www.pudn.com > g729.rar > cod_ld8a.c
/*
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 Coder_ld8a and Init_Coder_ld8a *
* ~~~~~~~~~~ ~~~~~~~~~~~~~~~ *
* *
* Init_Coder_ld8a(void); *
* *
* ->Initialization of variables for the coder section. *
* *
* *
* Coder_ld8a(Word16 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
#include "typedef.h"
#include "basic_op.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 Word16 old_speech[L_TOTAL];
static Word16 *speech, *p_window;
Word16 *new_speech; /* Global variable */
/* Weighted speech vector */
static Word16 old_wsp[L_FRAME+PIT_MAX];
static Word16 *wsp;
/* 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};
static Word16 lsp_old_q[M];
/* Filter's memory */
static Word16 mem_w0[M], mem_w[M], mem_zero[M];
static Word16 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;
/* Initialize lsp_old_q[] */
Copy(lsp_old, lsp_old_q, M);
Lsp_encw_reset();
Init_exc_err();
return;
}
/*-----------------------------------------------------------------*
* Functions Coder_ld8a *
* ~~~~~~~~~~ *
* Coder_ld8a(Word16 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(
Word16 ana[] /* output : Analysis parameters */
)
{
/* LPC analysis */
Word16 Aq_t[(MP1)*2]; /* A(z) quantized for the 2 subframes */
Word16 Ap_t[(MP1)*2]; /* A(z/gamma) for the 2 subframes */
Word16 *Aq, *Ap; /* Pointer on Aq_t and Ap_t */
/* Other vectors */
Word16 h1[L_SUBFR]; /* Impulse response h1[] */
Word16 xn[L_SUBFR]; /* Target vector for pitch search */
Word16 xn2[L_SUBFR]; /* Target vector for codebook search */
Word16 code[L_SUBFR]; /* Fixed codebook excitation */
Word16 y1[L_SUBFR]; /* Filtered adaptive excitation */
Word16 y2[L_SUBFR]; /* Filtered fixed codebook excitation */
Word16 g_coeff[4]; /* Correlations between xn & y1 */
Word16 g_coeff_cs[5];
Word16 exp_g_coeff_cs[5]; /* Correlations between xn, y1, & y2
, -2,
, -2, 2 */
/* Scalars */
Word16 i, j, k, i_subfr;
Word16 T_op, T0, T0_min, T0_max, T0_frac;
Word16 gain_pit, gain_code, index;
Word16 temp, taming;
Word32 L_temp;
/*------------------------------------------------------------------------*
* - 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 */
Word16 r_l[MP1], r_h[MP1]; /* Autocorrelations low and hi */
Word16 rc[M]; /* Reflection coefficients. */
Word16 lsp_new[M], lsp_new_q[M]; /* LSPs at 2th subframe */
/* LP analysis */
Autocorr(p_window, M, r_h, r_l); /* Autocorrelations */
Lag_window(M, r_h, r_l); /* Lag windowing */
Levinson(r_h, r_l, Ap_t, rc); /* Levinson Durbin */
Az_lsp(Ap_t, lsp_new, lsp_old); /* From 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 *
*----------------------------------------------------------------------*/
Residu(&Aq_t[0], &speech[0], &exc[0], L_SUBFR);
Residu(&Aq_t[MP1], &speech[L_SUBFR], &exc[L_SUBFR], L_SUBFR);
{
Word16 Ap1[MP1];
Ap = Ap_t;
Ap1[0] = 4096;
for(i=1; i<=M; i++) /* Ap1[i] = Ap[i] - 0.7 * Ap[i-1]; */
Ap1[i] = mysub(Ap[i], mult(Ap[i-1], 22938));
Syn_filt(Ap1, &exc[0], &wsp[0], L_SUBFR, mem_w, 1);
Ap += MP1;
for(i=1; i<=M; i++) /* Ap1[i] = Ap[i] - 0.7 * Ap[i-1]; */
Ap1[i] = mysub(Ap[i], mult(Ap[i-1], 22938));
Syn_filt(Ap1, &exc[L_SUBFR], &wsp[L_SUBFR], L_SUBFR, mem_w, 1);
}
/* Find open loop pitch lag */
T_op = Pitch_ol_fast(wsp, PIT_MAX, L_FRAME);
/* Range for closed loop pitch search in 1st subframe */
T0_min = sub(T_op, 3);
if (sub(T0_min,PIT_MIN)<0) {
T0_min = PIT_MIN;
}
T0_max = add(T0_min, 6);
if (sub(T0_max ,PIT_MAX)>0)
{
T0_max = PIT_MAX;
T0_min = sub(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 2 times. *
* - find the weighted LPC coefficients *
* - find the LPC residual signal res[] *
* - 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] = 4096;
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 *
*-----------------------------------------------------------------*/
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 (sub(gain_pit, GPCLIP) > 0) {
gain_pit = GPCLIP;
}
}
/* xn2[i] = xn[i] - y1[i] * gain_pit */
for (i = 0; i < L_SUBFR; i++)
{
L_temp = L_mult(y1[i], gain_pit);
L_temp = L_shl(L_temp, 1); /* gain_pit in Q14 */
xn2[i] = mysub(xn[i], extract_h(L_temp));
}
/*-----------------------------------------------------*
* - Innovative codebook search. *
*-----------------------------------------------------*/
index = ACELP_Code_A(xn2, h1, T0, sharp, code, y2, &i);
*ana++ = index; /* Positions index */
*ana++ = i; /* Signs index */
/*-----------------------------------------------------*
* - Quantization of gains. *
*-----------------------------------------------------*/
g_coeff_cs[0] = g_coeff[0]; /* */
exp_g_coeff_cs[0] = negate(g_coeff[1]); /* Q-Format:XXX -> JPN */
g_coeff_cs[1] = negate(g_coeff[2]); /* (xn,y1) -> -2 */
exp_g_coeff_cs[1] = negate(add(g_coeff[3], 1)); /* Q-Format:XXX -> JPN */
Corr_xy2( xn, y1, y2, g_coeff_cs, exp_g_coeff_cs ); /* Q0 Q0 Q12 ^Qx ^Q0 */
/* g_coeff_cs[3]:exp_g_coeff_cs[3] = */
/* g_coeff_cs[4]:exp_g_coeff_cs[4] = -2 */
/* g_coeff_cs[5]:exp_g_coeff_cs[5] = 2 */
*ana++ = Qua_gain(code, g_coeff_cs, exp_g_coeff_cs,
L_SUBFR, &gain_pit, &gain_code, taming);
/*------------------------------------------------------------*
* - Update pitch sharpening "sharp" with quantized gain_pit *
*------------------------------------------------------------*/
sharp = gain_pit;
if (sub(sharp, SHARPMAX) > 0) { sharp = SHARPMAX; }
if (sub(sharp, SHARPMIN) < 0) { sharp = SHARPMIN; }
/*------------------------------------------------------*
* - Find the total excitation *
* - update filters memories for finding the target *
* vector in the next subframe *
*------------------------------------------------------*/
for (i = 0; i < L_SUBFR; i++)
{
/* exc[i] = gain_pit*exc[i] + gain_code*code[i]; */
/* exc[i] in Q0 gain_pit in Q14 */
/* code[i] in Q13 gain_cod in Q1 */
L_temp = L_mult(exc[i+i_subfr], gain_pit);
L_temp = L_mac(L_temp, code[i], gain_code);
L_temp = L_shl(L_temp, 1);
exc[i+i_subfr] = round(L_temp);
}
update_exc_err(gain_pit, T0);
for (i = L_SUBFR-M, j = 0; i < L_SUBFR; i++, j++)
{
temp = extract_h(L_shl( L_mult(y1[i], gain_pit), 1) );
k = extract_h(L_shl( L_mult(y2[i], gain_code), 2) );
mem_w0[j] = mysub(xn[i], add(temp, k));
}
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;
}