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/* ITU-T G.729 Software Package Release 2 (November 2006) */
/*
ITU-T G.729 Annex C - Reference C code for floating point
implementation of G.729 Annex A
Version 1.01 of 15.September.98
*/
/*
----------------------------------------------------------------------
COPYRIGHT NOTICE
----------------------------------------------------------------------
ITU-T G.729 Annex C ANSI C source code
Copyright (C) 1998, AT&T, France Telecom, NTT, University of
Sherbrooke. All rights reserved.
----------------------------------------------------------------------
*/
/*-----------------------------------------------------------------*
* Functions init_decod_ld8a and decod_ld8a *
*-----------------------------------------------------------------*/
#include "typedef.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. *
*--------------------------------------------------------*/
/* Excitation vector */
static FLOAT old_exc[L_FRAME+PIT_MAX+L_INTERPOL];
static FLOAT *exc;
/* Lsp (Line spectral pairs) */
static FLOAT lsp_old[M]={
(F)0.9595, (F)0.8413, (F)0.6549, (F)0.4154, (F)0.1423,
(F)-0.1423, (F)-0.4154, (F)-0.6549, (F)-0.8413, (F)-0.9595};
static FLOAT mem_syn[M]; /* Synthesis filter's memory */
static FLOAT sharp; /* pitch sharpening of previous frame */
static int old_T0; /* integer delay of previous frame */
static FLOAT gain_code; /* Code gain */
static FLOAT gain_pitch; /* Pitch gain */
/*-----------------------------------------------------------------*
* Function init_decod_ld8a *
* ~~~~~~~~~~~~~~~ *
* *
* ->Initialization of variables for the decoder section. *
* *
*-----------------------------------------------------------------*/
void init_decod_ld8a(void)
{
/* 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 = (F)0.0;
gain_pitch = (F)0.0;
lsp_decw_reset() ;
return;
}
/*-----------------------------------------------------------------*
* Function decod_ld8a *
* ~~~~~~~~~~ *
* ->Main decoder routine. *
* *
*-----------------------------------------------------------------*/
void decod_ld8a(
int parm[], /* (i) : vector of synthesis parameters
parm[0] = bad frame indicator (bfi) */
FLOAT synth[], /* (o) : synthesis speech */
FLOAT A_t[], /* (o) : decoded LP filter in 2 subframes */
int *T2 /* (o) : decoded pitch lag in 2 subframes */
)
{
FLOAT *Az; /* Pointer on A_t */
FLOAT lsp_new[M]; /* Decoded LSP's */
FLOAT code[L_SUBFR]; /* ACELP codevector */
/* Scalars */
int i, i_subfr;
int T0, T0_frac, index;
int bfi, bad_pitch;
extern int bad_lsf; /* bad LSF indicator */
/* Test bad frame indicator (bfi) */
bfi = *parm++;
/* Decode the LSPs */
d_lsp(parm, lsp_new, bfi+bad_lsf );
parm += 2; /* Advance synthesis parameters pointer */
/*
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) {
/*-------------------------------------------------*
* - Find the adaptive codebook vector. *
*--------------------------------------------------*/
index = *parm++; /* pitch index */
if(i_subfr == 0)
{
i = *parm++; /* get parity check result */
bad_pitch = 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++;
if( (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++;
if( (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_g729() & (INT16)0x1fff; /* 13 bits random */
parm[1] = random_g729() & (INT16)0x000f; /* 4 bits random */
}
decod_ACELP(parm[1], parm[0], code);
parm +=2;
for (i = T0; i < L_SUBFR; i++) code[i] += sharp * code[i-T0];
/*-------------------------------------------------*
* - 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 (sharp > SHARPMAX) sharp = SHARPMAX;
if (sharp < SHARPMIN) sharp = SHARPMIN;
/*-------------------------------------------------------*
* - Find the total excitation. *
* - Find synthesis speech corresponding to exc[]. *
*-------------------------------------------------------*/
for (i = 0; i < L_SUBFR; i++)
exc[i+i_subfr] = gain_pitch*exc[i+i_subfr] + gain_code*code[i];
syn_filt(Az, &exc[i_subfr], &synth[i_subfr], L_SUBFR, mem_syn, 1);
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;
}