www.pudn.com > AVS_M_ver10.rar > dec_ace.c
/*
***********************************************************************
* COPYRIGHT AND WARRANTY INFORMATION
*
* Copyright 2007 Audio Video Coding Standard, Part Ⅹ
*
* 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 <float.h>
#include <stdlib.h>
#include <stdio.h>
#include <math.h>
#include "../include/amr_plus.h"
void decoder_acelp(
int prm[], /* input: parameters */
float A[], /* input: coefficients NxAz[M+1] */
int lg, /* input: frame length */
int codec_mode, /* input: AMR-WB+ mode (see cnst.h)*/
int bfi, /* input: 1=bad frame */
float exc[], /* i/o: exc[-(PIT_MAX+L_INTERPOL)..lg] */
float synth[], /* i/o: synth[-M..lg] */
int T_out[], /* out: LTP lag for high band */
float p_out[], /* out: LTP gain for high band */
int *pT, /* out: pitch for all subframe */
float *pgainT, /* out: pitch gain for all subfr */
int pit_adj, /* input: pitch adjust */
float stab_fac, /* input: stability of isf */
Decoder_State_Plus *st) /* i/o : coder memory state */
{
int i, i_subfr, select;
int T0, T0_frac, index, pit_flag, T0_min, T0_max;
float tmp, gain_pit, gain_code, voice_fac, ener, mean_ener_code, ener_wsyn;
float code[L_SUBFR], buf[M+L_OVLP];
float fac, exc2[L_SUBFR];
float *p_A, Ap1[1+M],Ap2[1+M],Ap3[1+TILT_ORDER],Ap[1+M];
short code3GPP[L_SUBFR], prm3GPP[10];
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 */
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);
}
/*------------------------------------------------------------------------*
* - decode mean_ener_code for gain decoder (d_gain2.c) *
*------------------------------------------------------------------------*/
index = *prm++;
/* decode mean energy with 2 bits : 18, 30, 42 or 54 dB */
if (!bfi) {
mean_ener_code = (((float)index) * 12.0f) + 18.0f;
}
/*------------------------------------------------------------------------*
* 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_ACELP/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 *
*------------------------------------------------------------------------*/
ener_wsyn = 0.01f;
p_A = A;
for (i_subfr = 0; i_subfr < lg; i_subfr += L_SUBFR)
{
pit_flag = i_subfr;
if (i_subfr == (2*L_SUBFR)) {
pit_flag = 0;
}
index = *prm++;
/*-------------------------------------------------*
* - Decode pitch lag *
*-------------------------------------------------*/
if (bfi) /* 如果发生帧错误 */
{
switch(st->bfi_count>4)/* 将连续坏帧数对4取模 */
{
case 1:/* 模1 */
T0 = st->old_T0 + 1;/* 将上帧的基音延迟加1作为当前帧的基音延迟 */
break;
case 2:/* 模2 */
T0 = st->old_T0 - 2;/* 将上帧的基音延迟减2作为当前帧的基音延迟 */
break;
case 3:/* 模3 */
T0 = st->old_T0 - 1;/* 将上帧的基音延迟减1作为当前帧的基音延迟 */
break;
case 0:/* 模0 */
T0 = st->old_T0 + 2;/* 将上帧的基音延迟加2作为当前帧的基音延迟 */
break;
}
if (T0 > PIT_MAX)/* 如果基音延迟大于上限 */
{
T0 = PIT_MAX;/* 将基音延迟设为上限 */
}
if (T0 < PIT_MIN)/* 如果基音延迟小于下限 */
{
T0 = PIT_MIN;/* 将基音延迟设为下限 */
}
T0_frac = 0;/* 基音延迟分数部分置0 */
}
else
{
if (pit_flag == 0)
{
if (index < (PIT_FR2-PIT_MIN)*4)
{
T0 = PIT_MIN + (index/4);
T0_frac = index - (T0 - PIT_MIN)*4;
}
else if (index < ( (PIT_FR2-PIT_MIN)*4 + (PIT_FR1-PIT_FR2)*2) )
{
index -= (PIT_FR2-PIT_MIN)*4;
T0 = PIT_FR2 + (index/2);
T0_frac = index - (T0 - PIT_FR2)*2;
T0_frac *= 2;
}
else {
T0 = index + PIT_FR1 - ((PIT_FR2-PIT_MIN)*4) - ((PIT_FR1-PIT_FR2)*2);
T0_frac = 0;
}
/* find T0_min and T0_max for subframe 2 or 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 */
{
T0 = T0_min + index/4;
T0_frac = index - (T0 - T0_min)*4;
}
}
/*-------------------------------------------------*
* - Find the pitch gain, the interpolation filter *
* and the adaptive codebook vector. *
*-------------------------------------------------*/
pred_lt4(&amt;exc[i_subfr], T0, T0_frac, L_SUBFR+1);
select = *prm++;
if (bfi) {
select = 1;
}
if (select == 0)
{
/* ADAPTIVE_EXCITATION_SEARCH */
/* find pitch excitation with lp filter */
for (i=0; i<L_SUBFR; i++) {
code[i] = (float)(0.26*exc[i-1+i_subfr] + 0.48*exc[i+i_subfr] + 0.26*exc[i+1+i_subfr]);
}
/* END_ADAPTIVE_EXCITATION_SEARCH */
mvr2r(code, &amt;exc[i_subfr], L_SUBFR);
}
/*-------------------------------------------------------*
* - Decode innovative codebook. *
* - Add the fixed-gain pitch contribution to code[]. *
*-------------------------------------------------------*/
{
int g;
for(g=0;g<8;g++) {
prm3GPP[g] = (short)prm[g];
}
}
if (codec_mode == MODE_9k6)
{
if (!bfi) {
D_ACELP_decode_4t(prm3GPP, 20, code3GPP);
}
prm+=4;
}
else if (codec_mode == MODE_11k2)
{
if (!bfi) {
D_ACELP_decode_4t(prm3GPP, 28, code3GPP);
}
prm+=4;
}
else if (codec_mode == MODE_12k8)
{
if (!bfi) {
D_ACELP_decode_4t(prm3GPP, 36, code3GPP);
}
prm+=4;
}
else if (codec_mode == MODE_14k4)
{
if (!bfi) {
D_ACELP_decode_4t(prm3GPP, 44, code3GPP);
}
prm+=4;
}
else if (codec_mode == MODE_16k)
{
if (!bfi) {
D_ACELP_decode_4t(prm3GPP, 52, code3GPP);
}
prm+=4;
}
else if (codec_mode == MODE_18k4)
{
if (!bfi) {
D_ACELP_decode_4t(prm3GPP, 64, code3GPP);
}
prm+=8;
}
else if (codec_mode == MODE_20k)
{
if (!bfi) {
D_ACELP_decode_4t(prm3GPP, 72, code3GPP);
}
prm+=8;
}
else if (codec_mode == MODE_23k2)
{
if (!bfi) {
D_ACELP_decode_4t(prm3GPP, 88, code3GPP);
}
prm+=8;
}
else
{
printf("invalid mode for acelp frame!\n");
exit(0);
}
{
int g;
for(g=0;g<L_SUBFR;g++) {
code[g] = (float) (code3GPP[g]/512);
}
}
if (bfi)
{
/* the innovative code doesn't need to be scaled (see D_gain2) */
for (i=0; i<L_SUBFR; i++)
{
code[i] = (float)E_UTIL_random(&amt;(st->seed_ace));
}
}
/*-------------------------------------------------------*
* - Add the fixed-gain pitch contribution to code[]. *
*-------------------------------------------------------*/
tmp = 0.0;
#ifndef PW_UPDATE
E_UTIL_f_preemph(code, TILT_CODE, L_SUBFR, &amt;tmp);
#endif
i = T0;
if (T0_frac > 2) {
i++;
}
E_GAIN_f_pitch_sharpening(code, i);
/*-------------------------------------------------*
* - Decode codebooks gains. *
*-------------------------------------------------*/
index = *prm++;
st->mem_gain_code[i_subfr/L_SUBFR] =
d_gain2_plus(index, code, L_SUBFR, &amt;gain_pit, &amt;gain_code, bfi,
mean_ener_code, &amt;(st->past_gpit), &amt;(st->past_gcode));
/*----------------------------------------------------------*
* Update parameters for the next subframe. *
* - tilt of code: 0.0 (unvoiced) to 0.5 (voiced) *
*----------------------------------------------------------*/
/* energy of pitch excitation */
ener = 0.0;
for (i=0; i<L_SUBFR; i++) {
ener += exc[i+i_subfr]*exc[i+i_subfr];
}
ener *= (gain_pit*gain_pit);
/* energy of innovative code excitation */
tmp = 0.0;
for (i=0; i<L_SUBFR; i++) {
tmp += code[i]*code[i];
}
tmp *= gain_code*gain_code;
/* find voice factor (1=voiced, -1=unvoiced) */
voice_fac = (float)((ener - tmp) / (ener + tmp + 0.01));
/*-------------------------------------------------------*
* - Find the total excitation. *
*-------------------------------------------------------*/
for (i = 0; i < L_SUBFR; i++) {
exc2[i] = gain_pit*exc[i+i_subfr];
}
for (i = 0; i < L_SUBFR; i++) {
exc[i+i_subfr] = gain_pit*exc[i+i_subfr] + gain_code*code[i];
}
/*-------------------------------------------------------*
* - Output pitch parameters for bass post-filter *
*-------------------------------------------------------*/
i = T0;
if (T0_frac > 2) {
i++;
}
if (i > PIT_MAX) {
i = PIT_MAX;
}
*pT++ = i;
*pgainT++ = gain_pit;
if (pit_adj == 0)
{
E_UTIL_synthesis(p_A, &amt;exc[i_subfr], &amt;synth[i_subfr], L_SUBFR, &amt;synth[i_subfr-M], 0);
}
else
{
/*------------------------------------------------------------*
* noise enhancer *
* ~~~~~~~~~~~~~~ *
* - Enhance excitation on noise. (modify gain of code) *
* If signal is noisy and LPC filter is stable, move gain *
* of code 1.5 dB toward gain of code threshold. *
* This decrease by 3 dB noise energy variation. *
*------------------------------------------------------------*/
fac = stab_fac; //ISF稳定性因子
/* 固定码本增益初始化修正 */
tmp = gain_code;
if (tmp < st->gc_threshold)
{
tmp = (float)(tmp*1.06);
if (tmp > st->gc_threshold)
{
tmp = st->gc_threshold;
}
}
else {
tmp = (float)(tmp/1.06);
if (tmp < st->gc_threshold)
{
tmp = st->gc_threshold;
}
}
st->gc_threshold = tmp;
/* 固定码本平滑增益的计算 */
if(fac>0.58)
gain_code = (float)((0.17*tmp) + (0.83*gain_code));
else
gain_code = (float)((0.83*tmp) + (0.17*gain_code));
/* 固定码本增益平滑，将固定码本乘以平滑增益 */
for (i=0; i<L_SUBFR; i++)
{
code[i] *= gain_code;
}
/*------------------------------------------------------------*
* pitch enhancer *
* ~~~~~~~~~~~~~~ *
* - Enhance excitation on voice. (HP filtering of code) *
* On voiced signal, filtering of code by a smooth fir HP *
* filter to decrease energy of code in low frequency. *
*------------------------------------------------------------*/
tmp = (float)(0.125*(1.0+voice_fac)); /* 0.25=voiced, 0=unvoiced */
exc2[0] += code[0] - (tmp*code[1]);
for (i=1; i<L_SUBFR-1; i++) {
exc2[i] += code[i] - (tmp*code[i-1]) - (tmp*code[i+1]);
}
exc2[L_SUBFR-1] += code[L_SUBFR-1] - (tmp*code[L_SUBFR-2]);
E_UTIL_synthesis(p_A, exc2, &amt;synth[i_subfr], L_SUBFR, &amt;synth[i_subfr-M], 0);
}
/* find weighted synthesis energy (for frame recovery on TCX mode) */
#ifdef PW_UPDATE
paraChoose(p_A,&amt;gamma1,&amt;gamma2,&amt;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, &amt;synth[i_subfr], code, L_SUBFR);
E_UTIL_formant(code,Ap2,L_SUBFR,st->mem_formant);
E_UTIL_tilt(code,Ap3,L_SUBFR,st->mem_tilt);
#endif
#ifndef PW_UPDATE
E_LPC_a_weight(p_A, Ap, GAMMA1, M);
E_UTIL_residu(Ap, &amt;synth[i_subfr], code, L_SUBFR);
E_UTIL_deemph(code, TILT_FAC, L_SUBFR, &amt;(st->mem_wsyn));
#endif
#ifdef NEW_TVC
#ifdef PW_UPDATE
if (i_subfr==192){
mvr2r(Ap1, &amt;st->wovlp[83], M+1);
mvr2r(Ap2, st->wovlp_ap2, M+1);
mvr2r(Ap3, st->wovlp_ap3, TILT_ORDER+1);
st->wovlp[80]= code[47];
st->wovlp[81]= code[31];
}
if (i_subfr==128){
st->wovlp[82]= code[63];
}
#else
if (i_subfr==192){
mvr2r(Ap, &amt;st->wovlp[83], M+1);
st->wovlp[80]= code[47];
st->wovlp[81]= code[31];
}
if (i_subfr==128){
st->wovlp[82]= code[63];
}
#endif
#endif
for (i=0; i<L_SUBFR; i++) {
ener_wsyn += code[i]*code[i];
}
p_A += (M+1);
} /* end of subframe loop */
/* RMS of weighted synthesis (for frame recovery on TCX mode) */
st->wsyn_rms = (float)sqrt(ener_wsyn/(float)lg);
/*----------------------------------------------------------*
* find 10ms ZIR in weighted domain for next tcx frame *
*----------------------------------------------------------*/
#ifndef NEW_TVC
mvr2r(&amt;synth[lg-M], buf, M);
set_zero(buf+M, L_OVLP);
E_UTIL_synthesis(p_A, buf+M, buf+M, L_OVLP, buf, 0);
#ifdef PW_UPDATE
{ float mem_temp1[M],mem_temp2[TILT_ORDER];
mvr2r(st->mem_formant,mem_temp1,M);
mvr2r(st->mem_tilt,mem_temp2,TILT_ORDER);
paraChoose(p_A,&amt;gamma1,&amt;gamma2,&amt;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, buf+M, st->wovlp, L_OVLP);
E_UTIL_formant(st->wovlp,Ap2,L_OVLP,mem_temp1);
E_UTIL_tilt(st->wovlp,Ap3,L_OVLP,mem_temp2);
}
#else
E_LPC_a_weight(p_A, Ap, GAMMA1, M); /* wAi of tcx is quantized */
E_UTIL_residu(Ap, buf+M, st->wovlp, L_OVLP);
tmp = st->mem_wsyn;
E_UTIL_deemph(st->wovlp, TILT_FAC, L_OVLP, &amt;tmp);
#endif
for (i=1; i<(L_OVLP/2); i++) {
st->wovlp[L_OVLP-i] *= ((float)i)/((float)(L_OVLP/2));
}
#endif
#ifdef NEW_TVC
mvr2r(&amt;synth[176], st->wovlp, 80);
set_zero(&amt;st->wovlp[100], 28);
st->gw_flg=1;
#endif
st->ovlp_size = 0; /* indicate ACELP frame to TCX */
/* update pitch value for bfi procedure */
st->old_T0_frac = T0_frac;
st->old_T0 = T0;
return;
}