www.pudn.com > g729_audio_encode.rar > cod_ld8k.c
/*
ITU-T G.729 Speech Coder with Annex B ANSI-C Source Code
Version 1.3 Last modified: August 1997
Copyright (c) 1996,
AT&T, France Telecom, NTT, Universite de Sherbrooke, Lucent Technologies,
Rockwell International
All rights reserved.
*/
/*-----------------------------------------------------------------*
* Functions Coder_ld8k and Init_Coder_ld8k *
* ~~~~~~~~~~ ~~~~~~~~~~~~~~~ *
*-----------------------------------------------------------------*/
#include
#include
#include "typedef.h"
#include "basic_op.h"
#include "ld8k.h"
#include "tab_ld8k.h"
#include "oper_32b.h"
#include "vad.h"
#include "dtx.h"
#include "sid.h"
/*-----------------------------------------------------------*
* Coder constant parameters (defined in "ld8k.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;
/* Zero vector */
static Word16 ai_zero[L_SUBFR+MP1];
static Word16 *zero;
/* 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_syn[M], mem_w0[M], mem_w[M];
static Word16 mem_err[M+L_SUBFR], *error;
static Word16 sharp;
/* For G.729B */
/* DTX variables */
static Word16 pastVad;
static Word16 ppastVad;
static Word16 seed;
/*-----------------------------------------------------------------*
* Function Init_Coder_ld8k *
* ~~~~~~~~~~~~~~~ *
* *
* Init_Coder_ld8k(void); *
* *
* ->Initialization of variables for the coder section. *
* - initialize pointers to speech buffer *
* - initialize static pointers *
* - set static vectors to zero *
* *
*-----------------------------------------------------------------*/
void Init_Coder_ld8k(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;
zero = ai_zero + MP1;
error = mem_err + M;
/* 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_syn, M);
Set_zero(mem_w, M);
Set_zero(mem_w0, M);
Set_zero(mem_err, M);
Set_zero(zero, L_SUBFR);
sharp = SHARPMIN;
/* Initialize lsp_old_q[] */
Copy(lsp_old, lsp_old_q, M);
Lsp_encw_reset();
Init_exc_err();
/* For G.729B */
/* Initialize VAD/DTX parameters */
pastVad = 1;
ppastVad = 1;
seed = INIT_SEED;
vad_init();
Init_lsfq_noise();
return;
}
/*-----------------------------------------------------------------*
* Functions Coder_ld8k *
* ~~~~~~~~~~ *
* Coder_ld8k(Word16 ana[], Word16 synth[]); *
* *
* ->Main coder function. *
* *
* *
* Input: *
* *
* 80 speech data should have been copied to vector new_speech[].*
* This vector is global and is declared in this function. *
* *
* Ouputs: *
* *
* ana[] ->analysis parameters. *
* synth[] ->Local synthesis (for debug purpose) *
* *
*-----------------------------------------------------------------*/
void Coder_ld8k(
Word16 ana[], /* output : Analysis parameters */
Word16 synth[], /* output : Local synthesis */
Word16 frame, /* input : frame counter */
Word16 vad_enable /* input : VAD enable flag */
)
{
/* LPC analysis */
Word16 r_l[NP+1], r_h[NP+1]; /* Autocorrelations low and hi */
Word16 rc[M]; /* Reflection coefficients. */
Word16 A_t[(MP1)*2]; /* A(z) unquantized for the 2 subframes */
Word16 Aq_t[(MP1)*2]; /* A(z) quantized for the 2 subframes */
Word16 Ap1[MP1]; /* A(z) with spectral expansion */
Word16 Ap2[MP1]; /* A(z) with spectral expansion */
Word16 *A, *Aq; /* Pointer on A_t and Aq_t */
Word16 lsp_new[M], lsp_new_q[M]; /* LSPs at 2th subframe */
Word16 lsf_int[M]; /* Interpolated LSF 1st subframe. */
Word16 lsf_new[M];
Word16 gamma1[2], gamma2[2]; /* Weighting factor for the 2 subframes */
/* 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, i_gamma;
Word16 T_op, T0, T0_min, T0_max, T0_frac;
Word16 gain_pit, gain_code, index;
Word16 temp;
Word32 L_temp;
/* For G.729B */
Word16 rh_nbe[MP1];
Word16 lsfq_mem[MA_NP][M];
Word16 exp_R0, Vad;
/*------------------------------------------------------------------------*
* - 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) *
*------------------------------------------------------------------------*/
/* LP analysis */
Autocorr(p_window, NP, r_h, r_l, &exp_R0); /* Autocorrelations */
Copy(r_h, rh_nbe, MP1);
Lag_window(NP, r_h, r_l); /* Lag windowing */
Levinson(r_h, r_l, &A_t[MP1], rc, &temp); /* Levinson Durbin */
Az_lsp(&A_t[MP1], lsp_new, lsp_old); /* From A(z) to lsp */
/* For G.729B */
/* ------ VAD ------- */
Lsp_lsf(lsp_new, lsf_new, M);
vad(rc[1], lsf_new, r_h, r_l, exp_R0, p_window, frame,
pastVad, ppastVad, &Vad);
Update_cng(rh_nbe, exp_R0, Vad);
/*--------------------------------------------------------------------*
* Find interpolated LPC parameters in all subframes (unquantized) *
* The interpolated parameters are in array A_t[] of size (M+1)*4 *
*--------------------------------------------------------------------*/
Int_lpc(lsp_old, lsp_new, lsf_int, lsf_new, A_t);
for (i=0; i0)
{
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 *
* - update the impulse response h1[] by including fixed-gain pitch *
* - find target vector for codebook search *
* - codebook search *
* - encode codebook address *
* - VQ of pitch and codebook gains *
* - find synthesis speech *
* - update states of weighting filter *
*------------------------------------------------------------------------*/
A = A_t; /* pointer to interpolated LPC parameters */
Aq = Aq_t; /* pointer to interpolated quantized LPC parameters */
i_gamma = 0;
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
{
/*---------------------------------------------------------------*
* Find the weighted LPC coefficients for the weighting filter. *
*---------------------------------------------------------------*/
Weight_Az(A, gamma1[i_gamma], M, Ap1);
Weight_Az(A, gamma2[i_gamma], M, Ap2);
i_gamma = add(i_gamma,1);
/*---------------------------------------------------------------*
* Compute impulse response, h1[], of weighted synthesis filter *
*---------------------------------------------------------------*/
for (i = 0; i <= M; i++) {
ai_zero[i] = Ap1[i];
}
Syn_filt(Aq, ai_zero, h1, L_SUBFR, zero, 0);
Syn_filt(Ap2, h1, h1, L_SUBFR, zero, 0);
/*------------------------------------------------------------------------*
* *
* Find the target vector for pitch search: *
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ *
* *
* |------| res[n] *
* speech[n]---| A(z) |-------- *
* |------| | |--------| error[n] |------| *
* zero -- (-)--| 1/A(z) |-----------| W(z) |-- target *
* exc |--------| |------| *
* *
* Instead of subtracting the zero-input response of filters from *
* the weighted input speech, the above configuration is used to *
* compute the target vector. This configuration gives better performance *
* with fixed-point implementation. The memory of 1/A(z) is updated by *
* filtering (res[n]-exc[n]) through 1/A(z), or simply by subtracting *
* the synthesis speech from the input speech: *
* error[n] = speech[n] - syn[n]. *
* The memory of W(z) is updated by filtering error[n] through W(z), *
* or more simply by subtracting the filtered adaptive and fixed *
* codebook excitations from the target: *
* target[n] - gain_pit*y1[n] - gain_code*y2[n] *
* as these signals are already available. *
* *
*------------------------------------------------------------------------*/
Residu(Aq, &speech[i_subfr], &exc[i_subfr], L_SUBFR); /* LPC residual */
Syn_filt(Aq, &exc[i_subfr], error, L_SUBFR, mem_err, 0);
Residu(Ap1, error, xn, L_SUBFR);
Syn_filt(Ap2, xn, xn, L_SUBFR, mem_w0, 0); /* target signal xn[]*/
/*----------------------------------------------------------------------*
* Closed-loop fractional pitch search *
*----------------------------------------------------------------------*/
T0 = Pitch_fr3(&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 unity gain pitch excitation (adaptive codebook entry) *
* with fractional interpolation. *
* - find filtered pitch exc. y1[]=exc[] convolve with h1[]) *
* - compute pitch gain and limit between 0 and 1.2 *
* - update target vector for codebook search *
* - find LTP residual. *
*-----------------------------------------------------------------*/
Pred_lt_3(&exc[i_subfr], T0, T0_frac, L_SUBFR);
Convolve(&exc[i_subfr], h1, y1, L_SUBFR);
gain_pit = G_pitch(xn, y1, g_coeff, L_SUBFR);
/* clip pitch gain if taming is necessary */
temp = test_err(T0, T0_frac);
if( temp == 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] = sub(xn[i], extract_h(L_temp));
}
/*-----------------------------------------------------*
* - Innovative codebook search. *
*-----------------------------------------------------*/
index = ACELP_Codebook(xn2, h1, T0, sharp, i_subfr, 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, temp);
/*------------------------------------------------------------*
* - 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 *
* - find synthesis speech corresponding to exc[] *
* - update filters memories for finding the target *
* vector in the next subframe *
* (update error[-m..-1] and mem_w_err[]) *
* update error function for taming process *
*------------------------------------------------------*/
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);
Syn_filt(Aq, &exc[i_subfr], &synth[i_subfr], L_SUBFR, mem_syn, 1);
for (i = L_SUBFR-M, j = 0; i < L_SUBFR; i++, j++)
{
mem_err[j] = sub(speech[i_subfr+i], synth[i_subfr+i]);
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] = sub(xn[i], add(temp, k));
}
A += MP1; /* interpolated LPC parameters for next subframe */
Aq += 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;
}
/*---------------------------------------------------------------------------*
* routine corr_xy2() *
* ~~~~~~~~~~~~~~~~~~~~ *
* Find the correlations between the target xn[], the filtered adaptive *
* codebook excitation y1[], and the filtered 1st codebook innovation y2[]. *
* g_coeff[2]:exp_g_coeff[2] = *
* g_coeff[3]:exp_g_coeff[3] = -2 *
* g_coeff[4]:exp_g_coeff[4] = 2 *
*---------------------------------------------------------------------------*/
void Corr_xy2(
Word16 xn[], /* (i) Q0 :Target vector. */
Word16 y1[], /* (i) Q0 :Adaptive codebook. */
Word16 y2[], /* (i) Q12 :Filtered innovative vector. */
Word16 g_coeff[], /* (o) Q[exp]:Correlations between xn,y1,y2 */
Word16 exp_g_coeff[] /* (o) :Q-format of g_coeff[] */
)
{
Word16 i,exp;
Word16 exp_y2y2,exp_xny2,exp_y1y2;
Word16 y2y2, xny2, y1y2;
Word32 L_acc;
Word16 scaled_y2[L_SUBFR]; /* Q9 */
/*------------------------------------------------------------------*
* Scale down y2[] from Q12 to Q9 to avoid overflow *
*------------------------------------------------------------------*/
for(i=0; i */
L_acc = 1; /* Avoid case of all zeros */
for(i=0; i */
L_acc = 1; /* Avoid case of all zeros */
for(i=0; i */
/* Compute scalar product */
L_acc = 1; /* Avoid case of all zeros */
for(i=0; i */
return;
}