (sub(T0_min,PIT_MIN)<0) {
T0_min = PIT_MIN;
}
T0_max = add(T0_min, 6);
if (sub(T0_max ,PIT_MAX)>0)
{
T0_max = PIT_MAX;
T0_min = sub(T0_max, 6);
}
/*------------------------------------------------------------------------*
* Loop for every subframe in the analysis frame *
*------------------------------------------------------------------------*
* To find the pitch and innovation parameters. The subframe size is *
* L_SUBFR and the loop is repeated 2 times. *
* - find the weighted LPC coefficients *
* - find the LPC residual signal res[] *
* - compute the target signal for pitch search *
* - compute impulse response of weighted synthesis filter (h1[]) *
* - find the closed-loop pitch parameters *
* - encode the pitch delay *
* - update the impulse response h1[] by including fixed-gain pitch *
* - find target vector for codebook search *
* - codebook search *
* - encode codebook address *
* - VQ of pitch and codebook gains *
* - find synthesis speech *
* - update states of weighting filter *
*------------------------------------------------------------------------*/
pAp = Ap; /* pointer to interpolated "unquantized"LPC parameters */
pAq = Aq; /* pointer to interpolated "quantized" LPC parameters */
i_gamma = 0;
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR) {
/*---------------------------------------------------------------*
* Find the weighted LPC coefficients for the weighting filter. *
*---------------------------------------------------------------*/
Weight_Az(pAp, gamma1[i_gamma], m_ap, Ap1);
Weight_Az(pAp, gamma2[i_gamma], m_ap, Ap2);
/*---------------------------------------------------------------*
* Compute impulse response, h1[], of weighted synthesis filter *
*---------------------------------------------------------------*/
for (i = 0; i <=m_ap; i++) ai_zero[i] = Ap1[i];
Syn_filte(m_aq, pAq, ai_zero, h1, L_SUBFR, zero, 0);
Syn_filte(m_ap, Ap2, h1, h1, L_SUBFR, zero, 0);
/*------------------------------------------------------------------------*
* *
* Find the target vector for pitch search: *
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ *
* *
* |------| res[n] *
* speech[n]---| A(z) |-------- *
* |------| | |--------| error[n] |------| *
* zero -- (-)--| 1/A(z) |-----------| W(z) |-- target *
* exc |--------| |------| *
* *
* Instead of subtracting the zero-input response of filters from *
* the weighted input speech, the above configuration is used to *
* compute the target vector. This configuration gives better performance *
* with fixed-point implementation. The memory of 1/A(z) is updated by *
* filtering (res[n]-exc[n]) through 1/A(z), or simply by subtracting *
* the synthesis speech from the input speech: *
* error[n] = speech[n] - syn[n]. *
* The memory of W(z) is updated by filtering error[n] through W(z), *
* or more simply by subtracting the filtered adaptive and fixed *
* codebook excitations from the target: *
* target[n] - gain_pit*y1[n] - gain_code*y2[n] *
* as these signals are already available. *
* *
*------------------------------------------------------------------------*/
Residue(m_aq, pAq, &speech[i_subfr], &exc[i_subfr], L_SUBFR); /* LPC residual */
for (i=0; i<L_SUBFR; i++) res2[i] = exc[i_subfr+i];
Syn_filte(m_aq, pAq, &exc[i_subfr], error, L_SUBFR,
&mem_err[M_BWD-m_aq], 0);
Residue(m_ap, Ap1, error, xn, L_SUBFR);
Syn_filte(m_ap, Ap2, xn, xn, L_SUBFR, &mem_w0[M_BWD-m_ap], 0); /* target signal xn[]*/
/*----------------------------------------------------------------------*
* Closed-loop fractional pitch search *
*----------------------------------------------------------------------*/
T0 = Pitch_fr3cp(&exc[i_subfr], xn, h1, L_SUBFR, T0_min, T0_max,
i_subfr, &T0_frac, rate);
index = Enc_lag3cp(T0, T0_frac, &T0_min, &T0_max,PIT_MIN,PIT_MAX,
i_subfr, rate);
*ana++ = index;
if ( (i_subfr == 0) && (rate != G729D) ) {
*ana = Parity_Pitch(index);
if( rate == G729E) {
*ana ^= (shr(index, 1) & 0x0001);
}
ana++;
}
/*-----------------------------------------------------------------*
* - find unity gain pitch excitation (adaptive codebook entry) *
* with fractional interpolation. *
* - find filtered pitch exc. y1[]=exc[] convolve with h1[]) *
* - compute pitch gain and limit between 0 and 1.2 *
* - update target vector for codebook search *
* - find LTP residual. *
*-----------------------------------------------------------------*/
Pred_lt_3(&exc[i_subfr], T0, T0_frac, L_SUBFR);
Convolve(&exc[i_subfr], h1, y1, L_SUBFR);
gain_pit = G_pitch(xn, y1, g_coeff, L_SUBFR);
/* clip pitch gain if taming is necessary */
taming = test_err(T0, T0_frac);
if( taming == 1){
if (sub(gain_pit, GPCLIP) > 0) {
gain_pit = GPCLIP;
}
}
/* xn2[i] = xn[i] - y1[i] * gain_pit */
for (i = 0; i < L_SUBFR; i++) {
L_temp = L_mult(y1[i], gain_pit);
L_temp = L_shl(L_temp, 1); /* gain_pit in Q14 */
xn2[i] = sub(xn[i], extract_h(L_temp));
}
/*-----------------------------------------------------*
* - Innovative codebook search. *
*-----------------------------------------------------*/
switch (rate) {
case G729: /* 8 kbit/s */
{
/* case 8 kbit/s */
index = ACELP_Codebook(xn2, h1, T0, sharp, i_subfr, code, y2, &i);
*ana++ = index; /* Positions index */
*ana++ = i; /* Signs index */
break;
}
case G729D: /* 6.4 kbit/s */
{
index = ACELP_CodebookD(xn2, h1, T0, sharp, code, y2, &i);
*ana++ = index; /* Positions index */
*ana++ = i; /* Signs index */
break;
}
case G729E: /* 11.8 kbit/s */
{
/*-----------------------------------------------------------------*
* Include fixed-gain pitch contribution into impulse resp. h[] *
*-----------------------------------------------------------------*/
pit_sharp = shl(sharp, 1); /* From Q14 to Q15 */
if(T0 < L_SUBFR) {
for (i = T0; i < L_SUBFR; i++){ /* h[i] += pitch_sharp*h[i-T0] */
h1[i] = add(h1[i], mult(h1[i-T0], pit_sharp));
}
}
/* calculate residual after long term prediction */
/* res2[i] -= exc[i+i_subfr] * gain_pit */
for (i = 0; i < L_SUBFR; i++) {
L_temp = L_mult(exc[i+i_subfr], gain_pit);
L_temp = L_shl(L_temp, 1); /* gain_pit in Q14 */
res2[i] = sub(res2[i], extract_h(L_temp));
}
if (lp_mode == 0) ACELP_10i40_35bits(xn2, res2, h1, code, y2, ana); /* Forward */
else ACELP_12i40_44bits(xn2, res2, h1, code, y2, ana); /* Backward */
ana += 5;
/*-----------------------------------------------------------------*
* Include fixed-gain pitch contribution into code[]. *
*-----------------------------------------------------------------*/
if(T0 < L_SUBFR) {
for (i = T0; i < L_SUBFR; i++) { /* code[i] += pitch_sharp*code[i-T0] */
code[i] = add(code[i], mult(code[i-T0], pit_sharp));
}
}
break;
}
default : {
printf("Unrecognized bit rate\n");
exit(-1);
}
} /* end of switch */
/*-----------------------------------------------------*
* - Quantization of gains. *
*-----------------------------------------------------*/
g_coeff_cs[0] = g_coeff[0]; /* <y1,y1> */
exp_g_coeff_cs[0] = negate(g_coeff[1]); /* Q-Format:XXX -> JPN */
g_coeff_cs[1] = negate(g_coeff[2]); /* (xn,y1) -> -2<xn,y1> */
exp_g_coeff_cs[1] = negate(add(g_coeff[3], 1)); /* Q-Format:XXX -> JPN */
Corr_xy2( xn, y1, y2, g_coeff_cs, exp_g_coeff_cs ); /* Q0 Q0 Q12 ^Qx ^Q0 */
/* g_coeff_cs[3]:exp_g_coeff_cs[3] = <y2,y2> */
/* g_coeff_cs[4]:exp_g_coeff_cs[4] = -2<xn,y2> */
/* g_coeff_cs[5]:exp_g_coeff_cs[5] = 2<y1,y2> */
if (rate == G729D)
index = Qua_gain_6k(code, g_coeff_cs, exp_g_coeff_cs, L_SUBFR,
&gain_pit, &gain_code, taming, past_qua_en);
else
index = Qua_gain_8k(code, g_coeff_cs, exp_g_coeff_cs, L_SUBFR,
&gain_pit, &gain_code, taming, past_qua_en);
*ana++ = index;
/*------------------------------------------------------------*
* - Update pitch sharpening "sharp" with quantized gain_pit *
*------------------------------------------------------------*/
sharp = gain_pit;
if (sub(sharp, SHARPMAX) > 0) sharp = SHARPMAX;
else {
if (sub(sharp, SHARPMIN) < 0) sharp = SHARPMIN;
}
/*------------------------------------------------------*
* - Find the total excitation *
* - find synthesis speech corresponding to exc[] *
* - update filters memories for finding the target *
* vector in the next subframe *
* (update error[-m..-1] and mem_w_err[]) *
* update error function for taming process *
*------------------------------------------------------*/
for (i = 0; i < L_SUBFR; i++) {
/* exc[i] = gain_pit*exc[i] + gain_code*code[i]; */
/* exc[i] in Q0 gain_pit in Q14 */
/* code[i] in Q13 gain_cod in Q1 */
L_temp = L_mult(exc[i+i_subfr], gain_pit);
L_temp = L_mac(L_temp, code[i], gain_code);
L_temp = L_shl(L_temp, 1);
exc[i+i_subfr] = round(L_temp);
}
update_exc_err(gain_pit, T0);
Syn_filte(m_aq, pAq, &exc[i_subfr], &synth_ptr[i_subfr], L_SUBFR,
&mem_syn[M_BWD-m_aq], 0);
for(j=0; j<M_BWD; j++) mem_syn[j] = synth_ptr[i_subfr+L_SUBFR-M_BWD+j];
for (i = L_SUBFR-M_BWD, j = 0; i < L_SUBFR; i++, j++) {
mem_err[j] = sub(speech[i_subfr+i], synth_ptr[i_subfr+i]);
temp = extract_h(L_shl( L_mult(y1[i], gain_pit), 1) );
k = extract_h(L_shl( L_mult(y2[i], gain_code), 2) );
mem_w0[j] = sub(xn[i], add(temp, k));
}
pAp += m_ap+1;
pAq += m_aq+1;
i_gamma = add(i_gamma,1);
}
/*--------------------------------------------------*
* Update signal for next frame. *
* -> shift to the left by L_FRAME: *
* speech[], wsp[] and exc[] *
*--------------------------------------------------*/
Copy(&old_speech[L_FRAME], &old_speech[0], L_TOTAL-L_FRAME);
Copy(&old_wsp[L_FRAME], &old_wsp[0], PIT_MAX);
Copy(&old_exc[L_FRAME], &old_exc[0], PIT_MAX+L_INTERPOL);
prev_lp_mode = lp_mode;
return;
}
