www.pudn.com > AVS_M_ver10.rar > dec_hf.c, change:2007-08-21,size:11157b
/*
***********************************************************************
* 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"
/*-----------------------------------------------------------------*
* Funtion init_decoder_hf *
* ~~~~~~~~~~~~~~~~~~~~~~~~ *
* *
* ->Initialization of variables for the decoder section. *
* (band 6400Hz to 10800Hz). *
* *
*-----------------------------------------------------------------*/
void init_decoder_hf(Decoder_StState *st)
{
int i;
/* Static vectors to zero */
set_zero(st->mem_syn_hf, MHF);
set_zero(st->past_q_isf_hf, MHF);
set_zero(st->past_q_isf_hf_other, MHF);
st->past_q_gain_hf = 0.0;
st->past_q_gain_hf_other = 0.0;
st->old_gain = 1.0;
st->threshold = 0.0;
/* Initialize the ISPs */
for (i=0; iispold_hf[i] = (float)cos(3.141592654*(float)(i+1)/(float)MHF);
}
st->ispold_hf[MHF-1] = 0.045f;
st->lp_amp = 0.0;
return;
}
/*-----------------------------------------------------------------*
* Funtion decoder_lf *
* ~~~~~~~~~~~~~~~~~~ *
* ->Principle decoder routine (band 6400Hz to 10800Hz). *
* *
*-----------------------------------------------------------------*/
void decoder_hf(
int mod[], /* (i) : mode for each 20ms frame (mode[4] */
int param[], /* (i) : parameters */
int param_other[], /* (i) : parameters for the right channel in case of mono decoding with stereo bitstream */
int mono_dec_stereo, /* 1=Mono decoding with stereo bitstream */
int bad_frame[], /* (i) : for each frame (bad_frame[4]) */
float AqLF[], /* (i) : decoded coefficients (AdLF[16]) */
float exc[], /* (i) : decoded excitation */
float synth_hf[], /* (o) : decoded synthesis */
float mem_lpc_hf[],
float *mem_gain_hf,
short *ramp_state,
Decoder_StState *st) /* i/o : decoder memory state */
{
/* LPC coefficients */
float ispnew[MHF];
float isfnew[MHF];
float Aq[(NB_SUBFR+1)*(MHF+1)]; /* A(z) for all subframes */
float *p_Aq;
float gain4[4], gain_hf[NB_SUBFR], gain_hf_other[NB_SUBFR];
float HF[L_SUBFR];
/* Scalars */
int i, k, mode, bfi, sf, nsf, *prm, *prm_other;
int i_subfr, index;
float gain, tmp;
int *prm_in_other;
float gdiff;
float thr_lo, thr_hi;
float gain_sum, gain_sum_other;
float gain_tmp[NB_SUBFR];
float isfnew_other[MHF];
/*----------- DECODE AND SYNTHESIZE UPPER-BAND SIGNAL (synth_hf[])
AND ADD IT TO UPSAMPLED LOWER-BAND SIGNAL (synthesis) ----------
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 decoding depends on bit allocation to HFs (nbits_hf) :
nbits_hf=0:
the lower-band signal post-filtered and upsampled is the actual synthesis
(no HF decoding and synthesis)
nbits_hf=4*16:
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)
*/
/* compute Ai (Aq) and correction scale factor (gain) for each subframe */
for (k=0; kpast_q_isf_hf, bfi, st->mean_isf_hf, st->dico1_isf_hf);
if(mono_dec_stereo == 0)
{
isf2isp(isfnew, ispnew, MHF);
/* interpolate Ai in ISP domain (Aq) */
int_lpc_np1(st->ispold_hf, ispnew, Aq, nsf, MHF);
mvr2r(ispnew, st->ispold_hf, MHF);
/* compute gain correction factor to match amplitude at 6.4kHz */
gain = match_gain_6k4(&AqLF[((k*4)+nsf)*(M+1)], &Aq[nsf*(MHF+1)]);
/* interpolate per subframe */
int_gain(st->old_gain, gain, gain_hf, nsf);
st->old_gain = gain;
}
/* decode 4 gains */
d_gain_hf(prm[2], gain4, &(st->past_q_gain_hf), bfi);
prm += 3;
/* decode gain correction offsets (in dB) for each subframe */
if (mode == 3) { /* 80+10-ms TCX */
/* decode gain offset: -10.5, -7.5, -4.5, -1.5, +1.5, +4.5, +7.5, +10.5 dB */
for (i=0; i<16; i++) {
index = *prm++;
tmp = (3.0f*((float)index)) - 10.5f;
if (bfi) {
tmp = 0.0;
}
if (bad_frame[k+1] && (i<8)) {
tmp = 0.0;
}
if (bad_frame[k+2] && (i>=8)) {
tmp = 0.0;
}
if (mono_dec_stereo == 1)
{
gain_hf[i] = tmp + gain4[(i>>2)];
gain_sum += gain_hf[i];
}
else
{
gain_hf[i] += tmp + gain4[(i>>2)];
}
}
} else if (mode == 2) { /* 40+5-ms TCX */
/* decode gain offset: -4.5, -1.5, +1.5, +4.5 dB */
for (i=0; i<8; i++) {
index = *prm++;
tmp = (3.0f*((float)index)) - 4.5f;
if (bfi || bad_frame[k+1]) {
tmp = 0.0;
}
if (mono_dec_stereo == 1)
{
gain_hf[i] = tmp + gain4[(i>>1)];
gain_sum += gain_hf[i];
}
else
{
gain_hf[i] += tmp + gain4[(i>>1)];
}
}
} else { /* ACELP or 20+2.5-ms TCX */
if (mono_dec_stereo == 1)
{
for (i=0; i<4; i++) {
gain_hf[i] = gain4[i];
gain_sum += gain_hf[i];
}
}
else
{
for (i=0; i<4; i++) {
gain_hf[i] += gain4[i];
}
}
}
if(mono_dec_stereo == 1)
{
/* decode 4 gains */
d_gain_hf(prm_other[2], gain4, &(st->past_q_gain_hf_other), bfi);
prm_other += 3;
/* decode gain correction offsets (in dB) for each subframe */
if (mode == 3) { /* 80+10-ms TCX */
/* decode gain offset: -10.5, -7.5, -4.5, -1.5, +1.5, +4.5, +7.5, +10.5 dB */
for (i=0; i<16; i++) {
index = *prm_other++;
tmp = (3.0f*((float)index)) - 10.5f;
if (bfi) {
tmp = 0.0;
}
if (bad_frame[k+1] && (i<8)) {
tmp = 0.0;
}
if (bad_frame[k+2] && (i>=8)) {
tmp = 0.0;
}
gain_hf_other[i] = tmp + gain4[(i>>2)];
gain_sum_other += gain_hf_other[i];
}
} else if (mode == 2) { /* 40+5-ms TCX */
/* decode gain offset: -4.5, -1.5, +1.5, +4.5 dB */
for (i=0; i<8; i++) {
index = *prm_other++;
tmp = (3.0f*((float)index)) - 4.5f;
if (bfi || bad_frame[k+1]) {
tmp = 0.0;
}
gain_hf_other[i] = tmp + gain4[(i>>1)];
gain_sum_other += gain_hf_other[i];
}
} else { /* ACELP or 20+2.5-ms TCX */
for (i=0; i<4; i++) {
gain_hf_other[i] = gain4[i];
gain_sum_other += gain_hf_other[i];
}
}
thr_lo = -20.0f * (float)nsf;
thr_hi = 20.0f * (float)nsf;
gdiff = gain_sum - gain_sum_other;
/* decode ISFs and convert ISFs to cosine domain */
d_isf_hf(prm_in_other, isfnew_other, st->past_q_isf_hf_other, bfi, st->mean_isf_hf, st->dico1_isf_hf);
if (gdiff < thr_lo)
{
for (i = 0; i < MHF; i++)
{
isfnew[i] = isfnew_other[i];
}
}
else if (gdiff > thr_hi)
{
}
else
{
for (i = 0; i < MHF; i++)
{
isfnew[i] = 0.5f * (isfnew[i] + isfnew_other[i]);
}
}
isf2isp(isfnew, ispnew, MHF);
int_lpc_np1(st->ispold_hf, ispnew, Aq, nsf, MHF);
mvr2r(ispnew, st->ispold_hf, MHF);
/* compute gain correction factor to match amplitude at 6.4kHz */
gain = match_gain_6k4(&AqLF[((k*4)+nsf)*(M+1)], &Aq[nsf*(MHF+1)]);
/* interpolate per subframe */
int_gain(st->old_gain, gain, gain_tmp, nsf);
for (i = 0; i < nsf; i++)
{
gain_hf[i] += gain_tmp[i];
gain_hf_other[i] += gain_tmp[i];
}
st->old_gain = gain;
}
for (i = 0; i < MHF+1; i++)
{
mem_lpc_hf[i] = Aq[nsf*(MHF+1)+i];
}
/* HF synthesis and smooting */
p_Aq = Aq;
for (sf=0; sflp_amp));
/* filtering through HF LP filter */
E_UTIL_synthesisPlus(p_Aq, MHF, HF, HF, L_SUBFR, st->mem_syn_hf, 1);
/* smooth energy of HF signal on subframe (HF[]) */
smooth_ener_hf(HF, &(st->threshold));
for (i=0; i