www.pudn.com > AVS_M_ver10.rar > cod_ace.c
/*
***********************************************************************
* COPYRIGHT AND WARRANTY INFORMATION
*
* Copyright 2007 Audio Video Coding Standard, Part ¢ú
*
* This software module was developed by AVS Audio sub-group
*
* DISCLAIMER OF WARRANTY
*
* These software programs are available to the users without any
* license fee or royalty on an "as is" basis. The AVS disclaims
* any and all warranties, whether express, implied, or statutory,
* including any implied warranties of merchantability or of fitness
* for a particular purpose. In no event shall the contributors or
* the AVS be liable for any incidental, punitive, or consequential
* damages of any kind whatsoever arising from the use of this program.
*
* This disclaimer of warranty extends to the user of this program
* and user's customers, employees, agents, transferees, successors,
* and assigns.
*
* The AVS does not represent or warrant that the program furnished
* hereunder are free of infringement of any third-party patents.
* Commercial implementations of AVS, including shareware, may be
* subject to royalty fees to patent holders. Information regarding
* the AVS patent policy is available from the AVS Web site at
* http://www.avs.org.cn
*
* THIS IS NOT A GRANT OF PATENT RIGHTS - SEE THE AVS PATENT POLICY.
************************************************************************
*/
#include
#include
#include
#include
#include "../include/amr_plus.h"
#ifdef PW_UPDATE
void coder_acelp(
float A[], /* input: coefficients 4xAz[M+1] */
float Aq[], /* input: coefficients 4xAz_q[M+1] */
float speech[], /* input: speech[-M..lg] */
float *mem_wsp, /* in/out: wsp memory */
float *mem_wsyn, /* in/out: wsyn memory */
float synth[], /* in/out: synth[-M..lg] */
float exc[], /* in/out: exc[-(PIT_MAX+L_INTERPOL)..lg+1] */
float wovlp[], /* out: wovlp[0..128] */
int lg, /* input: frame length */
int codec_mode, /* input: AMR_WB+ mode (see cnst.h) */
float norm_corr,
float norm_corr2,
int T_op, /* input: open-loop LTP */
int T_op2, /* input: open-loop LTP */
int T_out[], /* output: integer pitch-lag */
float p_out[], /* output: pitch gain */
float c_out[], /* output: fixed codebook gain */
int pit_adj,
int *prm, /* output: acelp parameters */
float *mem_formant,
float *mem_tilt,
float *mem_formant_zir,
float *mem_tilt_zir)
{
int i, i_subfr, select;
int T0, T0_min, T0_max, index, pit_flag;
long int T0_frac;
float tmp, ener, max, mean_ener_code;
float gain_pit, gain_code, voice_fac;
float g_corr[5]; /*norm_corr, norm_corr2*/
float *p_A, *p_Aq, Ap1[M+1],Ap2[M+1],Ap3[TILT_ORDER+1];
float h1[L_SUBFR];
float code[L_SUBFR];
short code3GPP[L_SUBFR];
float error[M+L_SUBFR];
float cn[L_SUBFR];
float xn[L_SUBFR];
float xn2[L_SUBFR];
float dn[L_SUBFR]; /* Correlation between xn and h1 */
float y1[L_SUBFR]; /* Filtered adaptive excitation */
float y2[L_SUBFR]; /* Filtered adaptive excitation */
int PIT_MIN; /* Minimum pitch lag with resolution 1/4 */
int PIT_FR2; /* Minimum pitch lag with resolution 1/2 */
int PIT_FR1; /* Minimum pitch lag with resolution 1 */
int PIT_MAX; /* Maximum pitch lag */
float exc_origin[L_SUBFR]; /* original excitation */
float exc_high[L_SUBFR]; /* high frequent of adaptive excitation */
float exc_low[L_SUBFR]; /* low frequent of adaptive excitation */
/* temporary var to calculate correlation */
double corr_temp;
double ener_temp1;
double ener_temp2;
float mem_temp1[M],mem_temp2[TILT_ORDER];
if(pit_adj ==0) {
PIT_MIN = PIT_MIN_12k8;
PIT_FR2 = PIT_FR2_12k8;
PIT_FR1 = PIT_FR1_12k8;
PIT_MAX = PIT_MAX_12k8;
}
else {
i = (((pit_adj*PIT_MIN_12k8)+(FSCALE_DENOM/2))/FSCALE_DENOM)-PIT_MIN_12k8;
PIT_MIN = PIT_MIN_12k8 + i;
PIT_FR2 = PIT_FR2_12k8 - i;
PIT_FR1 = PIT_FR1_12k8;
PIT_MAX = PIT_MAX_12k8 + (6*i);
}
T_op *= OPL_DECIM;
T_op2 *= OPL_DECIM;
/* range for closed loop pitch search in 1st subframe */
T0_min = T_op - 8;
if (T0_min < PIT_MIN) {
T0_min = PIT_MIN;
}
T0_max = T0_min + 15;
if (T0_max > PIT_MAX) {
T0_max = PIT_MAX;
T0_min = T0_max - 15;
}
/*------------------------------------------------------------------------*
* Find and quantize mean_ener_code for gain quantizer (q_gain2_live.c) *
* This absolute reference replace the gains prediction. *
* This modification for AMR_WB+ have the following advantage: *
* - better quantization on attacks (onset, drum impulse, etc.) *
* - robust on packet lost. *
* - independent of previous mode (can be TCX with alg. 7 bits quantizer).*
*------------------------------------------------------------------------*/
max=0.0;
mean_ener_code = 0.0;
p_Aq = Aq;
for (i_subfr=0; i_subfr max) {
max = ener;
}
mean_ener_code += 0.25f*ener;
p_Aq += (M+1);
}
/* reduce mean energy on voiced signal */
mean_ener_code -= 5.0f*norm_corr;
mean_ener_code -= 5.0f*norm_corr2;
/* quantize mean energy with 2 bits : 18, 30, 42 or 54 dB */
tmp = (mean_ener_code-18.0f) / 12.0f;
index = (int)floor(tmp + 0.5);
if (index < 0) {
index = 0;
}
if (index > 3) {
index = 3;
}
mean_ener_code = (((float)index) * 12.0f) + 18.0f;
/* limit mean energy to be able to quantize attack (table limited to +24dB) */
while ((mean_ener_code < (max-27.0)) && (index < 3)) {
index++;
mean_ener_code += 12.0;
}
*prm = index; prm++;
/*------------------------------------------------------------------------*
* 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 L_FRAME_PLUS/L_SUBFR times. *
* - compute impulse response of weighted synthesis filter (h1[]) *
* - compute the target signal for pitch search *
* - find the closed-loop pitch parameters *
* - encode the pitch dealy *
* - update the impulse response h1[] by including fixed-gain pitch *
* - find target vector for codebook search *
* - correlation between target vector and impulse response *
* - codebook search *
* - encode codebook address *
* - VQ of pitch and codebook gains *
* - find synthesis speech *
* - update states of weighting filter *
*------------------------------------------------------------------------*/
p_A = A;
p_Aq = Aq;
for (i_subfr=0; i_subfr PIT_MAX) {
T0_max = PIT_MAX;
T0_min = T0_max - 15;
}
}
/*-----------------------------------------------------------------------*
Find the target vector for pitch search:
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
|------| res[n]
speech[n]---| A(z) |--------
|------| | |-------------|
zero -- (-)--| 1/A(z/gamma)|----- target xn[]
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.
*-----------------------------------------------------------------------*/
/* ...... TVC inter-frame smoothing module START ...... */
#ifdef NEW_TVC
if (i_subfr==192){
wovlp[65]=*mem_wsp;
paraChoose(p_Aq,&gamma1,&gamma2,&gamma3);
E_LPC_a_weight(p_Aq, Ap1, gamma1, M);
E_LPC_a_weight(p_Aq, Ap2, gamma2, M);
E_LPC_a_weight(p_Aq, Ap3, gamma3, TILT_ORDER);
E_UTIL_residu(Ap1, &speech[i_subfr], xn, L_SUBFR);
E_UTIL_formant(xn,Ap2,L_SUBFR,mem_formant);
E_UTIL_tilt(xn,Ap3,L_SUBFR,mem_tilt);
mvr2r(xn, wovlp, L_SUBFR);
*mem_wsp=wovlp[65];
}
#endif
/* ...... TVC inter-frame smoothing module END ...... */
//Perceptual Weighting
/* find WSP (normaly done for pitch ol) */
paraChoose(p_Aq,&gamma1,&gamma2,&gamma3);
E_LPC_a_weight(p_A, Ap1, gamma1, M);
E_LPC_a_weight(p_A, Ap2, gamma2, M);
E_LPC_a_weight(p_A, Ap3, gamma3, TILT_ORDER);
E_UTIL_residu(Ap1, &speech[i_subfr], xn, L_SUBFR);
E_UTIL_formant(xn,Ap2,L_SUBFR,mem_formant);
E_UTIL_tilt(xn,Ap3,L_SUBFR,mem_tilt);
/* find ZIR in weighted domain */
mvr2r(&synth[i_subfr-M], error, M);
set_zero(error+M, L_SUBFR);
E_UTIL_synthesis(p_Aq, error+M, error+M, L_SUBFR, error, 0);
E_UTIL_residu(Ap1, error+M, xn2, L_SUBFR);
E_UTIL_formant(xn2,Ap2,L_SUBFR,mem_formant_zir);
E_UTIL_tilt(xn2,Ap3,L_SUBFR,mem_tilt_zir);
/* target xn = wsp - ZIR in weighted domain */
for (i=0; i cn[] */
set_zero(code, M);
mvr2r(xn, code+M, L_SUBFR/2);
//Perceptual Weighting
set_zero(mem_temp1,M);
set_zero(mem_temp2,TILT_ORDER);
E_UTIL_detilt(code+M,Ap3,L_SUBFR/2,mem_temp2);
E_UTIL_deformant(code+M,Ap2,L_SUBFR/2,mem_temp1);
E_UTIL_synthesis(Ap1, code+M, code+M, L_SUBFR/2, code, 0);
E_UTIL_residu(p_Aq, code+M, cn, L_SUBFR/2);
/* second half: res[] --> cn[] (approximated and faster) */
mvr2r(&exc[i_subfr+(L_SUBFR/2)], cn+(L_SUBFR/2), L_SUBFR/2);
/*---------------------------------------------------------------*
* Compute impulse response, h1[], of weighted synthesis filter *
*---------------------------------------------------------------*/
set_zero(h1, L_SUBFR);
mvr2r(Ap1, h1, M+1);
E_UTIL_synthesis(p_Aq, h1, h1, L_SUBFR, &h1[M+1], 0);
//Perceptual Weighting
set_zero(mem_temp1,M);
set_zero(mem_temp2,TILT_ORDER);
E_UTIL_formant(h1,Ap2,L_SUBFR,mem_temp1);
E_UTIL_tilt(h1,Ap3,L_SUBFR,mem_temp2);
/*----------------------------------------------------------------------*
* Closed-loop fractional pitch search *
*----------------------------------------------------------------------*/
/* find closed loop fractional pitch lag */
T0 = E_GAIN_closed_loop_search(&exc[i_subfr], xn, h1, T0_min, T0_max, &T0_frac,
pit_flag, PIT_FR2, PIT_FR1);
/* encode pitch lag */
if (pit_flag == 0) { /* if 1st/3rd subframe */
/*--------------------------------------------------------------*
* The pitch range for the 1st/3rd subframe is encoded with *
* 9 bits and is divided as follows: *
* PIT_MIN to PIT_FR2-1 resolution 1/4 (frac = 0,1,2 or 3) *
* PIT_FR2 to PIT_FR1-1 resolution 1/2 (frac = 0 or 2) *
* PIT_FR1 to PIT_MAX resolution 1 (frac = 0) *
*--------------------------------------------------------------*/
if (T0 < PIT_FR2) {
index = T0*4 + T0_frac - (PIT_MIN*4);
} else if (T0 < PIT_FR1) {
index = T0*2 + (T0_frac>>1) - (PIT_FR2*2) + ((PIT_FR2-PIT_MIN)*4);
} else {
index = T0 - PIT_FR1 + ((PIT_FR2-PIT_MIN)*4) + ((PIT_FR1-PIT_FR2)*2);
}
/* find T0_min and T0_max for subframe 2 and 4 */
T0_min = T0 - 8;
if (T0_min < PIT_MIN) {
T0_min = PIT_MIN;
}
T0_max = T0_min + 15;
if (T0_max > PIT_MAX) {
T0_max = PIT_MAX;
T0_min = T0_max - 15;
}
} else { /* if subframe 2 or 4 */
/*--------------------------------------------------------------*
* The pitch range for subframe 2 or 4 is encoded with 6 bits: *
* T0_min to T0_max resolution 1/4 (frac = 0,1,2 or 3) *
*--------------------------------------------------------------*/
i = T0 - T0_min;
index = i*4 + T0_frac;
}
/* store pitch-lag for high band coding */
T_out[i_subfr/L_SUBFR] = T0;
*prm = index; prm++;
/*----------------------------------------------------------------*
* Adapted excitation search *
* - find unity gain pitch excitation (adaptive codebook entry) *
* with fractional interpolation. *
* - decide if lowpass filter is used by compare correlation *
* - find filtered pitch exc. y1[]=exc[] convolved with h1[]) *
*-----------------------------------------------------------------*/
/* find pitch excitation */
mvr2r(&exc[i_subfr], exc_origin, L_SUBFR);
pred_lt4(&exc[i_subfr], T0, T0_frac, L_SUBFR+1);
/* compare correlation to choose lowpass or not */
corr_temp = ener_temp1 = ener_temp2= 0.0;
for(i=0; i 0.19){
/* use exc as adaptive excitation */
select = 1;
}
else{
/* use exc_low as adaptive excitation */
select = 0;
mvr2r(exc_low,exc+i_subfr,64);
}
/* get synthetical signal with adaptive excitation */
/* END_ADAPTIVE_EXCITATION_SEARCH */
E_UTIL_f_convolve(exc+i_subfr, h1, y1);
/* get pitch gain */
gain_pit = E_ACELP_xy1_corr(xn,y1,g_corr);
*prm = select; prm++;
/*-----------------------------------------------------------------*
* - update target vector for codebook search *
*-----------------------------------------------------------------*/
E_ACELP_codebook_target_update(xn, xn2, y1, gain_pit);
E_ACELP_codebook_target_update(cn, cn, &exc[i_subfr], gain_pit);
/*-----------------------------------------------------------------*
* - include fixed-gain pitch contribution into impulse resp. h1[] *
*-----------------------------------------------------------------*/
tmp = 0.0;
//E_UTIL_f_preemph(h1, TILT_CODE, L_SUBFR, &tmp);
//tmp = 0.0;
//E_UTIL_f_preemph(cn, TILT_CODE, L_SUBFR, &tmp);
if (T0_frac > 2) {
T0++;
}
E_GAIN_f_pitch_sharpening(h1, T0);
/*-----------------------------------------------------------------*
* - Correlation between target xn2[] and impulse response h1[] *
* - Innovative codebook search *
*-----------------------------------------------------------------*/
E_ACELP_xh_corr(xn2, dn, h1);
if (codec_mode == MODE_9k6) {
E_ACELP_4t_newall(dn, cn, h1, code3GPP, y2, 20, 0, (Word32*)prm);
prm += 4;
} else if (codec_mode == MODE_11k2) {
E_ACELP_4t_newall(dn, cn, h1, code3GPP, y2, 28, 0, (Word32*)prm);
prm += 4;
} else if (codec_mode == MODE_12k8) {
E_ACELP_4t_newall(dn, cn, h1, code3GPP, y2, 36, 0, (Word32*)prm);
prm += 4;
} else if (codec_mode == MODE_14k4) {
E_ACELP_4t_newall(dn, cn, h1, code3GPP, y2, 44, 0, (Word32*)prm);
prm += 4;
} else if (codec_mode == MODE_16k) {
E_ACELP_4t_newall(dn, cn, h1, code3GPP, y2, 52, 0, (Word32*)prm);
prm += 4;
} else if (codec_mode == MODE_18k4) {
E_ACELP_4t_newall(dn, cn, h1, code3GPP, y2, 64, 0, (Word32*)prm);
prm += 8;
} else if (codec_mode == MODE_20k) {
E_ACELP_4t_newall(dn, cn, h1, code3GPP, y2, 72, 0, (Word32*)prm);
prm += 8;
} else if (codec_mode == MODE_23k2) {
E_ACELP_4t_newall(dn, cn, h1, code3GPP, y2, 88, 0, (Word32*)prm);
prm += 8;
} else {
printf("invalid mode for acelp frame!\n");
exit(0);
}
{
int g;
for(g=0;g PIT_MAX) {
T0_max = PIT_MAX;
T0_min = T0_max - 15;
}
/*------------------------------------------------------------------------*
* Find and quantize mean_ener_code for gain quantizer (q_gain2_live.c) *
* This absolute reference replace the gains prediction. *
* This modification for AMR_WB+ have the following advantage: *
* - better quantization on attacks (onset, drum impulse, etc.) *
* - robust on packet lost. *
* - independent of previous mode (can be TCX with alg. 7 bits quantizer).*
*------------------------------------------------------------------------*/
max=0.0;
mean_ener_code = 0.0;
p_Aq = Aq;
for (i_subfr=0; i_subfr max) {
max = ener;
}
mean_ener_code += 0.25f*ener;
p_Aq += (M+1);
}
/* reduce mean energy on voiced signal */
mean_ener_code -= 5.0f*norm_corr;
mean_ener_code -= 5.0f*norm_corr2;
/* quantize mean energy with 2 bits : 18, 30, 42 or 54 dB */
tmp = (mean_ener_code-18.0f) / 12.0f;
index = (int)floor(tmp + 0.5);
if (index < 0) {
index = 0;
}
if (index > 3) {
index = 3;
}
mean_ener_code = (((float)index) * 12.0f) + 18.0f;
/* limit mean energy to be able to quantize attack (table limited to +24dB) */
while ((mean_ener_code < (max-27.0)) && (index < 3)) {
index++;
mean_ener_code += 12.0;
}
*prm = index; prm++;
/*------------------------------------------------------------------------*
* 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 L_FRAME_PLUS/L_SUBFR times. *
* - compute impulse response of weighted synthesis filter (h1[]) *
* - compute the target signal for pitch search *
* - find the closed-loop pitch parameters *
* - encode the pitch dealy *
* - update the impulse response h1[] by including fixed-gain pitch *
* - find target vector for codebook search *
* - correlation between target vector and impulse response *
* - codebook search *
* - encode codebook address *
* - VQ of pitch and codebook gains *
* - find synthesis speech *
* - update states of weighting filter *
*------------------------------------------------------------------------*/
p_A = A;
p_Aq = Aq;
for (i_subfr=0; i_subfr PIT_MAX) {
T0_max = PIT_MAX;
T0_min = T0_max - 15;
}
}
/*-----------------------------------------------------------------------*
Find the target vector for pitch search:
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
|------| res[n]
speech[n]---| A(z) |--------
|------| | |-------------|
zero -- (-)--| 1/A(z/gamma)|----- target xn[]
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.
*-----------------------------------------------------------------------*/
/* ...... TVC inter-frame smoothing module START ...... */
#ifdef NEW_TVC
if (i_subfr==192){
wovlp[65]=*mem_wsp;
E_LPC_a_weight(p_Aq, Ap, GAMMA1, M);
E_UTIL_residu(Ap, &speech[i_subfr], xn, L_SUBFR);
E_UTIL_deemph(xn, TILT_FAC, L_SUBFR, mem_wsp);
mvr2r(xn, wovlp, L_SUBFR);
*mem_wsp=wovlp[65];
}
#endif
/* ...... TVC inter-frame smoothing module END ...... */
/* find WSP (normaly done for pitch ol) */
E_LPC_a_weight(p_A, Ap, GAMMA1, M);
E_UTIL_residu(Ap, &speech[i_subfr], xn, L_SUBFR);
E_UTIL_deemph(xn, TILT_FAC, L_SUBFR, mem_wsp);
/* find ZIR in weighted domain */
mvr2r(&synth[i_subfr-M], error, M);
set_zero(error+M, L_SUBFR);
E_UTIL_synthesis(p_Aq, error+M, error+M, L_SUBFR, error, 0);
E_LPC_a_weight(p_A, Ap, GAMMA1, M);
E_UTIL_residu(Ap, error+M, xn2, L_SUBFR);
E_UTIL_deemph(xn2, TILT_FAC, L_SUBFR, mem_wsyn);
/* target xn = wsp - ZIR in weighted domain */
for (i=0; i cn[] */
set_zero(code, M);
mvr2r(xn, code+M, L_SUBFR/2);
tmp = 0.0;
E_UTIL_f_preemph(code+M, TILT_FAC, L_SUBFR/2, &tmp);
E_LPC_a_weight(p_A, Ap, GAMMA1, M);
E_UTIL_synthesis(Ap, code+M, code+M, L_SUBFR/2, code, 0);
E_UTIL_residu(p_Aq, code+M, cn, L_SUBFR/2);
/* second half: res[] --> cn[] (approximated and faster) */
mvr2r(&exc[i_subfr+(L_SUBFR/2)], cn+(L_SUBFR/2), L_SUBFR/2);
/*---------------------------------------------------------------*
* Compute impulse response, h1[], of weighted synthesis filter *
*---------------------------------------------------------------*/
E_LPC_a_weight(p_A, Ap, GAMMA1, M);
set_zero(h1, L_SUBFR);
mvr2r(Ap, h1, M+1);
E_UTIL_synthesis(p_Aq, h1, h1, L_SUBFR, &h1[M+1], 0);
tmp = 0.0;
E_UTIL_deemph(h1, TILT_FAC, L_SUBFR, &tmp);
/*----------------------------------------------------------------------*
* Closed-loop fractional pitch search *
*----------------------------------------------------------------------*/
/* find closed loop fractional pitch lag */
T0 = E_GAIN_closed_loop_search(&exc[i_subfr], xn, h1, T0_min, T0_max, &T0_frac,
pit_flag, PIT_FR2, PIT_FR1);
/* encode pitch lag */
if (pit_flag == 0) { /* if 1st/3rd subframe */
/*--------------------------------------------------------------*
* The pitch range for the 1st/3rd subframe is encoded with *
* 9 bits and is divided as follows: *
* PIT_MIN to PIT_FR2-1 resolution 1/4 (frac = 0,1,2 or 3) *
* PIT_FR2 to PIT_FR1-1 resolution 1/2 (frac = 0 or 2) *
* PIT_FR1 to PIT_MAX resolution 1 (frac = 0) *
*--------------------------------------------------------------*/
if (T0 < PIT_FR2) {
index = T0*4 + T0_frac - (PIT_MIN*4);
} else if (T0 < PIT_FR1) {
index = T0*2 + (T0_frac>>1) - (PIT_FR2*2) + ((PIT_FR2-PIT_MIN)*4);
} else {
index = T0 - PIT_FR1 + ((PIT_FR2-PIT_MIN)*4) + ((PIT_FR1-PIT_FR2)*2);
}
/* find T0_min and T0_max for subframe 2 and 4 */
T0_min = T0 - 8;
if (T0_min < PIT_MIN) {
T0_min = PIT_MIN;
}
T0_max = T0_min + 15;
if (T0_max > PIT_MAX) {
T0_max = PIT_MAX;
T0_min = T0_max - 15;
}
} else { /* if subframe 2 or 4 */
/*--------------------------------------------------------------*
* The pitch range for subframe 2 or 4 is encoded with 6 bits: *
* T0_min to T0_max resolution 1/4 (frac = 0,1,2 or 3) *
*--------------------------------------------------------------*/
i = T0 - T0_min;
index = i*4 + T0_frac;
}
/* store pitch-lag for high band coding */
T_out[i_subfr/L_SUBFR] = T0;
*prm = index; prm++;
/*-----------------------------------------------------------------*
* - find unity gain pitch excitation (adaptive codebook entry) *
* with fractional interpolation. *
* - find filtered pitch exc. y1[]=exc[] convolved with h1[]) *
* - compute pitch gain1 *
*-----------------------------------------------------------------*/
/* find pitch exitation */
mvr2r(&exc[i_subfr], exc_origin, L_SUBFR);
pred_lt4(&exc[i_subfr], T0, T0_frac, L_SUBFR+1);
/* compare correlation to choose lowpass or not */
corr_temp = ener_temp1 = ener_temp2= 0.0;
for(i=0; i 0.19){
/* use exc as adaptive excitation */
select = 1;
}
else{
/* use exc_low as adaptive excitation */
select = 0;
mvr2r(exc_low,exc+i_subfr,64);
}
/* get synthetical signal with adaptive excitation */
/* END_ADAPTIVE_EXCITATION_SEARCH */
E_UTIL_f_convolve(exc+i_subfr, h1, y1);
/* get pitch gain */
gain_pit = E_ACELP_xy1_corr(xn,y1,g_corr);
*prm = select; prm++;
/*-----------------------------------------------------------------*
* - update target vector for codebook search *
*-----------------------------------------------------------------*/
E_ACELP_codebook_target_update(xn, xn2, y1, gain_pit);
E_ACELP_codebook_target_update(cn, cn, &exc[i_subfr], gain_pit);
/*-----------------------------------------------------------------*
* - include fixed-gain pitch contribution into impulse resp. h1[] *
*-----------------------------------------------------------------*/
tmp = 0.0;
E_UTIL_f_preemph(h1, TILT_CODE, L_SUBFR, &tmp);
tmp = 0.0;
E_UTIL_f_preemph(cn, TILT_CODE, L_SUBFR, &tmp);
if (T0_frac > 2) {
T0++;
}
E_GAIN_f_pitch_sharpening(h1, T0);
/*-----------------------------------------------------------------*
* - Correlation between target xn2[] and impulse response h1[] *
* - Innovative codebook search *
*-----------------------------------------------------------------*/
E_ACELP_xh_corr(xn2, dn, h1);
if (codec_mode == MODE_9k6) {
E_ACELP_4t_newall(dn, cn, h1, code3GPP, y2, 20, 0, (Word32*)prm);
prm += 4;
} else if (codec_mode == MODE_11k2) {
E_ACELP_4t_newall(dn, cn, h1, code3GPP, y2, 28, 0, (Word32*)prm);
prm += 4;
} else if (codec_mode == MODE_12k8) {
E_ACELP_4t_newall(dn, cn, h1, code3GPP, y2, 36, 0, (Word32*)prm);
prm += 4;
} else if (codec_mode == MODE_14k4) {
E_ACELP_4t_newall(dn, cn, h1, code3GPP, y2, 44, 0, (Word32*)prm);
prm += 4;
} else if (codec_mode == MODE_16k) {
E_ACELP_4t_newall(dn, cn, h1, code3GPP, y2, 52, 0, (Word32*)prm);
prm += 4;
} else if (codec_mode == MODE_18k4) {
E_ACELP_4t_newall(dn, cn, h1, code3GPP, y2, 64, 0, (Word32*)prm);
prm += 8;
} else if (codec_mode == MODE_20k) {
E_ACELP_4t_newall(dn, cn, h1, code3GPP, y2, 72, 0, (Word32*)prm);
prm += 8;
} else if (codec_mode == MODE_23k2) {
E_ACELP_4t_newall(dn, cn, h1, code3GPP, y2, 88, 0, (Word32*)prm);
prm += 8;
} else {
printf("invalid mode for acelp frame!\n");
exit(0);
}
{
int g;
for(g=0;g