www.pudn.com > AVS_M_ver10.rar > cod_hf.c
/*
***********************************************************************
* COPYRIGHT AND WARRANTY INFORMATION
*
* Copyright 2007 Audio Video Coding Standard, Part ¢ú
*
* This software module was developed by AVS Audio sub-group
*
* DISCLAIMER OF WARRANTY
*
* These software programs are available to the users without any
* license fee or royalty on an "as is" basis. The AVS disclaims
* any and all warranties, whether express, implied, or statutory,
* including any implied warranties of merchantability or of fitness
* for a particular purpose. In no event shall the contributors or
* the AVS be liable for any incidental, punitive, or consequential
* damages of any kind whatsoever arising from the use of this program.
*
* This disclaimer of warranty extends to the user of this program
* and user's customers, employees, agents, transferees, successors,
* and assigns.
*
* The AVS does not represent or warrant that the program furnished
* hereunder are free of infringement of any third-party patents.
* Commercial implementations of AVS, including shareware, may be
* subject to royalty fees to patent holders. Information regarding
* the AVS patent policy is available from the AVS Web site at
* http://www.avs.org.cn
*
* THIS IS NOT A GRANT OF PATENT RIGHTS - SEE THE AVS PATENT POLICY.
************************************************************************
*/
#include
#include
#include
#include
#include "../include/amr_plus.h"
/*----------- CODE AND SYNTHESIZE UPPER-BAND SIGNAL (synth_hf[]) ----------
bitstream pointer = ptr
coding modes (coder type=ACELP,TCX and frame length=20,40,80 ms) for upper-band are copied from lower-band
HF coding mode depends on bit allocation to HFs (nbits_hf) :
nbits_hf = 4*16=64
the lower-band excitation is randomized, folded, and shaped first in time-domain
with subframe gains then in frequency domain with a LP filter (plus energy smoothing)
nbits_hf = 4*16+640=704
the lower-band is quantized by ACELP/TCX model, using the same mode as lower band.
*/
/*------------------------------------------------------*
* HF analysis, quantization & synthesis *
*------------------------------------------------------*/
void init_coder_hf(Coder_StState *st)
{
int i;
/* Static vectors to zero */
set_zero(st->mem_hf1, MHF);
set_zero(st->mem_hf2, MHF);
set_zero(st->mem_hf3, MHF);
set_zero(st->past_q_isf_hf, MHF);
st->old_gain = 1.0;
/* isp initialization */
for (i=0; iispold_hf[i] = (float)cos(3.141592654*(float)(i+1)/(float)MHF);
st->ispold_hf[MHF-1] = 0.045f;
mvr2r(st->ispold_hf, st->ispold_q_hf, MHF);
return;
}
void coder_hf(
int mod[], /* (i) : mode for each 20ms frame (mode[4] */
float AqLF[], /* (i) : Q coeff in lower band (AdLF[16]) */
float speech[], /* (i) : speech vector [-M..L_FRAME_PLUS] */
float speech_hf[], /* (i) : HF speech vec [-MHF..L_FRAME_PLUS+L_NEXT] */
float synth_hf[], /* (o) : HF synthesis [0..L_FRAME_PLUS] */
float window[], /* (i) : window for LPC analysis */
int param[], /* (o) : parameters (NB_DIV*NPRM_HF_DIV) */
int fscale,
Coder_StState *st) /* i/o : coder memory state */
{
/* LPC coefficients */
float r[MHF+1]; /* Autocorrelations of windowed speech */
float A[(NB_SUBFR+1)*(MHF+1)];
float Aq[(NB_SUBFR+1)*(MHF+1)];
float *p_A, *p_Aq, Ap[MHF+1];
float ispnew[NB_DIV*MHF]; /* LSPs at 4nd subframe */
float ispnew_q[MHF]; /* LSPs at 4nd subframe */
float isfnew[MHF];
float gain_hf[NB_SUBFR], gain4[4];
float HF[L_SUBFR];
float buf[MHF+L_SUBFR];
/* Scalars */
int i, j, k, i_subfr, sf, nsf, index, mode, *prm;
float tmp, gain, ener;
/*---------------------------------------------------------------*
* Perform HF LP analysis four times (every 20 ms) *
* - autocorrelation + lag windowing *
* - Levinson-Durbin algorithm to find a[] *
* - convert a[] to isp[] *
* - interpol isp[] *
*---------------------------------------------------------------*/
for (k=0; kispold_hf, MHF);
int_lpc_np1(st->ispold_hf, &ispnew[k*MHF], &A[k*4*(MHF+1)], 4, MHF);
mvr2r(&ispnew[k*MHF], st->ispold_hf, MHF);
}
p_A = A;
for (k=0; kpast_q_isf_hf, prm, st->mean_isf_hf, st->dico1_isf_hf);
prm += 2;
/* Convert isfs to the cosine domain */
isf2isp(isfnew, ispnew_q, MHF);
/* interpol quantized isp for local synthesis */
int_lpc_np1(st->ispold_q_hf, ispnew_q, Aq, nsf, MHF);
mvr2r(ispnew_q, st->ispold_q_hf, MHF);
/* gain factor to match amplitude at 6.4kHz */
gain = match_gain_6k4(&AqLF[((k*4)+nsf)*(M+1)], &Aq[nsf*(MHF+1)]);
int_gain(st->old_gain, gain, gain_hf, nsf);
st->old_gain = gain;
/*------------------------------------------------------*
* find gain in weighted domain *
*------------------------------------------------------*/
p_Aq = Aq;
for (sf=0; sfmem_hf1, buf, MHF);
E_UTIL_synthesisPlus(p_Aq, MHF, buf+MHF, buf+MHF, L_SUBFR, st->mem_hf1, 1);
E_LPC_a_weight(p_A, Ap, GAMMA_HF, MHF);
E_UTIL_residuPlus(p_A, MHF, buf+MHF, HF, L_SUBFR);
E_UTIL_synthesisPlus(Ap, MHF, HF, HF, L_SUBFR, st->mem_hf2, 1);
E_UTIL_residuPlus(p_A, MHF, &speech_hf[i_subfr], buf, L_SUBFR);
E_UTIL_synthesisPlus(Ap, MHF, buf, buf, L_SUBFR, st->mem_hf3, 1);
ener = 0.001f;
for (i=0; i>2];
/* quantize gain at: -10.5, -7.5, -4.5, -1.5, +1.5, +4.5, +7.5, +10.5 dB */
tmp = gain_hf[i] - gain;
index = (int)floor(((10.5 + tmp) / 3.0) + 0.5);
if (index > 7) {
index = 7;
}
if (index < 0) {
index = 0;
}
prm[i] = index;
gain_hf[i] = gain + (3.0f*((float)index)) - 10.5f;
}
prm += nsf;
}
else if (mode == 2) /* TCX 40 */
{
for (i=0; i<4; i++)
{
tmp = 0.0;
for (j=0; j<2; j++) {
tmp += gain_hf[(2*i)+j];
}
gain4[i] = 0.5f*tmp;
}
q_gain_hf(gain4, gain4, prm);
prm++;
for (i=0; i>1];
/* quantize gain at: -4.5, -1.5, +1.5, +4.5 dB */
tmp = gain_hf[i] - gain;
index = (int)floor(((4.5 + tmp) / 3.0) + 0.5);
if (index > 3) {
index = 3;
}
if (index < 0) {
index = 0;
}
prm[i] = index;
gain_hf[i] = gain + (3.0f*((float)index)) - 4.5f;
}
prm += nsf;
}
else /* ACELP/TCX 20 */
{
q_gain_hf(gain_hf, gain_hf, prm);
prm++;
}
if (mode == 2) {
k++;
} else if (mode == 3) {
k+=3;
}
}
return;
}