mirror of
https://github.com/RetroDECK/Supermodel.git
synced 2024-11-23 06:15:37 +00:00
325 lines
8.9 KiB
C++
325 lines
8.9 KiB
C++
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/* this file is a part of amp software, (C) tomislav uzelac 1996,1997
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*/
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/* layer2.c MPEG audio layer2 support
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*
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* Created by: Tomislav Uzelac Mar 1996
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* merged with amp, May 19 1997
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*/
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#include "amp.h"
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#include "audio.h"
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#include "getbits.h"
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#include "transform.h"
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#define LAYER2
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#include "layer2.h"
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int layer2_frame(struct AUDIO_HEADER *header,int cnt)
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{
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int i,s,sb,ch,gr,bitrate,bound=0;
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char (*nbal)[],(*bit_alloc_index)[][16];
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unsigned char allocation[2][32];
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unsigned char scfsi[2][32];
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float scalefactor[2][32][3];
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float subband_sample[2][32][36];
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int sblimit,nlevels,grouping;
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float c,d;
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int no_of_bits,mpi;
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unsigned short sb_sample_buf[3];
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int hsize,fs,mean_frame_size;
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bit_alloc_index=(char (*)[][16])&t_alloc0;
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nbal=(char (*)[])&t_nbal0; // shut up compiler
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sblimit = 0;
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hsize=4;
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if (header->protection_bit==0) hsize+=2;
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bitrate=t_bitrate[header->ID][3-header->layer][header->bitrate_index];
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fs=t_sampling_frequency[header->ID][header->sampling_frequency];
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if (header->ID) mean_frame_size=144000*bitrate/fs;
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else mean_frame_size=72000*bitrate/fs;
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/* layers 1 and 2 do not have a 'bit reservoir'
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*/
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append=data=0;
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fillbfr(mean_frame_size + header->padding_bit - hsize);
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switch (header->mode)
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{
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case 0 :
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case 2 : nch=2; bound=32; bitrate=bitrate/2;
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break;
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case 3 : nch=1; bound=32;
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break;
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case 1 : nch=2; bitrate=bitrate/2; bound=(header->mode_extension+1)*4;
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}
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if (header->ID==1) switch (header->sampling_frequency) {
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case 0 : switch (bitrate) /* 0 = 44.1 kHz */
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{
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case 56 :
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case 64 :
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case 80 : bit_alloc_index=(char (*)[][16])&t_alloc0;
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nbal=(char (*)[])&t_nbal0;
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sblimit=27;
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break;
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case 96 :
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case 112 :
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case 128 :
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case 160 :
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case 192 : bit_alloc_index=(char (*)[][16])&t_alloc1;
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nbal=(char (*)[])&t_nbal1;
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sblimit=30;
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break;
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case 32 :
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case 48 : bit_alloc_index=(char (*)[][16])&t_alloc2;
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nbal=(char (*)[])&t_nbal2;
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sblimit=8;
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break;
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default : printf(" bit alloc info no gud ");
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}
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break;
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case 1 : switch (bitrate) /* 1 = 48 kHz */
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{
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case 56 :
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case 64 :
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case 80 :
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case 96 :
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case 112 :
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case 128 :
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case 160 :
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case 192 : bit_alloc_index=(char (*)[][16])&t_alloc0;
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nbal=(char (*)[])&t_nbal0;
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sblimit=27;
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break;
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case 32 :
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case 48 : bit_alloc_index=(char (*)[][16])&t_alloc2;
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nbal=(char (*)[])&t_nbal2;
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sblimit=8;
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break;
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default : printf(" bit alloc info no gud ");
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}
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break;
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case 2 : switch (bitrate) /* 2 = 32 kHz */
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{
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case 56 :
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case 64 :
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case 80 : bit_alloc_index=(char (*)[][16])&t_alloc0;
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nbal=(char (*)[])&t_nbal0;
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sblimit=27;
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break;
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case 96 :
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case 112 :
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case 128 :
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case 160 :
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case 192 : bit_alloc_index=(char (*)[][16])&t_alloc1;
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nbal=(char (*)[])&t_nbal1;
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sblimit=30;
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break;
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case 32 :
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case 48 : bit_alloc_index=(char (*)[][16])&t_alloc3;
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nbal=(char (*)[])&t_nbal3;
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sblimit=12;
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break;
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default : printf("bit alloc info not ok\n");
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}
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break;
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default : printf("sampling freq. not ok/n");
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} else {
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bit_alloc_index=(char (*)[][16])&t_allocMPG2;
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nbal=(char (*)[])&t_nbalMPG2;
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sblimit=30;
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}
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/*
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* bit allocation per subband per channel decoding *****************************
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*/
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if (bound==32) bound=sblimit; /* bound=32 means there is no intensity stereo */
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for (sb=0;sb<bound;sb++)
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for (ch=0;ch<nch;ch++)
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allocation[ch][sb]=getbits((*nbal)[sb]);
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for (sb=bound;sb<sblimit;sb++)
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allocation[1][sb] = allocation[0][sb] = getbits((*nbal)[sb]);
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/*
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* scfsi ***********************************************************************
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*/
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for (sb=0;sb<sblimit;sb++)
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for (ch=0;ch<nch;ch++)
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if (allocation[ch][sb]!=0) scfsi[ch][sb]=getbits(2);
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else scfsi[ch][sb]=0;
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/*
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* scalefactors ****************************************************************
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*/
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for (sb=0;sb<sblimit;sb++)
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for (ch=0;ch<nch;ch++)
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if (allocation[ch][sb]!=0) {
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scalefactor[ch][sb][0]=(float)t_scalefactor[getbits(6)];
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switch (scfsi[ch][sb])
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{
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case 0: scalefactor[ch][sb][1]=(float)t_scalefactor[getbits(6)];
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scalefactor[ch][sb][2]=(float)t_scalefactor[getbits(6)];
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break;
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case 1: scalefactor[ch][sb][2]=(float)t_scalefactor[getbits(6)];
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scalefactor[ch][sb][1]=scalefactor[ch][sb][0];
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break;
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case 2: scalefactor[ch][sb][1]=(float)scalefactor[ch][sb][0];
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scalefactor[ch][sb][2]=scalefactor[ch][sb][0];
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break;
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case 3: scalefactor[ch][sb][2]=(float)t_scalefactor[getbits(6)];
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scalefactor[ch][sb][1]=scalefactor[ch][sb][2];
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}
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}
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else scalefactor[ch][sb][0]=scalefactor[ch][sb][1]=\
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scalefactor[ch][sb][2]=0.0;
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/*
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* samples *********************************************************************
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*/
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for (gr=0;gr<12;gr++) {
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/*
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* normal ********************************
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*/
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for (sb=0;sb<bound;sb++)
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for (ch=0;ch<nch;ch++)
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if (allocation[ch][sb]!=0) {
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mpi=(*bit_alloc_index)[sb][allocation[ch][sb]];
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no_of_bits=t_bpc[mpi];
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c=(float)t_c[mpi];
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d=(float)t_d[mpi];
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grouping=t_grouping[mpi];
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nlevels=t_nlevels[mpi];
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if (grouping) {
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int samplecode=getbits(no_of_bits);
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convert_samplecode(samplecode,grouping,sb_sample_buf);
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for (s=0;s<3;s++)
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subband_sample[ch][sb][3*gr+s]=requantize_sample (sb_sample_buf[s],nlevels,c,d,scalefactor[ch][sb][gr/4]);
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} else {
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for (s=0;s<3;s++) sb_sample_buf[s]=getbits(no_of_bits);
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for (s=0;s<3;s++) {
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/*subband_sample[ch][sb][3*gr+s]=requantize_sample (sb_sample_buf[s],nlevels,c,d,scalefactor[ch][sb][gr/4]);*/
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subband_sample[ch][sb][3*gr+s]=(t_dd[mpi]+sb_sample_buf[s]*t_nli[mpi])*c*scalefactor[ch][sb][gr>>2];
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}
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}
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} else
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for (s=0;s<3;s++) subband_sample[ch][sb][3*gr+s]=0;
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/*
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* joint stereo ********************************************
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*/
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for (sb=bound;sb<sblimit;sb++)
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if (allocation[0][sb]!=0) {
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/*ispravka!
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*/
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mpi=(*bit_alloc_index)[sb][allocation[0][sb]];
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no_of_bits=t_bpc[mpi];
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c=(float)t_c[mpi];
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d=(float)t_d[mpi];
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grouping=t_grouping[mpi];
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nlevels=t_nlevels[mpi];
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if (grouping) {
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int samplecode=getbits(no_of_bits);
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convert_samplecode(samplecode,grouping,sb_sample_buf);
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for (s=0;s<3;s++) {
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subband_sample[0][sb][3*gr+s]=requantize_sample (sb_sample_buf[s],nlevels,c,d,scalefactor[0][sb][gr/4]);
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subband_sample[1][sb][3*gr+s]=subband_sample[0][sb][3*gr+s];
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}
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} else {
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for (s=0;s<3;s++) sb_sample_buf[s]=getbits(no_of_bits);
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for (s=0;s<3;s++) {
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subband_sample[0][sb][3*gr+s]=subband_sample[1][sb][3*gr+s]=\
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(t_dd[mpi]+sb_sample_buf[s]*t_nli[mpi])*c*scalefactor[0][sb][gr>>2];
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}
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}
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} else for (s=0;s<3;s++) {
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subband_sample[0][sb][3*gr+s]=0;
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subband_sample[1][sb][3*gr+s]=0;
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}
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/*
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* the rest *******************************************
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*/
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for (sb=sblimit;sb<32;sb++)
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for (ch=0;ch<nch;ch++)
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for (s=0;s<3;s++) subband_sample[ch][sb][3*gr+s]=0;
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}
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/*
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* this is, in fact, horrible, but I had to adjust it to amp/mp3. The hack to make downmixing
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* work is as ugly as possible.
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*/
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if (A_DOWNMIX && header->mode!=3) {
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for (ch=0;ch<nch;ch++)
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for (sb=0;sb<32;sb++)
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for (i=0;i<36;i++)
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subband_sample[0][sb][i]=(subband_sample[0][sb][i]+subband_sample[1][sb][i])*0.5f;
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nch=1;
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}
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for (ch=0;ch<nch;ch++) {
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for (sb=0;sb<32;sb++)
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for (i=0;i<18;i++) res[sb][i]=subband_sample[ch][sb][i];
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for (i=0;i<18;i++)
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poly(ch,i);
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}
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printout();
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for (ch=0;ch<nch;ch++) {
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for (sb=0;sb<32;sb++)
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for (i=0;i<18;i++) res[sb][i]=subband_sample[ch][sb][i+18];
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for (i=0;i<18;i++)
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poly(ch,i);
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}
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printout();
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if (A_DOWNMIX && header->mode!=3) nch=2;
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return 0;
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}
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/****************************************************************************/
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/****************************************************************************/
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void convert_samplecode(unsigned int samplecode,unsigned int nlevels,unsigned short* sb_sample_buf)
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{
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int i;
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for (i=0;i<3;i++) {
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*sb_sample_buf=samplecode%nlevels;
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samplecode=samplecode/nlevels;
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sb_sample_buf++;
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}
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}
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float requantize_sample(unsigned short s4,unsigned short nlevels,float c,float d,float factor)
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{
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register float s,s2,s3;
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s3=(float) (-1.0+s4*2.0/(nlevels+1));
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s2=c*(s3+d);
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s=factor*s2;
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return s;
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}
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