/* 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
* 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 : ACELP_CO.C
* Used for the floating point version of G.729 main body
* (not for G.729A)
*/
/**
*
* @param x (i) :Target vector
* @param h (i) :Impulse response of filters
* @param t0 (i) :Pitch lag
* @param pitch_sharp (i) :Last quantized pitch gain
* @param i_subfr (i) :Indicator of 1st subframe,
* @param code (o) :Innovative codebook
* @param y (o) :Filtered innovative codebook
* @param sign (o) :Signs of 4 pulses
* @return index of pulses positions
*/
public int ACELP_codebook(
float[] x,
float[] h,
int t0,
float pitch_sharp,
int i_subfr,
float[] code,
float[] y,
IntReference sign
)
{
var DIM_RR = Ld8k.DIM_RR;
var L_SUBFR = Ld8k.L_SUBFR;
int i, index;
var dn = new float[L_SUBFR];
var rr = new float[DIM_RR];
/*----------------------------------------------------------------*
* Include fixed-gain pitch contribution into impulse resp. h[] *
* Find correlations of h[] needed for the codebook search. *
*-----------------------------------------------------------------*/
if (t0 < L_SUBFR)
{
for (i = t0; i < L_SUBFR; i++)
{
h[i] += pitch_sharp * h[i - t0];
}
}
cor_h(h, rr);
/*----------------------------------------------------------------*
* Compute correlation of target vector with impulse response. *
*-----------------------------------------------------------------*/
CorFunc.cor_h_x(h, x, dn); /* backward filtered target vector dn */
/*----------------------------------------------------------------*
* Find innovative codebook. *
*-----------------------------------------------------------------*/
index = d4i40_17(dn, rr, h, code, y, sign, i_subfr);
/*------------------------------------------------------*
* - Add the fixed-gain pitch contribution to code[]. *
*-------------------------------------------------------*/
if (t0 < L_SUBFR)
{
for (i = t0; i < L_SUBFR; i++)
{
code[i] += pitch_sharp * code[i - t0];
}
}
return(index);
}
/**
* Encoder routine ( speech data should be in new_speech ).
*
* @param ana output: analysis parameters
*/
public void coder_ld8k(
int[] ana
)
{
/* LPC coefficients */
var r = new float[MP1]; /* Autocorrelations low and hi */
var A_t = new float[MP1 * 2]; /* A(z) unquantized for the 2 subframes */
var Aq_t = new float[MP1 * 2]; /* A(z) quantized for the 2 subframes */
var Ap1 = new float[MP1]; /* A(z) with spectral expansion */
var Ap2 = new float[MP1]; /* A(z) with spectral expansion */
float[] A, Aq; /* Pointer on A_t and Aq_t */
int A_offset, Aq_offset;
/* LSP coefficients */
float[] lsp_new = new float[M], lsp_new_q = new float[M]; /* LSPs at 2th subframe */
var lsf_int = new float[M]; /* Interpolated LSF 1st subframe. */
var lsf_new = new float[M];
/* Variable added for adaptive gamma1 and gamma2 of the PWF */
var rc = new float[M]; /* Reflection coefficients */
var gamma1 = new float[2]; /* Gamma1 for 1st and 2nd subframes */
var gamma2 = new float[2]; /* Gamma2 for 1st and 2nd subframes */
/* Other vectors */
var synth = new float[L_FRAME]; /* Buffer for synthesis speech */
var h1 = new float[L_SUBFR]; /* Impulse response h1[] */
var xn = new float[L_SUBFR]; /* Target vector for pitch search */
var xn2 = new float[L_SUBFR]; /* Target vector for codebook search */
var code = new float[L_SUBFR]; /* Fixed codebook excitation */
var y1 = new float[L_SUBFR]; /* Filtered adaptive excitation */
var y2 = new float[L_SUBFR]; /* Filtered fixed codebook excitation */
var g_coeff = new float[5]; /* Correlations between xn, y1, & y2:
* <y1,y1>, <xn,y1>, <y2,y2>, <xn,y2>,<y1,y2>*/
/* Scalars */
int i, j, i_gamma, i_subfr;
var iRef = new IntReference();
int T_op, t0;
IntReference t0_min = new IntReference(), t0_max = new IntReference(), t0_frac = new IntReference();
int index, taming;
float gain_pit, gain_code = 0.0f;
FloatReference _gain_pit = new FloatReference(), _gain_code = new FloatReference();
var ana_offset = 0;
/*------------------------------------------------------------------------*
* - 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 */
Lpc.autocorr(p_window, p_window_offset, M, r); /* Autocorrelations */
Lpc.lag_window(M, r); /* Lag windowing */
Lpc.levinson(r, A_t, MP1, rc); /* Levinson Durbin */
Lpc.az_lsp(A_t, MP1, lsp_new, lsp_old); /* From A(z) to lsp */
/* LSP quantization */
quaLsp.qua_lsp(lsp_new, lsp_new_q, ana);
ana_offset += 2; /* Advance analysis parameters pointer */
/*--------------------------------------------------------------------*
* Find interpolated LPC parameters in all subframes (both quantized *
* and unquantized). *
* The interpolated parameters are in array A_t[] of size (M+1)*4 *
* and the quantized interpolated parameters are in array Aq_t[] *
*--------------------------------------------------------------------*/
Lpcfunc.int_lpc(lsp_old, lsp_new, lsf_int, lsf_new, A_t);
Lpcfunc.int_qlpc(lsp_old_q, lsp_new_q, Aq_t);
/* update the LSPs for the next frame */
for (i = 0; i < M; i++)
{
lsp_old[i] = lsp_new[i];
lsp_old_q[i] = lsp_new_q[i];
}
/*----------------------------------------------------------------------*
* - Find the weighting factors *
*----------------------------------------------------------------------*/
pwf.perc_var(gamma1, gamma2, lsf_int, lsf_new, rc);
/*----------------------------------------------------------------------*
* - 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 *
*----------------------------------------------------------------------*/
Lpcfunc.weight_az(A_t, 0, gamma1[0], M, Ap1);
Lpcfunc.weight_az(A_t, 0, gamma2[0], M, Ap2);
Filter.residu(Ap1, 0, speech, speech_offset, wsp, wsp_offset, L_SUBFR);
Filter.syn_filt(Ap2, 0, wsp, wsp_offset, wsp, wsp_offset, L_SUBFR, mem_w, 0, 1);
Lpcfunc.weight_az(A_t, MP1, gamma1[1], M, Ap1);
Lpcfunc.weight_az(A_t, MP1, gamma2[1], M, Ap2);
Filter.residu(Ap1, 0, speech, speech_offset + L_SUBFR, wsp, wsp_offset + L_SUBFR, L_SUBFR);
Filter.syn_filt(Ap2, 0, wsp, wsp_offset + L_SUBFR, wsp, wsp_offset + L_SUBFR, L_SUBFR, mem_w, 0, 1);
/* Find open loop pitch lag for whole speech frame */
T_op = Pitch.pitch_ol(wsp, wsp_offset, PIT_MIN, PIT_MAX, L_FRAME);
/* range for closed loop pitch search in 1st subframe */
t0_min.value = T_op - 3;
if (t0_min.value < PIT_MIN)
{
t0_min.value = PIT_MIN;
}
t0_max.value = t0_min.value + 6;
if (t0_max.value > PIT_MAX)
{
t0_max.value = PIT_MAX;
t0_min.value = t0_max.value - 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 *
* - 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 */
A_offset = 0;
Aq = Aq_t; /* pointer to interpolated quantized LPC parameters */
Aq_offset = 0;
i_gamma = 0;
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
{
/*---------------------------------------------------------------*
* Find the weighted LPC coefficients for the weighting filter. *
*---------------------------------------------------------------*/
Lpcfunc.weight_az(A, A_offset, gamma1[i_gamma], M, Ap1);
Lpcfunc.weight_az(A, A_offset, gamma2[i_gamma], M, Ap2);
i_gamma++;
/*---------------------------------------------------------------*
* Compute impulse response, h1[], of weighted synthesis filter *
*---------------------------------------------------------------*/
for (i = 0; i <= M; i++)
{
ai_zero[i] = Ap1[i];
}
Filter.syn_filt(Aq, Aq_offset, ai_zero, 0, h1, 0, L_SUBFR, zero, zero_offset, 0);
Filter.syn_filt(Ap2, 0, h1, 0, h1, 0, L_SUBFR, zero, zero_offset, 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. *
* *
*------------------------------------------------------------------------*/
Filter.residu(
Aq,
Aq_offset,
speech,
speech_offset + i_subfr,
exc,
exc_offset + i_subfr,
L_SUBFR); /* LPC residual */
Filter.syn_filt(Aq, Aq_offset, exc, exc_offset + i_subfr, error, error_offset, L_SUBFR, mem_err, 0, 0);
Filter.residu(Ap1, 0, error, error_offset, xn, 0, L_SUBFR);
Filter.syn_filt(Ap2, 0, xn, 0, xn, 0, L_SUBFR, mem_w0, 0, 0); /* target signal xn[]*/
/*----------------------------------------------------------------------*
* Closed-loop fractional pitch search *
*----------------------------------------------------------------------*/
t0 = Pitch.pitch_fr3(
exc,
exc_offset + i_subfr,
xn,
h1,
L_SUBFR,
t0_min.value,
t0_max.value,
i_subfr,
t0_frac);
index = Pitch.enc_lag3(t0, t0_frac.value, t0_min, t0_max, PIT_MIN, PIT_MAX, i_subfr);
ana[ana_offset] = index;
ana_offset++;
if (i_subfr == 0)
{
ana[ana_offset] = PParity.parity_pitch(index);
ana_offset++;
}
/*-----------------------------------------------------------------*
* - 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. *
*-----------------------------------------------------------------*/
PredLt3.pred_lt_3(exc, exc_offset + i_subfr, t0, t0_frac.value, L_SUBFR);
Filter.convolve(exc, exc_offset + i_subfr, h1, y1, L_SUBFR);
gain_pit = Pitch.g_pitch(xn, y1, g_coeff, L_SUBFR);
/* clip pitch gain if taming is necessary */
taming = this.taming.test_err(t0, t0_frac.value);
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. *
*-----------------------------------------------------*/
iRef.value = i;
index = acelpCo.ACELP_codebook(xn2, h1, t0, sharp, i_subfr, code, y2, iRef);
i = iRef.value;
ana[ana_offset] = index; /* Positions index */
ana_offset++;
ana[ana_offset] = i; /* Signs index */
ana_offset++;
/*-----------------------------------------------------*
* - Quantization of gains. *
*-----------------------------------------------------*/
CorFunc.corr_xy2(xn, y1, y2, g_coeff);
_gain_pit.value = gain_pit;
_gain_code.value = gain_code;
ana[ana_offset] = quaGain.qua_gain(code, g_coeff, L_SUBFR, _gain_pit, _gain_code, taming);
gain_pit = _gain_pit.value;
gain_code = _gain_code.value;
ana_offset++;
/*------------------------------------------------------------*
* - 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 *
* - find synthesis speech corresponding to exc[] *
* - update filters' memories for finding the target *
* vector in the next subframe *
* (update error[-m..-1] and mem_w0[]) *
* update error function for taming process *
*------------------------------------------------------*/
for (i = 0; i < L_SUBFR; i++)
{
exc[exc_offset + i + i_subfr] = gain_pit * exc[exc_offset + i + i_subfr] + gain_code * code[i];
}
this.taming.update_exc_err(gain_pit, t0);
Filter.syn_filt(Aq, Aq_offset, exc, exc_offset + i_subfr, synth, i_subfr, L_SUBFR, mem_syn, 0, 1);
for (i = L_SUBFR - M, j = 0; i < L_SUBFR; i++, j++)
{
mem_err[j] = speech[speech_offset + i_subfr + i] - synth[i_subfr + i];
mem_w0[j] = xn[i] - gain_pit * y1[i] - gain_code * y2[i];
}
A_offset += MP1; /* interpolated LPC parameters for next subframe */
Aq_offset += MP1;
}
/*--------------------------------------------------*
* Update signal for next frame. *
* -> shift to the left by L_FRAME: *
* speech[], wsp[] and exc[] *
*--------------------------------------------------*/
Util.copy(old_speech, L_FRAME, old_speech, L_TOTAL - L_FRAME);
Util.copy(old_wsp, L_FRAME, old_wsp, PIT_MAX);
Util.copy(old_exc, L_FRAME, old_exc, PIT_MAX + L_INTERPOL);
}
