www.pudn.com > g729a.rar > DEC_LD8A.C, change:2005-05-31,size:9859b
/*
ITU-T G.729A Speech Coder ANSI-C Source Code
Version 1.1 Last modified: September 1996
Copyright (c) 1996,
AT&T, France Telecom, NTT, Universite de Sherbrooke
All rights reserved.
*/
/*-----------------------------------------------------------------*
* Functions Init_Decod_ld8a and Decod_ld8a *
*-----------------------------------------------------------------*/
#include "typedef.h"
#include "basic_op.h"
#include "ld8a.h"
/*---------------------------------------------------------------*
* Decoder constant parameters (defined in "ld8a.h") *
*---------------------------------------------------------------*
* L_FRAME : Frame size. *
* L_SUBFR : Sub-frame size. *
* M : LPC order. *
* MP1 : LPC order+1 *
* PIT_MIN : Minimum pitch lag. *
* PIT_MAX : Maximum pitch lag. *
* L_INTERPOL : Length of filter for interpolation *
* PRM_SIZE : Size of vector containing analysis parameters *
*---------------------------------------------------------------*/
/*--------------------------------------------------------*
* Static memory allocation. *
*--------------------------------------------------------*/
static Word16 synth_buf[L_FRAME+M]; /* Synthesis */
Word16 *synth;
/* Excitation vector */
static Word16 old_exc[L_FRAME+PIT_MAX+L_INTERPOL];
static Word16 *exc;
/* Lsp (Line spectral pairs) */
static Word16 lsp_old[M]={
30000, 26000, 21000, 15000, 8000, 0, -8000,-15000,-21000,-26000};
/* Filter's memory */
static Word16 mem_syn[M];
static Word16 sharp; /* pitch sharpening of previous frame */
static Word16 old_T0; /* integer delay of previous frame */
static Word16 gain_code; /* Code gain */
static Word16 gain_pitch; /* Pitch gain */
/*-----------------------------------------------------------------*
* Function Init_Decod_ld8a *
* ~~~~~~~~~~~~~~~ *
* *
* ->Initialization of variables for the decoder section. *
* *
*-----------------------------------------------------------------*/
void Init_Decod_ld8a(void)
{
Word16 i;
for (i=0; i<M; i++) synth_buf[i] = 0;
synth = synth_buf + M;
/* Initialize static pointer */
exc = old_exc + PIT_MAX + L_INTERPOL;
/* Static vectors to zero */
Set_zero(old_exc, PIT_MAX+L_INTERPOL);
Set_zero(mem_syn, M);
sharp = SHARPMIN;
old_T0 = 60;
gain_code = 0;
gain_pitch = 0;
Lsp_decw_reset();
return;
}
/*-----------------------------------------------------------------*
* Function Decod_ld8a *
* ~~~~~~~~~~ *
* ->Main decoder routine. *
* *
*-----------------------------------------------------------------*/
void Decod_ld8a(
Word16 parm[], /* (i) : vector of synthesis parameters
parm[0] = bad frame indicator (bfi) */
Word16 synth[], /* (o) : synthesis speech */
Word16 A_t[], /* (o) : decoded LP filter in 2 subframes */
Word16 *T2 /* (o) : decoded pitch lag in 2 subframes */
)
{
Word16 *Az; /* Pointer on A_t */
Word16 lsp_new[M]; /* LSPs */
Word16 code[L_SUBFR]; /* ACELP codevector */
/* Scalars */
Word16 i, j, i_subfr;
Word16 T0, T0_frac, index;
Word16 bfi;
Word32 L_temp;
Word16 bad_pitch; /* bad pitch indicator */
extern Word16 bad_lsf; /* bad LSF indicator */
/* Test bad frame indicator (bfi) */
bfi = *parm++;
/* Decode the LSPs */
D_lsp(parm, lsp_new, add(bfi, bad_lsf));
parm += 2;
/*
Note: "bad_lsf" is introduce in case the standard is used with
channel protection.
*/
/* Interpolation of LPC for the 2 subframes */
Int_qlpc(lsp_old, lsp_new, A_t);
/* update the LSFs for the next frame */
Copy(lsp_new, lsp_old, M);
/*------------------------------------------------------------------------*
* Loop for every subframe in the analysis frame *
*------------------------------------------------------------------------*
* The subframe size is L_SUBFR and the loop is repeated L_FRAME/L_SUBFR *
* times *
* - decode the pitch delay *
* - decode algebraic code *
* - decode pitch and codebook gains *
* - find the excitation and compute synthesis speech *
*------------------------------------------------------------------------*/
Az = A_t; /* pointer to interpolated LPC parameters */
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
{
index = *parm++; /* pitch index */
if(i_subfr == 0)
{
i = *parm++; /* get parity check result */
bad_pitch = add(bfi, i);
if( bad_pitch == 0)
{
Dec_lag3(index, PIT_MIN, PIT_MAX, i_subfr, &T0, &T0_frac);
old_T0 = T0;
}
else /* Bad frame, or parity error */
{
T0 = old_T0;
T0_frac = 0;
old_T0 = add( old_T0, 1);
if( sub(old_T0, PIT_MAX) > 0) {
old_T0 = PIT_MAX;
}
}
}
else /* second subframe */
{
if( bfi == 0)
{
Dec_lag3(index, PIT_MIN, PIT_MAX, i_subfr, &T0, &T0_frac);
old_T0 = T0;
}
else
{
T0 = old_T0;
T0_frac = 0;
old_T0 = add( old_T0, 1);
if( sub(old_T0, PIT_MAX) > 0) {
old_T0 = PIT_MAX;
}
}
}
*T2++ = T0;
/*-------------------------------------------------*
* - Find the adaptive codebook vector. *
*-------------------------------------------------*/
Pred_lt_3(&exc[i_subfr], T0, T0_frac, L_SUBFR);
/*-------------------------------------------------------*
* - Decode innovative codebook. *
* - Add the fixed-gain pitch contribution to code[]. *
*-------------------------------------------------------*/
if(bfi != 0) /* Bad frame */
{
parm[0] = Random() & (Word16)0x1fff; /* 13 bits random */
parm[1] = Random() & (Word16)0x000f; /* 4 bits random */
}
Decod_ACELP(parm[1], parm[0], code);
parm +=2;
j = shl(sharp, 1); /* From Q14 to Q15 */
if(sub(T0, L_SUBFR) <0 ) {
for (i = T0; i < L_SUBFR; i++) {
code[i] = add(code[i], mult(code[i-T0], j));
}
}
/*-------------------------------------------------*
* - Decode pitch and codebook gains. *
*-------------------------------------------------*/
index = *parm++; /* index of energy VQ */
Dec_gain(index, code, L_SUBFR, bfi, &gain_pitch, &gain_code);
/*-------------------------------------------------------------*
* - Update pitch sharpening "sharp" with quantized gain_pitch *
*-------------------------------------------------------------*/
sharp = gain_pitch;
if (sub(sharp, SHARPMAX) > 0) { sharp = SHARPMAX; }
if (sub(sharp, SHARPMIN) < 0) { sharp = SHARPMIN; }
/*-------------------------------------------------------*
* - Find the total excitation. *
* - Find synthesis speech corresponding to exc[]. *
*-------------------------------------------------------*/
for (i = 0; i < L_SUBFR; i++)
{
/* exc[i] = gain_pitch*exc[i] + gain_code*code[i]; */
/* exc[i] in Q0 gain_pitch in Q14 */
/* code[i] in Q13 gain_codeode in Q1 */
L_temp = L_mult(exc[i+i_subfr], gain_pitch);
L_temp = L_mac(L_temp, code[i], gain_code);
L_temp = L_shl(L_temp, 1);
exc[i+i_subfr] = round(L_temp);
}
/*
Overflow = 0;
Syn_filt(Az, &exc[i_subfr], &synth[i_subfr], L_SUBFR, mem_syn, 0);
if(Overflow != 0)
*/
if(Syn_filt(Az, &exc[i_subfr], &synth[i_subfr], L_SUBFR, mem_syn, 0) != 0)
{
/* In case of overflow in the synthesis */
/* -> Scale down vector exc[] and redo synthesis */
for(i=0; i<PIT_MAX+L_INTERPOL+L_FRAME; i++)
old_exc[i] = shr(old_exc[i], 2);
Syn_filt(Az, &exc[i_subfr], &synth[i_subfr], L_SUBFR, mem_syn, 1);
}
else
Copy(&synth[i_subfr+L_SUBFR-M], mem_syn, M);
Az += MP1; /* interpolated LPC parameters for next subframe */
}
/*--------------------------------------------------*
* Update signal for next frame. *
* -> shift to the left by L_FRAME exc[] *
*--------------------------------------------------*/
Copy(&old_exc[L_FRAME], &old_exc[0], PIT_MAX+L_INTERPOL);
return;
}