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Supermodel/Src/Sound/MPEG/layer2.cpp

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/**
** Supermodel
** A Sega Model 3 Arcade Emulator.
** Copyright 2011 Bart Trzynadlowski, Nik Henson
**
** This file is part of Supermodel.
**
** Supermodel is free software: you can redistribute it and/or modify it under
** the terms of the GNU General Public License as published by the Free
** Software Foundation, either version 3 of the License, or (at your option)
** any later version.
**
** Supermodel is distributed in the hope that it will be useful, but WITHOUT
** ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
** FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for
** more details.
**
** You should have received a copy of the GNU General Public License along
** with Supermodel. If not, see <http://www.gnu.org/licenses/>.
**/
/*
* layer2.cpp
*
* Amp library internal module.
*/
/* this file is a part of amp software, (C) tomislav uzelac 1996,1997
*/
/* layer2.c MPEG audio layer2 support
*
* Created by: Tomislav Uzelac Mar 1996
* merged with amp, May 19 1997
*/
#include "amp.h"
#include "audio.h"
#include "getbits.h"
#include "transform.h"
#define LAYER2
#include "layer2.h"
#include "Supermodel.h"
int layer2_frame(struct AUDIO_HEADER *header,int cnt)
{
int i,s,sb,ch,gr,bitrate,bound=0;
char (*nbal)[],(*bit_alloc_index)[][16];
unsigned char allocation[2][32];
unsigned char scfsi[2][32];
float scalefactor[2][32][3];
float subband_sample[2][32][36];
int sblimit,nlevels,grouping;
float c,d;
int no_of_bits,mpi;
unsigned short sb_sample_buf[3];
int hsize,fs,mean_frame_size;
bit_alloc_index=(char (*)[][16])&t_alloc0;
nbal=(char (*)[])&t_nbal0; // shut up compiler
sblimit = 0;
hsize=4;
if (header->protection_bit==0) hsize+=2;
bitrate=t_bitrate[header->ID][3-header->layer][header->bitrate_index];
fs=t_sampling_frequency[header->ID][header->sampling_frequency];
if (header->ID) mean_frame_size=144000*bitrate/fs;
else mean_frame_size=72000*bitrate/fs;
/* layers 1 and 2 do not have a 'bit reservoir'
*/
append=data=0;
fillbfr(mean_frame_size + header->padding_bit - hsize);
switch (header->mode)
{
case 0 :
case 2 : nch=2; bound=32; bitrate=bitrate/2;
break;
case 3 : nch=1; bound=32;
break;
case 1 : nch=2; bitrate=bitrate/2; bound=(header->mode_extension+1)*4;
}
if (header->ID==1) switch (header->sampling_frequency) {
case 0 : switch (bitrate) /* 0 = 44.1 kHz */
{
case 56 :
case 64 :
case 80 : bit_alloc_index=(char (*)[][16])&t_alloc0;
nbal=(char (*)[])&t_nbal0;
sblimit=27;
break;
case 96 :
case 112 :
case 128 :
case 160 :
case 192 : bit_alloc_index=(char (*)[][16])&t_alloc1;
nbal=(char (*)[])&t_nbal1;
sblimit=30;
break;
case 32 :
case 48 : bit_alloc_index=(char (*)[][16])&t_alloc2;
nbal=(char (*)[])&t_nbal2;
sblimit=8;
break;
default : ErrorLog("Internal error in MPEG decoder (%s:%d).", __FILE__, __LINE__);
}
break;
case 1 : switch (bitrate) /* 1 = 48 kHz */
{
case 56 :
case 64 :
case 80 :
case 96 :
case 112 :
case 128 :
case 160 :
case 192 : bit_alloc_index=(char (*)[][16])&t_alloc0;
nbal=(char (*)[])&t_nbal0;
sblimit=27;
break;
case 32 :
case 48 : bit_alloc_index=(char (*)[][16])&t_alloc2;
nbal=(char (*)[])&t_nbal2;
sblimit=8;
break;
default : ErrorLog("Internal error in MPEG decoder (%s:%d).", __FILE__, __LINE__);
}
break;
case 2 : switch (bitrate) /* 2 = 32 kHz */
{
case 56 :
case 64 :
case 80 : bit_alloc_index=(char (*)[][16])&t_alloc0;
nbal=(char (*)[])&t_nbal0;
sblimit=27;
break;
case 96 :
case 112 :
case 128 :
case 160 :
case 192 : bit_alloc_index=(char (*)[][16])&t_alloc1;
nbal=(char (*)[])&t_nbal1;
sblimit=30;
break;
case 32 :
case 48 : bit_alloc_index=(char (*)[][16])&t_alloc3;
nbal=(char (*)[])&t_nbal3;
sblimit=12;
break;
default : ErrorLog("Internal error in MPEG decoder (%s:%d).", __FILE__, __LINE__);
}
break;
default : ErrorLog("Internal error in MPEG decoder (%s:%d).", __FILE__, __LINE__); // sampling frequency no good
} else {
bit_alloc_index=(char (*)[][16])&t_allocMPG2;
nbal=(char (*)[])&t_nbalMPG2;
sblimit=30;
}
/*
* bit allocation per subband per channel decoding *****************************
*/
if (bound==32) bound=sblimit; /* bound=32 means there is no intensity stereo */
for (sb=0;sb<bound;sb++)
for (ch=0;ch<nch;ch++)
allocation[ch][sb]=getbits((*nbal)[sb]);
for (sb=bound;sb<sblimit;sb++)
allocation[1][sb] = allocation[0][sb] = getbits((*nbal)[sb]);
/*
* scfsi ***********************************************************************
*/
for (sb=0;sb<sblimit;sb++)
for (ch=0;ch<nch;ch++)
if (allocation[ch][sb]!=0) scfsi[ch][sb]=getbits(2);
else scfsi[ch][sb]=0;
/*
* scalefactors ****************************************************************
*/
for (sb=0;sb<sblimit;sb++)
for (ch=0;ch<nch;ch++)
if (allocation[ch][sb]!=0) {
scalefactor[ch][sb][0]=(float)t_scalefactor[getbits(6)];
switch (scfsi[ch][sb])
{
case 0: scalefactor[ch][sb][1]=(float)t_scalefactor[getbits(6)];
scalefactor[ch][sb][2]=(float)t_scalefactor[getbits(6)];
break;
case 1: scalefactor[ch][sb][2]=(float)t_scalefactor[getbits(6)];
scalefactor[ch][sb][1]=scalefactor[ch][sb][0];
break;
case 2: scalefactor[ch][sb][1]=(float)scalefactor[ch][sb][0];
scalefactor[ch][sb][2]=scalefactor[ch][sb][0];
break;
case 3: scalefactor[ch][sb][2]=(float)t_scalefactor[getbits(6)];
scalefactor[ch][sb][1]=scalefactor[ch][sb][2];
}
}
else scalefactor[ch][sb][0]=scalefactor[ch][sb][1]=\
scalefactor[ch][sb][2]=0.0;
/*
* samples *********************************************************************
*/
for (gr=0;gr<12;gr++) {
/*
* normal ********************************
*/
for (sb=0;sb<bound;sb++)
for (ch=0;ch<nch;ch++)
if (allocation[ch][sb]!=0) {
mpi=(*bit_alloc_index)[sb][allocation[ch][sb]];
no_of_bits=t_bpc[mpi];
c=(float)t_c[mpi];
d=(float)t_d[mpi];
grouping=t_grouping[mpi];
nlevels=t_nlevels[mpi];
if (grouping) {
int samplecode=getbits(no_of_bits);
convert_samplecode(samplecode,grouping,sb_sample_buf);
for (s=0;s<3;s++)
subband_sample[ch][sb][3*gr+s]=requantize_sample (sb_sample_buf[s],nlevels,c,d,scalefactor[ch][sb][gr/4]);
} else {
for (s=0;s<3;s++) sb_sample_buf[s]=getbits(no_of_bits);
for (s=0;s<3;s++) {
/*subband_sample[ch][sb][3*gr+s]=requantize_sample (sb_sample_buf[s],nlevels,c,d,scalefactor[ch][sb][gr/4]);*/
subband_sample[ch][sb][3*gr+s]=(t_dd[mpi]+sb_sample_buf[s]*t_nli[mpi])*c*scalefactor[ch][sb][gr>>2];
}
}
} else
for (s=0;s<3;s++) subband_sample[ch][sb][3*gr+s]=0;
/*
* joint stereo ********************************************
*/
for (sb=bound;sb<sblimit;sb++)
if (allocation[0][sb]!=0) {
/*ispravka!
*/
mpi=(*bit_alloc_index)[sb][allocation[0][sb]];
no_of_bits=t_bpc[mpi];
c=(float)t_c[mpi];
d=(float)t_d[mpi];
grouping=t_grouping[mpi];
nlevels=t_nlevels[mpi];
if (grouping) {
int samplecode=getbits(no_of_bits);
convert_samplecode(samplecode,grouping,sb_sample_buf);
for (s=0;s<3;s++) {
subband_sample[0][sb][3*gr+s]=requantize_sample (sb_sample_buf[s],nlevels,c,d,scalefactor[0][sb][gr/4]);
subband_sample[1][sb][3*gr+s]=subband_sample[0][sb][3*gr+s];
}
} else {
for (s=0;s<3;s++) sb_sample_buf[s]=getbits(no_of_bits);
for (s=0;s<3;s++) {
subband_sample[0][sb][3*gr+s]=subband_sample[1][sb][3*gr+s]=\
(t_dd[mpi]+sb_sample_buf[s]*t_nli[mpi])*c*scalefactor[0][sb][gr>>2];
}
}
} else for (s=0;s<3;s++) {
subband_sample[0][sb][3*gr+s]=0;
subband_sample[1][sb][3*gr+s]=0;
}
/*
* the rest *******************************************
*/
for (sb=sblimit;sb<32;sb++)
for (ch=0;ch<nch;ch++)
for (s=0;s<3;s++) subband_sample[ch][sb][3*gr+s]=0;
}
/*
* this is, in fact, horrible, but I had to adjust it to amp/mp3. The hack to make downmixing
* work is as ugly as possible.
*/
if (A_DOWNMIX && header->mode!=3) {
for (ch=0;ch<nch;ch++)
for (sb=0;sb<32;sb++)
for (i=0;i<36;i++)
subband_sample[0][sb][i]=(subband_sample[0][sb][i]+subband_sample[1][sb][i])*0.5f;
nch=1;
}
for (ch=0;ch<nch;ch++) {
for (sb=0;sb<32;sb++)
for (i=0;i<18;i++) res[sb][i]=subband_sample[ch][sb][i];
for (i=0;i<18;i++)
poly(ch,i);
}
printout();
for (ch=0;ch<nch;ch++) {
for (sb=0;sb<32;sb++)
for (i=0;i<18;i++) res[sb][i]=subband_sample[ch][sb][i+18];
for (i=0;i<18;i++)
poly(ch,i);
}
printout();
if (A_DOWNMIX && header->mode!=3) nch=2;
return 0;
}
/****************************************************************************/
/****************************************************************************/
void convert_samplecode(unsigned int samplecode,unsigned int nlevels,unsigned short* sb_sample_buf)
{
int i;
for (i=0;i<3;i++) {
*sb_sample_buf=samplecode%nlevels;
samplecode=samplecode/nlevels;
sb_sample_buf++;
}
}
float requantize_sample(unsigned short s4,unsigned short nlevels,float c,float d,float factor)
{
register float s,s2,s3;
s3=(float) (-1.0+s4*2.0/(nlevels+1));
s2=c*(s3+d);
s=factor*s2;
return s;
}
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