/*
**************************************************************************
* Function: Post_Filter
* Purpose: postfiltering of synthesis speech.
* Description:
* The postfiltering process is described as follows:
*
* - inverse filtering of syn[] through A(z/0.7) to get res2[]
* - tilt compensation filtering; 1 - MU*k*z^-1
* - synthesis filtering through 1/A(z/0.75)
* - adaptive gain control
*
**************************************************************************
*/
int Post_Filter (
Post_FilterState *st, /* i/o : post filter states */
enum Mode mode, /* i : AMR mode */
Word16 *syn, /* i/o : synthesis speech (postfiltered is output) */
Word16 *Az_4 /* i : interpolated LPC parameters in all subfr. */
)
{
/*-------------------------------------------------------------------*
* Declaration of parameters *
*-------------------------------------------------------------------*/
Word16 Ap3[MP1], Ap4[MP1]; /* bandwidth expanded LP parameters */
Word16 *Az; /* pointer to Az_4: */
/* LPC parameters in each subframe */
Word16 i_subfr; /* index for beginning of subframe */
Word16 h[L_H];
Word16 i;
Word16 temp1, temp2;
Word32 L_tmp;
Word16 *syn_work = &st->synth_buf[M];
move16 ();
/*-----------------------------------------------------*
* Post filtering *
*-----------------------------------------------------*/
Copy (syn, syn_work , L_FRAME);
Az = Az_4;
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
{
/* Find weighted filter coefficients Ap3[] and ap[4] */
test ();
test ();
if (sub(mode, MR122) == 0 || sub(mode, MR102) == 0)
{
Weight_Ai (Az, gamma3_MR122, Ap3);
Weight_Ai (Az, gamma4_MR122, Ap4);
}
else
{
Weight_Ai (Az, gamma3, Ap3);
Weight_Ai (Az, gamma4, Ap4);
}
/* filtering of synthesis speech by A(z/0.7) to find res2[] */
Residu (Ap3, &syn_work[i_subfr], st->res2, L_SUBFR);
/* tilt compensation filter */
/* impulse response of A(z/0.7)/A(z/0.75) */
Copy (Ap3, h, M + 1);
Set_zero (&h[M + 1], L_H - M - 1);
Syn_filt (Ap4, h, h, L_H, &h[M + 1], 0);
/* 1st correlation of h[] */
L_tmp = L_mult (h[0], h[0]);
for (i = 1; i < L_H; i++)
{
L_tmp = L_mac (L_tmp, h[i], h[i]);
}
temp1 = extract_h (L_tmp);
L_tmp = L_mult (h[0], h[1]);
for (i = 1; i < L_H - 1; i++)
{
L_tmp = L_mac (L_tmp, h[i], h[i + 1]);
}
temp2 = extract_h (L_tmp);
test ();
if (temp2 <= 0)
{
temp2 = 0;
move16 ();
}
else
{
temp2 = mult (temp2, MU);
temp2 = div_s (temp2, temp1);
}
preemphasis (st->preemph_state, st->res2, temp2, L_SUBFR);
/* filtering through 1/A(z/0.75) */
Syn_filt (Ap4, st->res2, &syn[i_subfr], L_SUBFR, st->mem_syn_pst, 1);
/* scale output to input */
agc (st->agc_state, &syn_work[i_subfr], &syn[i_subfr],
AGC_FAC, L_SUBFR);
Az += MP1;
}
/* update syn_work[] buffer */
Copy (&syn_work[L_FRAME - M], &syn_work[-M], M);
return 0;
}
void Post_Filter(
Word16 *syn, /* in/out: synthesis speech (postfiltered is output) */
Word16 *Az_4, /* input : interpolated LPC parameters in all subframes */
Word16 *T, /* input : decoded pitch lags in all subframes */
Word16 Vad
)
{
/*-------------------------------------------------------------------*
* Declaration of parameters *
*-------------------------------------------------------------------*/
Word16 res2_pst[L_SUBFR]; /* res2[] after pitch postfiltering */
Word16 syn_pst[L_FRAME]; /* post filtered synthesis speech */
Word16 Ap3[MP1], Ap4[MP1]; /* bandwidth expanded LP parameters */
Word16 *Az; /* pointer to Az_4: */
/* LPC parameters in each subframe */
Word16 t0_max, t0_min; /* closed-loop pitch search range */
Word16 i_subfr; /* index for beginning of subframe */
Word16 h[L_H];
Word16 i, j;
Word16 temp1, temp2;
Word32 L_tmp;
postfilt_type* ppost_filt = pg729dec->ppost_filt;
/*-----------------------------------------------------*
* Post filtering *
*-----------------------------------------------------*/
Az = Az_4;
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
{
/* Find pitch range t0_min - t0_max */
t0_min = sub(*T++, 3);
t0_max = add(t0_min, 6);
if (sub(t0_max, PIT_MAX) > 0) {
t0_max = PIT_MAX;
t0_min = sub(t0_max, 6);
}
/* Find weighted filter coefficients Ap3[] and ap[4] */
Weight_Az(Az, GAMMA2_PST, M, Ap3);
Weight_Az(Az, GAMMA1_PST, M, Ap4);
/* filtering of synthesis speech by A(z/GAMMA2_PST) to find res2[] */
Residu(Ap3, &syn[i_subfr], ppost_filt->res2, L_SUBFR);
/* scaling of "res2[]" to avoid energy overflow */
for (j=0; j<L_SUBFR; j++)
{
ppost_filt->scal_res2[j] = shr(ppost_filt->res2[j], 2);
}
/* pitch postfiltering */
if (sub(Vad, 1) == 0)
pit_pst_filt(ppost_filt->res2, ppost_filt->scal_res2, t0_min, t0_max, L_SUBFR, res2_pst);
else
for (j=0; j<L_SUBFR; j++)
res2_pst[j] = ppost_filt->res2[j];
/* tilt compensation filter */
/* impulse response of A(z/GAMMA2_PST)/A(z/GAMMA1_PST) */
Copy(Ap3, h, M+1);
Set_zero(&h[M+1], L_H-M-1);
Syn_filt(Ap4, h, h, L_H, &h[M+1], 0);
/* 1st correlation of h[] */
L_tmp = L_mult(h[0], h[0]);
for (i=1; i<L_H; i++) L_tmp = L_mac(L_tmp, h[i], h[i]);
temp1 = extract_h(L_tmp);
L_tmp = L_mult(h[0], h[1]);
for (i=1; i<L_H-1; i++) L_tmp = L_mac(L_tmp, h[i], h[i+1]);
temp2 = extract_h(L_tmp);
if(temp2 <= 0) {
temp2 = 0;
}
else {
temp2 = mult(temp2, MU);
temp2 = div_s(temp2, temp1);
}
preemphasis(res2_pst, temp2, L_SUBFR);
/* filtering through 1/A(z/GAMMA1_PST) */
Syn_filt(Ap4, res2_pst, &syn_pst[i_subfr], L_SUBFR, ppost_filt->mem_syn_pst, 1);
/* scale output to input */
agc(&syn[i_subfr], &syn_pst[i_subfr], L_SUBFR);
/* update res2[] buffer; shift by L_SUBFR */
Copy(&ppost_filt->res2[L_SUBFR-PIT_MAX], &ppost_filt->res2[-PIT_MAX], PIT_MAX);
Copy(&ppost_filt->scal_res2[L_SUBFR-PIT_MAX], &ppost_filt->scal_res2[-PIT_MAX], PIT_MAX);
Az += MP1;
}
/* update syn[] buffer */
Copy(&syn[L_FRAME-M], &syn[-M], M);
/* overwrite synthesis speech by postfiltered synthesis speech */
Copy(syn_pst, syn, L_FRAME);
return;
}
void Post_Filter (
INT16 *syn, /* in/out: synthesis speech (postfiltered is output) */
INT16 *Az_4 /* input: interpolated LPC parameters in all subframes */
)
{
/*-------------------------------------------------------------------*
* Declaration of parameters *
*-------------------------------------------------------------------*/
INT16 syn_pst[L_FRAME]; /* post filtered synthesis speech */
INT16 Ap3[MP1], Ap4[MP1]; /* bandwidth expanded LP parameters */
INT16 *Az; /* pointer to Az_4: */
/* LPC parameters in each subframe */
INT16 i_subfr; /* index for beginning of subframe */
INT16 h[L_H];
INT16 i;
INT16 temp1, temp2;
//INT32 L_tmp;
register INT32 tmp_hi=0;
register UINT32 tmp_lo=0;
VPP_EFR_PROFILE_FUNCTION_ENTER(Post_Filter);
/*-----------------------------------------------------*
* Post filtering *
*-----------------------------------------------------*/
Az = Az_4;
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
{
/* Find weighted filter coefficients Ap3[] and ap[4] */
Weight_Ai (Az, F_gamma3, Ap3);
Weight_Ai (Az, F_gamma4, Ap4);
/* filtering of synthesis speech by A(z/0.7) to find res2[] */
Residu (Ap3, &syn[i_subfr], res2, L_SUBFR);
/* tilt compensation filter */
/* impulse response of A(z/0.7)/A(z/0.75) */
Copy (Ap3, h, M + 1);
//Set_zero (&h[M + 1], L_H - M - 1);
memset ((INT8*)&h[M + 1], 0, (L_H - M - 1)<<1);
Syn_filt (Ap4, h, h, L_H, &h[M + 1], 0);
/* 1st correlation of h[] */
//L_tmp = L_MULT(h[0], h[0]);
VPP_MLX16(tmp_hi,tmp_lo, h[0], h[0]);
for (i = 1; i < L_H; i++)
{
//L_tmp = L_MAC(L_tmp, h[i], h[i]);
VPP_MLA16(tmp_hi,tmp_lo, h[i], h[i]);
}
//temp1 = extract_h (L_tmp);
//temp1 = EXTRACT_H(VPP_SCALE64_TO_16(tmp_hi,tmp_lo));
temp1 = L_SHR_D((INT32)tmp_lo, 15);
//L_tmp = L_MULT(h[0], h[1]);
VPP_MLX16(tmp_hi,tmp_lo, h[0], h[1]);
for (i = 1; i < L_H - 1; i++)
{
//L_tmp = L_MAC(L_tmp, h[i], h[i + 1]);
VPP_MLA16(tmp_hi,tmp_lo, h[i], h[i + 1]);
}
//temp2 = extract_h (L_tmp);
//temp2 = EXTRACT_H(VPP_SCALE64_TO_16(tmp_hi,tmp_lo));
temp2 = L_SHR_D((INT32)tmp_lo, 15);
if (temp2 <= 0)
{
temp2 = 0;
}
else
{
//temp2 = mult (temp2, MU);
temp2 = MULT(temp2, MU);
temp2 = div_s (temp2, temp1);
}
preemphasis (res2, temp2, L_SUBFR);
/* filtering through 1/A(z/0.75) */
Syn_filt (Ap4, res2, &syn_pst[i_subfr], L_SUBFR, mem_syn_pst, 1);
/* scale output to input */
agc (&syn[i_subfr], &syn_pst[i_subfr], AGC_FAC, L_SUBFR);
Az += MP1;
}
/* update syn[] buffer */
Copy (&syn[L_FRAME - M], &syn[-M], M);
/* overwrite synthesis speech by postfiltered synthesis speech */
Copy (syn_pst, syn, L_FRAME);
VPP_EFR_PROFILE_FUNCTION_EXIT(Post_Filter);
return;
}
