double cAppliOptimTriplet::ResiduTriplet(const ElRotation3D & aR1,const ElRotation3D & aR2,const ElRotation3D & aR3)
{
std::vector<double> aVRes;
for (int aK=0 ; aK<int(mIm1->VFullPtOf3().size()) ; aK++)
{
std::vector<Pt3dr> aW1;
std::vector<Pt3dr> aW2;
AddSegOfRot(aW1,aW2,aR1,mIm1->VFullPtOf3()[aK]);
AddSegOfRot(aW1,aW2,aR2,mIm2->VFullPtOf3()[aK]);
AddSegOfRot(aW1,aW2,aR3,mIm3->VFullPtOf3()[aK]);
bool OkI;
Pt3dr aI = InterSeg(aW1,aW2,OkI);
if (OkI)
{
double aRes1 = Residu(mIm1->Im(),aR1,aI,mIm1->VFullPtOf3()[aK]);
double aRes2 = Residu(mIm2->Im(),aR2,aI,mIm2->VFullPtOf3()[aK]);
double aRes3 = Residu(mIm3->Im(),aR3,aI,mIm3->VFullPtOf3()[aK]);
/*
double aRes2 = Residu(mIm2->Im(),R2(),aI,mVP2[aK]);
*/
aVRes.push_back((aRes1+aRes2+aRes3)/3.0);
}
}
return MedianeSup(aVRes);
}
REAL SystLinSurResolu::Residu(Im1D_REAL8 anIm,INT iEq) const
{
AssertIndexEqValide(iEq);
AssertIndexGoodNbVar(anIm.tx());
return Residu(anIm.data(),iEq);
}
double cPairOfTriplet::ResiduMoy(const ElRotation3D & aR1,const ElRotation3D & aR2)
{
std::vector<double> aVRes;
for (int aK=0 ; aK<int(mFullVP1.size()) ; aK++)
{
std::vector<Pt3dr> aW1;
std::vector<Pt3dr> aW2;
AddSegOfRot(aW1,aW2,aR1,mFullVP1[aK]);
AddSegOfRot(aW1,aW2,aR2,mFullVP2[aK]);
bool OkI;
Pt3dr aI = InterSeg(aW1,aW2,OkI);
if (OkI)
{
double aRes1 = Residu(mIm1->Im(),aR1,aI,mFullVP1[aK]);
double aRes2 = Residu(mIm2->Im(),aR2,aI,mFullVP2[aK]);
aVRes.push_back((aRes1+aRes2)/2.0);
}
}
return MedianeSup(aVRes);
}
void pre_big(
enum Mode mode, /* i : coder mode */
const Word16 gamma1[], /* i : spectral exp. factor 1 */
const Word16 gamma1_12k2[],/* i : spectral exp. factor 1 for EFR */
const Word16 gamma2[], /* i : spectral exp. factor 2 */
Word16 A_t[], /* i : A(z) unquantized, for 4 subframes, Q12 */
Word16 frameOffset, /* i : Start position in speech vector, Q0 */
Word16 speech[], /* i : speech, Q0 */
Word16 mem_w[], /* i/o: synthesis filter memory state, Q0 */
Word16 wsp[], /* o : weighted speech Q0 */
Flag *pOverflow /* o : overflow indicator */
)
{
Word16 Ap1[MP1]; /* A(z) with spectral expansion */
Word16 Ap2[MP1]; /* A(z) with spectral expansion */
const Word16 *g1; /* Pointer to correct gammma1 vector */
Word16 aOffset;
Word16 i;
if (mode <= MR795)
{
g1 = gamma1;
}
else
{
g1 = gamma1_12k2;
}
if (frameOffset > 0)
{
aOffset = 2 * MP1;
}
else
{
aOffset = 0;
}
/* process two subframes (which form the "big" subframe) */
for (i = 0; i < 2; i++)
{
Weight_Ai(&A_t[aOffset], g1, Ap1);
Weight_Ai(&A_t[aOffset], gamma2, Ap2);
Residu(Ap1, &speech[frameOffset], &wsp[frameOffset], L_SUBFR);
Syn_filt(Ap2, &wsp[frameOffset], &wsp[frameOffset], L_SUBFR, mem_w, 1);
aOffset = add(aOffset, MP1, pOverflow);
frameOffset = add(frameOffset, L_SUBFR, pOverflow);
}
return;
}
/*----------------------------------------------------------------------------
* Post - adaptive postfilter main function
*----------------------------------------------------------------------------
*/
void Post(
Word16 t0, /* input : pitch delay given by coder */
Word16 *signal_ptr, /* input : input signal (pointer to current subframe */
Word16 *coeff, /* input : LPC coefficients for current subframe */
Word16 *sig_out, /* output: postfiltered output */
Word16 *vo, /* output: voicing decision 0 = uv, > 0 delay */
Word16 Vad /* input : frame type */
)
{
/* Local variables and arrays */
Word16 apond1[MP1]; /* s.t. denominator coeff. */
Word16 sig_ltp[L_SUBFRP1]; /* H0 output signal */
Word16 *sig_ltp_ptr;
Word16 parcor0;
/* Compute weighted LPC coefficients */
Weight_Az(coeff, GAMMA1_PST, M, apond1);
Weight_Az(coeff, GAMMA2_PST, M, apond2);
/* Compute A(gamma2) residual */
Residu(apond2, signal_ptr, res2_ptr, L_SUBFR);
/* Harmonic filtering */
sig_ltp_ptr = sig_ltp + 1;
if (sub(Vad, 1) == 0)
pst_ltp(t0, res2_ptr, sig_ltp_ptr, vo);
else {
*vo = 0;
Copy(res2_ptr, sig_ltp_ptr, L_SUBFR);
}
/* Save last output of 1/A(gamma1) */
/* (from preceding subframe) */
sig_ltp[0] = *ptr_mem_stp;
/* Controls short term pst filter gain and compute parcor0 */
calc_st_filt(apond2, apond1, &parcor0, sig_ltp_ptr);
/* 1/A(gamma1) filtering, mem_stp is updated */
Syn_filt(apond1, sig_ltp_ptr, sig_ltp_ptr, L_SUBFR, mem_stp, 1);
/* Tilt filtering */
filt_mu(sig_ltp, sig_out, parcor0);
/* Gain control */
scale_st(signal_ptr, sig_out, &gain_prec);
/**** Update for next subframe */
Copy(&res2[L_SUBFR], &res2[0], MEM_RES2);
return;
}
REAL SystLinSurResolu::L2SomResiduPond(Im1D_REAL8 aPt) const
{
AssertIndexGoodNbVar(aPt.tx());
REAL aRes = 0.0;
REAL *aDataP = aPt.data();
for (INT iEq=0 ; iEq<mNbEqCur ; iEq++)
aRes += mDataPds[iEq] * ElSquare(Residu(aDataP,iEq));
return aRes;
}
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 Coder_ld8a(
Word16 ana[], /* output : Analysis parameters */
Word16 frame, /* input : frame counter */
Word16 vad_enable /* input : VAD enable flag */
)
{
/* LPC analysis */
Word16 Aq_t[(MP1)*2]; /* A(z) quantized for the 2 subframes */
Word16 Ap_t[(MP1)*2]; /* A(z/gamma) for the 2 subframes */
Word16 *Aq, *Ap; /* Pointer on Aq_t and Ap_t */
/* 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 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 temp, taming;
Word32 L_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) *
*------------------------------------------------------------------------*/
{
/* Temporary vectors */
Word16 r_l[NP+1], r_h[NP+1]; /* Autocorrelations low and hi */
Word16 rc[M]; /* Reflection coefficients. */
Word16 lsp_new[M], lsp_new_q[M]; /* LSPs at 2th subframe */
/* For G.729B */
Word16 rh_nbe[MP1];
Word16 lsf_new[M];
Word16 lsfq_mem[MA_NP][M];
Word16 exp_R0, Vad;
/* LP analysis */
Autocorr(p_window, NP, r_h, r_l, &exp_R0); /* Autocorrelations */
Copy(r_h, rh_nbe, MP1);
Lag_window(NP, r_h, r_l); /* Lag windowing */
Levinson(r_h, r_l, Ap_t, rc, &temp); /* Levinson Durbin */
Az_lsp(Ap_t, lsp_new, lsp_old); /* From A(z) to lsp */
/* For G.729B */
/* ------ VAD ------- */
Lsp_lsf(lsp_new, lsf_new, M);
vad(rc[1], lsf_new, r_h, r_l, exp_R0, p_window, frame,
pastVad, ppastVad, &Vad);
Update_cng(rh_nbe, exp_R0, Vad);
/* ---------------------- */
/* Case of Inactive frame */
/* ---------------------- */
if ((Vad == 0) && (vad_enable == 1)){
Get_freq_prev(lsfq_mem);
Cod_cng(exc, pastVad, lsp_old_q, Aq_t, ana, lsfq_mem, &seed);
Update_freq_prev(lsfq_mem);
ppastVad = pastVad;
pastVad = Vad;
/* Update wsp, mem_w and mem_w0 */
Aq = Aq_t;
for(i_subfr=0; i_subfr < L_FRAME; i_subfr += L_SUBFR) {
/* Residual signal in xn */
Residu(Aq, &speech[i_subfr], xn, L_SUBFR);
Weight_Az(Aq, GAMMA1, M, Ap_t);
/* Compute wsp and mem_w */
Ap = Ap_t + MP1;
Ap[0] = 4096;
for(i=1; i<=M; i++) /* Ap[i] = Ap_t[i] - 0.7 * Ap_t[i-1]; */
Ap[i] = sub(Ap_t[i], mult(Ap_t[i-1], 22938));
Syn_filt(Ap, xn, &wsp[i_subfr], L_SUBFR, mem_w, 1);
/* Compute mem_w0 */
for(i=0; i<L_SUBFR; i++) {
xn[i] = sub(xn[i], exc[i_subfr+i]); /* residu[] - exc[] */
}
Syn_filt(Ap_t, xn, xn, L_SUBFR, mem_w0, 1);
Aq += MP1;
}
sharp = SHARPMIN;
/* Update memories for next frames */
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);
return;
} /* End of inactive frame case */
/* -------------------- */
/* Case of Active frame */
/* -------------------- */
/* Case of active frame */
*ana++ = 1;
seed = INIT_SEED;
ppastVad = pastVad;
pastVad = Vad;
/* LSP quantization */
Qua_lsp(lsp_new, lsp_new_q, ana);
ana += 2; /* Advance analysis parameters pointer */
/*--------------------------------------------------------------------*
* Find interpolated LPC parameters in all subframes *
* The interpolated parameters are in array Aq_t[]. *
*--------------------------------------------------------------------*/
Int_qlpc(lsp_old_q, lsp_new_q, Aq_t);
/* Compute A(z/gamma) */
Weight_Az(&Aq_t[0], GAMMA1, M, &Ap_t[0]);
Weight_Az(&Aq_t[MP1], GAMMA1, M, &Ap_t[MP1]);
/* update the LSPs for the next frame */
Copy(lsp_new, lsp_old, M);
Copy(lsp_new_q, lsp_old_q, M);
}
/*----------------------------------------------------------------------*
* - Find the weighted input speech w_sp[] for the whole speech frame *
* - Find the open-loop pitch delay *
*----------------------------------------------------------------------*/
Residu(&Aq_t[0], &speech[0], &exc[0], L_SUBFR);
Residu(&Aq_t[MP1], &speech[L_SUBFR], &exc[L_SUBFR], L_SUBFR);
{
Word16 Ap1[MP1];
Ap = Ap_t;
Ap1[0] = 4096;
for(i=1; i<=M; i++) /* Ap1[i] = Ap[i] - 0.7 * Ap[i-1]; */
Ap1[i] = sub(Ap[i], mult(Ap[i-1], 22938));
Syn_filt(Ap1, &exc[0], &wsp[0], L_SUBFR, mem_w, 1);
Ap += MP1;
for(i=1; i<=M; i++) /* Ap1[i] = Ap[i] - 0.7 * Ap[i-1]; */
Ap1[i] = sub(Ap[i], mult(Ap[i-1], 22938));
Syn_filt(Ap1, &exc[L_SUBFR], &wsp[L_SUBFR], L_SUBFR, mem_w, 1);
}
/* Find open loop pitch lag */
T_op = Pitch_ol_fast(wsp, 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 *
* - find target vector for codebook search *
* - codebook search *
* - VQ of pitch and codebook gains *
* - update states of weighting filter *
*------------------------------------------------------------------------*/
Aq = Aq_t; /* pointer to interpolated quantized LPC parameters */
Ap = Ap_t; /* pointer to weighted LPC coefficients */
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
{
/*---------------------------------------------------------------*
* Compute impulse response, h1[], of weighted synthesis filter *
*---------------------------------------------------------------*/
h1[0] = 4096;
Set_zero(&h1[1], L_SUBFR-1);
Syn_filt(Ap, h1, h1, L_SUBFR, &h1[1], 0);
/*----------------------------------------------------------------------*
* Find the target vector for pitch search: *
*----------------------------------------------------------------------*/
Syn_filt(Ap, &exc[i_subfr], xn, L_SUBFR, mem_w0, 0);
/*---------------------------------------------------------------------*
* Closed-loop fractional pitch search *
*---------------------------------------------------------------------*/
T0 = Pitch_fr3_fast(&exc[i_subfr], xn, h1, L_SUBFR, T0_min, T0_max,
i_subfr, &T0_frac);
index = Enc_lag3(T0, T0_frac, &T0_min, &T0_max,PIT_MIN,PIT_MAX,i_subfr);
*ana++ = index;
if (i_subfr == 0) {
*ana++ = Parity_Pitch(index);
}
/*-----------------------------------------------------------------*
* - find filtered pitch exc *
* - compute pitch gain and limit between 0 and 1.2 *
* - update target vector for codebook search *
*-----------------------------------------------------------------*/
Syn_filt(Ap, &exc[i_subfr], y1, L_SUBFR, mem_zero, 0);
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. *
*-----------------------------------------------------*/
index = ACELP_Code_A(xn2, h1, T0, sharp, code, y2, &i);
*ana++ = index; /* Positions index */
*ana++ = i; /* Signs index */
/*-----------------------------------------------------*
* - 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> */
*ana++ = Qua_gain(code, g_coeff_cs, exp_g_coeff_cs,
L_SUBFR, &gain_pit, &gain_code, taming);
/*------------------------------------------------------------*
* - Update pitch sharpening "sharp" with quantized gain_pit *
*------------------------------------------------------------*/
sharp = gain_pit;
if (sub(sharp, SHARPMAX) > 0) { sharp = SHARPMAX; }
if (sub(sharp, SHARPMIN) < 0) { sharp = SHARPMIN; }
/*------------------------------------------------------*
* - Find the total excitation *
* - update filters memories for finding the target *
* vector in the next subframe *
*------------------------------------------------------*/
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);
for (i = L_SUBFR-M, j = 0; i < L_SUBFR; i++, j++)
{
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));
}
Aq += MP1; /* interpolated LPC parameters for next subframe */
Ap += MP1;
}
/*--------------------------------------------------*
* 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);
return;
}
/*
**************************************************************************
* 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 Coder_ld8a(
g729a_encoder_state *state,
Word16 ana[] /* output : Analysis parameters */
)
{
/* LPC analysis */
Word16 Aq_t[(MP1)*2]; /* A(z) quantized for the 2 subframes */
Word16 Ap_t[(MP1)*2]; /* A(z/gamma) for the 2 subframes */
Word16 *Aq, *Ap; /* Pointer on Aq_t and Ap_t */
/* 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 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 temp, taming;
Word32 L_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) *
*------------------------------------------------------------------------*/
{
/* Temporary vectors */
Word16 r_l[MP1], r_h[MP1]; /* Autocorrelations low and hi */
Word16 rc[M]; /* Reflection coefficients. */
Word16 lsp_new[M], lsp_new_q[M]; /* LSPs at 2th subframe */
/* LP analysis */
Autocorr(state->p_window, M, r_h, r_l); /* Autocorrelations */
Lag_window(M, r_h, r_l); /* Lag windowing */
Levinson(r_h, r_l, Ap_t, rc); /* Levinson Durbin */
Az_lsp(Ap_t, lsp_new, state->lsp_old); /* From A(z) to lsp */
/* LSP quantization */
Qua_lsp(state, lsp_new, lsp_new_q, ana);
ana += 2; /* Advance analysis parameters pointer */
/*--------------------------------------------------------------------*
* Find interpolated LPC parameters in all subframes *
* The interpolated parameters are in array Aq_t[]. *
*--------------------------------------------------------------------*/
Int_qlpc(state->lsp_old_q, lsp_new_q, Aq_t);
/* Compute A(z/gamma) */
Weight_Az(&Aq_t[0], GAMMA1, M, &Ap_t[0]);
Weight_Az(&Aq_t[MP1], GAMMA1, M, &Ap_t[MP1]);
/* update the LSPs for the next frame */
Copy(lsp_new, state->lsp_old, M);
Copy(lsp_new_q, state->lsp_old_q, M);
}
/*----------------------------------------------------------------------*
* - Find the weighted input speech w_sp[] for the whole speech frame *
* - Find the open-loop pitch delay *
*----------------------------------------------------------------------*/
Residu(&Aq_t[0], &(state->speech[0]), &(state->exc[0]), L_SUBFR);
Residu(&Aq_t[MP1], &(state->speech[L_SUBFR]), &(state->exc[L_SUBFR]), L_SUBFR);
{
Word16 Ap1[MP1];
Ap = Ap_t;
Ap1[0] = 4096;
for(i=1; i<=M; i++) /* Ap1[i] = Ap[i] - 0.7 * Ap[i-1]; */
Ap1[i] = sub(Ap[i], mult(Ap[i-1], 22938));
Syn_filt(Ap1, &(state->exc[0]), &(state->wsp[0]), L_SUBFR, state->mem_w, 1);
Ap += MP1;
for(i=1; i<=M; i++) /* Ap1[i] = Ap[i] - 0.7 * Ap[i-1]; */
Ap1[i] = sub(Ap[i], mult(Ap[i-1], 22938));
Syn_filt(Ap1, &(state->exc[L_SUBFR]), &(state->wsp[L_SUBFR]), L_SUBFR, state->mem_w, 1);
}
/* Find open loop pitch lag */
T_op = Pitch_ol_fast(state->wsp, PIT_MAX, L_FRAME);
/* Range for closed loop pitch search in 1st subframe */
T0_min = T_op - 3;
T0_max = T0_min + 6;
if (T0_min < PIT_MIN)
{
T0_min = PIT_MIN;
T0_max = PIT_MIN + 6;
}
else if (T0_max > PIT_MAX)
{
T0_max = PIT_MAX;
T0_min = PIT_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 *
* - find target vector for codebook search *
* - codebook search *
* - VQ of pitch and codebook gains *
* - update states of weighting filter *
*------------------------------------------------------------------------*/
Aq = Aq_t; /* pointer to interpolated quantized LPC parameters */
Ap = Ap_t; /* pointer to weighted LPC coefficients */
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
{
/*---------------------------------------------------------------*
* Compute impulse response, h1[], of weighted synthesis filter *
*---------------------------------------------------------------*/
h1[0] = 4096;
Set_zero(&h1[1], L_SUBFR-1);
Syn_filt(Ap, h1, h1, L_SUBFR, &h1[1], 0);
/*----------------------------------------------------------------------*
* Find the target vector for pitch search: *
*----------------------------------------------------------------------*/
Syn_filt(Ap, &(state->exc[i_subfr]), xn, L_SUBFR, state->mem_w0, 0);
/*---------------------------------------------------------------------*
* Closed-loop fractional pitch search *
*---------------------------------------------------------------------*/
T0 = Pitch_fr3_fast(&(state->exc[i_subfr]), xn, h1, L_SUBFR, T0_min, T0_max,
i_subfr, &T0_frac);
index = Enc_lag3(T0, T0_frac, &T0_min, &T0_max,PIT_MIN,PIT_MAX,i_subfr);
*ana++ = index;
if (i_subfr == 0) {
*ana++ = Parity_Pitch(index);
}
/*-----------------------------------------------------------------*
* - find filtered pitch exc *
* - compute pitch gain and limit between 0 and 1.2 *
* - update target vector for codebook search *
*-----------------------------------------------------------------*/
Syn_filt(Ap, &(state->exc[i_subfr]), y1, L_SUBFR, state->mem_zero, 0);
gain_pit = G_pitch(xn, y1, g_coeff, L_SUBFR);
/* clip pitch gain if taming is necessary */
taming = test_err(state, T0, T0_frac);
if( taming == 1){
if (gain_pit > GPCLIP) {
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 */
L_temp = ((Word32)y1[i] * gain_pit) << 2;
xn2[i] = sub(xn[i], extract_h(L_temp));
}
/*-----------------------------------------------------*
* - Innovative codebook search. *
*-----------------------------------------------------*/
index = ACELP_Code_A(xn2, h1, T0, state->sharp, code, y2, &i);
*ana++ = index; /* Positions index */
*ana++ = i; /* Signs index */
/*-----------------------------------------------------*
* - 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> */
*ana++ = Qua_gain(code, g_coeff_cs, exp_g_coeff_cs,
L_SUBFR, &gain_pit, &gain_code, taming);
/*------------------------------------------------------------*
* - Update pitch sharpening "sharp" with quantized gain_pit *
*------------------------------------------------------------*/
state->sharp = gain_pit;
if (state->sharp > SHARPMAX) { state->sharp = SHARPMAX; }
else if (state->sharp < SHARPMIN) { state->sharp = SHARPMIN; }
/*------------------------------------------------------*
* - Find the total excitation *
* - update filters memories for finding the target *
* vector in the next subframe *
*------------------------------------------------------*/
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);
L_temp = (Word32)(state->exc[i+i_subfr]) * (Word32)gain_pit +
(Word32)code[i] * (Word32)gain_code;
L_temp <<= 2;
state->exc[i+i_subfr] = g_round(L_temp);
}
update_exc_err(state, gain_pit, T0);
for (i = L_SUBFR-M, j = 0; i < L_SUBFR; i++, j++)
{
temp = ((Word32)y1[i] * (Word32)gain_pit) >> 14;
k = ((Word32)y2[i] * (Word32)gain_code) >> 13;
state->mem_w0[j] = sub(xn[i], add(temp, k));
}
Aq += MP1; /* interpolated LPC parameters for next subframe */
Ap += MP1;
}
/*--------------------------------------------------*
* Update signal for next frame. *
* -> shift to the left by L_FRAME: *
* speech[], wsp[] and exc[] *
*--------------------------------------------------*/
Copy(&(state->old_speech[L_FRAME]), &(state->old_speech[0]), L_TOTAL-L_FRAME);
Copy(&(state->old_wsp[L_FRAME]), &(state->old_wsp[0]), PIT_MAX);
Copy(&(state->old_exc[L_FRAME]), &(state->old_exc[0]), PIT_MAX+L_INTERPOL);
}
/*
********************************************************************************
* PUBLIC PROGRAM CODE
********************************************************************************
*/
int subframePreProc(
enum Mode mode, /* i : coder mode */
const Word16 gamma1[], /* i : spectral exp. factor 1 */
const Word16 gamma1_12k2[],/* i : spectral exp. factor 1 for EFR */
const Word16 gamma2[], /* i : spectral exp. factor 2 */
Word16 *A, /* i : A(z) unquantized for the 4 subframes */
Word16 *Aq, /* i : A(z) quantized for the 4 subframes */
Word16 *speech, /* i : speech segment */
Word16 *mem_err, /* i : pointer to error signal */
Word16 *mem_w0, /* i : memory of weighting filter */
Word16 *zero, /* i : pointer to zero vector */
Word16 ai_zero[], /* o : history of weighted synth. filter */
Word16 exc[], /* o : long term prediction residual */
Word16 h1[], /* o : impulse response */
Word16 xn[], /* o : target vector for pitch search */
Word16 res2[], /* o : long term prediction residual */
Word16 error[] /* o : error of LPC synthesis filter */
)
{
Word16 i;
Word16 Ap1[MP1]; /* A(z) with spectral expansion */
Word16 Ap2[MP1]; /* A(z) with spectral expansion */
const Word16 *g1; /* Pointer to correct gammma1 vector */
/*---------------------------------------------------------------*
* mode specific pointer to gamma1 values *
*---------------------------------------------------------------*/
test (); test ();
if ( sub(mode, MR122) == 0 || sub(mode, MR102) == 0 )
{
g1 = gamma1_12k2; move16 ();
}
else
{
g1 = gamma1; move16 ();
}
/*---------------------------------------------------------------*
* Find the weighted LPC coefficients for the weighting filter. *
*---------------------------------------------------------------*/
Weight_Ai(A, g1, Ap1);
Weight_Ai(A, gamma2, Ap2);
/*---------------------------------------------------------------*
* Compute impulse response, h1[], of weighted synthesis filter *
*---------------------------------------------------------------*/
for (i = 0; i <= M; i++)
{
ai_zero[i] = Ap1[i]; move16 ();
}
Syn_filt(Aq, ai_zero, h1, L_SUBFR, zero, 0);
Syn_filt(Ap2, h1, h1, L_SUBFR, zero, 0);
/*------------------------------------------------------------------------*
* *
* Find the target vector for pitch search: *
* *
*------------------------------------------------------------------------*/
/* LPC residual */
Residu(Aq, speech, res2, L_SUBFR);
Copy(res2, exc, L_SUBFR);
Syn_filt(Aq, exc, error, L_SUBFR, mem_err, 0);
Residu(Ap1, error, xn, L_SUBFR);
/* target signal xn[]*/
Syn_filt(Ap2, xn, xn, L_SUBFR, mem_w0, 0);
return 0;
}
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;
}