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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.
----------------------------------------------------------------------
*/
/*
File : COD_LD8A.C
Used for the floating point version of G.729A only
(not for G.729 main body)
*/
/*-----------------------------------------------------------------*
* Functions coder_ld8a and init_coder_ld8a *
* ~~~~~~~~~~ ~~~~~~~~~~~~~~~ *
* *
* init_coder_ld8a(void); *
* *
* ->Initialization of variables for the coder section. *
* *
* *
* coder_ld8a(Short ana[]); *
* *
* ->Main coder function. *
* *
* *
* Input: *
* *
* 80 speech data should have beee copy to vector new_speech[]. *
* This vector is global and is declared in this function. *
* *
* Ouputs: *
* *
* ana[] ->analysis parameters. *
* *
*-----------------------------------------------------------------*/
#include
#include "typedef.h"
#include "ld8a.h"
/*-----------------------------------------------------------*
* Coder constant parameters (defined in "ld8a.h") *
*-----------------------------------------------------------*
* L_WINDOW : LPC analysis window size. *
* L_NEXT : Samples of next frame needed for autocor. *
* L_FRAME : Frame size. *
* L_SUBFR : Sub-frame size. *
* M : LPC order. *
* MP1 : LPC order+1 *
* L_TOTAL : Total size of speech buffer. *
* PIT_MIN : Minimum pitch lag. *
* PIT_MAX : Maximum pitch lag. *
* L_INTERPOL : Length of filter for interpolation *
*-----------------------------------------------------------*/
/*--------------------------------------------------------*
* Static memory allocation. *
*--------------------------------------------------------*/
/* Speech vector */
static FLOAT old_speech[L_TOTAL];
static FLOAT *speech, *p_window;
FLOAT *new_speech; /* Global variable */
/* Weighted speech vector */
static FLOAT old_wsp[L_FRAME+PIT_MAX];
static FLOAT *wsp;
/* Excitation vector */
static FLOAT old_exc[L_FRAME+PIT_MAX+L_INTERPOL];
static FLOAT *exc;
/* LSP Line spectral frequencies */
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 lsp_old_q[M];
/* Filter's memory */
static FLOAT mem_w0[M], mem_w[M], mem_zero[M];
static FLOAT sharp;
/*-----------------------------------------------------------------*
* Function init_coder_ld8a *
* ~~~~~~~~~~~~~~~ *
* *
* init_coder_ld8a(void); *
* *
* ->Initialization of variables for the coder section. *
* - initialize pointers to speech buffer *
* - initialize static pointers *
* - set static vectors to zero *
*-----------------------------------------------------------------*/
void init_coder_ld8a(void)
{
/*----------------------------------------------------------------------*
* Initialize pointers to speech vector. *
* *
* *
* |--------------------|-------------|-------------|------------| *
* previous speech sf1 sf2 L_NEXT *
* *
* <---------------- Total speech vector (L_TOTAL) -----------> *
* <---------------- LPC analysis window (L_WINDOW) -----------> *
* | <-- present frame (L_FRAME) --> *
* old_speech | <-- new speech (L_FRAME) --> *
* p_window | | *
* speech | *
* new_speech *
*-----------------------------------------------------------------------*/
new_speech = old_speech + L_TOTAL - L_FRAME; /* New speech */
speech = new_speech - L_NEXT; /* Present frame */
p_window = old_speech + L_TOTAL - L_WINDOW; /* For LPC window */
/* Initialize static pointers */
wsp = old_wsp + PIT_MAX;
exc = old_exc + PIT_MAX + L_INTERPOL;
/* Static vectors to zero */
set_zero(old_speech, L_TOTAL);
set_zero(old_exc, PIT_MAX+L_INTERPOL);
set_zero(old_wsp, PIT_MAX);
set_zero(mem_w, M);
set_zero(mem_w0, M);
set_zero(mem_zero, M);
sharp = SHARPMIN;
copy(lsp_old, lsp_old_q, M);
lsp_encw_reset() ;
init_exc_err();
return;
}
/*-----------------------------------------------------------------*
* Function coder_ld8a *
* ~~~~~~~~~~ *
* coder_ld8a(int ana[]); *
* *
* ->Main coder function. *
* *
* *
* Input: *
* *
* 80 speech data should have beee copy to vector new_speech[]. *
* This vector is global and is declared in this function. *
* *
* Ouputs: *
* *
* ana[] ->analysis parameters. *
* *
*-----------------------------------------------------------------*/
void coder_ld8a(
int ana[] /* output: analysis parameters */
)
{
/* LPC coefficients */
FLOAT Aq_t[(MP1)*2]; /* A(z) quantized for the 2 subframes */
FLOAT Ap_t[(MP1)*2]; /* A(z) with spectral expansion */
FLOAT *Aq, *Ap; /* Pointer on Aq_t and Ap_t */
/* Other vectors */
FLOAT h1[L_SUBFR]; /* Impulse response h1[] */
FLOAT xn[L_SUBFR]; /* Target vector for pitch search */
FLOAT xn2[L_SUBFR]; /* Target vector for codebook search */
FLOAT code[L_SUBFR]; /* Fixed codebook excitation */
FLOAT y1[L_SUBFR]; /* Filtered adaptive excitation */
FLOAT y2[L_SUBFR]; /* Filtered fixed codebook excitation */
FLOAT g_coeff[5]; /* Correlations between xn, y1, & y2:
, , , ,*/
/* Scalars */
int i, j, i_subfr;
int T_op, T0, T0_min, T0_max, T0_frac;
int index;
FLOAT gain_pit, gain_code;
int taming;
/*------------------------------------------------------------------------*
* - 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 */
FLOAT r[MP1]; /* Autocorrelations */
FLOAT rc[M]; /* Reflexion coefficients */
FLOAT lsp_new[M]; /* lsp coefficients */
FLOAT lsp_new_q[M]; /* Quantized lsp coeff. */
/* LP analysis */
autocorr(p_window, M, r); /* Autocorrelations */
lag_window(M, r); /* Lag windowing */
levinson(r, Ap_t, rc); /* Levinson Durbin */
az_lsp(Ap_t, lsp_new, lsp_old); /* Convert A(z) to lsp */
/* 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 for the whole speech frame *
* - Set the range for searching closed-loop pitch in 1st subframe *
*----------------------------------------------------------------------*/
residu(&Aq_t[0], &speech[0], &exc[0], L_SUBFR);
residu(&Aq_t[MP1], &speech[L_SUBFR], &exc[L_SUBFR], L_SUBFR);
{
FLOAT Ap1[MP1];
Ap = Ap_t;
Ap1[0] = (F)1.0;
for(i=1; i<=M; i++)
Ap1[i] = Ap[i] - (F)0.7 * Ap[i-1];
syn_filt(Ap1, &exc[0], &wsp[0], L_SUBFR, mem_w, 1);
Ap += MP1;
for(i=1; i<=M; i++)
Ap1[i] = Ap[i] - (F)0.7 * Ap[i-1];
syn_filt(Ap1, &exc[L_SUBFR], &wsp[L_SUBFR], L_SUBFR, mem_w, 1);
}
/* Find open loop pitch lag for whole speech frame */
T_op = pitch_ol_fast(wsp, L_FRAME);
/* Range for closed loop pitch search in 1st subframe */
T0_min = T_op - 3;
if (T0_min < PIT_MIN) T0_min = PIT_MIN;
T0_max = T0_min + 6;
if (T0_max > PIT_MAX)
{
T0_max = PIT_MAX;
T0_min = 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 L_FRAME/L_SUBFR times. *
* - find the weighted LPC coefficients *
* - find the LPC residual signal *
* - 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] = (F)1.0;
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 *
* - find LTP residual. *
*-----------------------------------------------------------------*/
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 (gain_pit > GPCLIP) {
gain_pit = GPCLIP;
}
}
for (i = 0; i < L_SUBFR; i++)
xn2[i] = xn[i] - y1[i]*gain_pit;
/*-----------------------------------------------------*
* - Innovative codebook search. *
*-----------------------------------------------------*/
index = ACELP_code_A(xn2, h1, T0, sharp, code, y2, &i);
*ana++ = index; /* Positions index */
*ana++ = i; /* Signs index */
/*------------------------------------------------------*
* - Compute the correlations , , *
* - Vector quantize gains. *
*------------------------------------------------------*/
corr_xy2(xn, y1, y2, g_coeff);
*ana++ =qua_gain(code, g_coeff, L_SUBFR, &gain_pit, &gain_code,
taming);
/*------------------------------------------------------------*
* - Update pitch sharpening "sharp" with quantized gain_pit *
*------------------------------------------------------------*/
sharp = gain_pit;
if (sharp > SHARPMAX) sharp = SHARPMAX;
if (sharp < SHARPMIN) sharp = SHARPMIN;
/*------------------------------------------------------*
* - Find the total excitation *
* - update filters' memories for finding the target *
* vector in the next subframe (mem_w0[]) *
*------------------------------------------------------*/
for (i = 0; i < L_SUBFR; i++)
exc[i+i_subfr] = gain_pit*exc[i+i_subfr] + gain_code*code[i];
update_exc_err(gain_pit, T0);
for (i = L_SUBFR-M, j = 0; i < L_SUBFR; i++, j++)
mem_w0[j] = xn[i] - gain_pit*y1[i] - gain_code*y2[i];
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;
}