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jkxr/Projects/Android/jni/OpenJK/code/mp3code/cup.c

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Initial Commit Opens in Android Studio but haven't even tried to build it yet (it won't.. I know that much!)
2022-09-18 15:37:21 +00:00
/*____________________________________________________________________________
FreeAmp - The Free MP3 Player
MP3 Decoder originally Copyright (C) 1995-1997 Xing Technology
Corp. http://www.xingtech.com
Portions Copyright (C) 1998-1999 EMusic.com
This program 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 2 of the License, or
(at your option) any later version.
This program 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 this program; if not, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
$Id: cup.c,v 1.3 1999/10/19 07:13:08 elrod Exp $
____________________________________________________________________________*/
/**** cup.c ***************************************************
MPEG audio decoder Layer I/II mpeg1 and mpeg2
should be portable ANSI C, should be endian independent
mod 2/21/95 2/21/95 add bit skip, sb limiting
mods 11/15/95 for Layer I
******************************************************************/
/******************************************************************
MPEG audio software decoder portable ANSI c.
Decodes all Layer I/II to 16 bit linear pcm.
Optional stereo to mono conversion. Optional
output sample rate conversion to half or quarter of
native mpeg rate. dec8.c adds oupuut conversion features.
-------------------------------------
int audio_decode_init(MPEG_HEAD *h, int framebytes_arg,
int reduction_code, int transform_code, int convert_code,
int freq_limit)
initilize decoder:
return 0 = fail, not 0 = success
MPEG_HEAD *h input, mpeg header info (returned by call to head_info)
pMP3Stream->framebytes input, mpeg frame size (returned by call to head_info)
reduction_code input, sample rate reduction code
0 = full rate
1 = half rate
2 = quarter rate
transform_code input, ignored
convert_code input, channel conversion
convert_code: 0 = two chan output
1 = convert two chan to mono
2 = convert two chan to left chan
3 = convert two chan to right chan
freq_limit input, limits bandwidth of pcm output to specified
frequency. Special use. Set to 24000 for normal use.
---------------------------------
void audio_decode_info( DEC_INFO *info)
information return:
Call after audio_decode_init. See mhead.h for
information returned in DEC_INFO structure.
---------------------------------
IN_OUT audio_decode(unsigned char *bs, void *pcmbuf)
decode one mpeg audio frame:
bs input, mpeg bitstream, must start with
sync word. Caution: may read up to 3 bytes
beyond end of frame.
pcmbuf output, pcm samples.
IN_OUT structure returns:
Number bytes conceptually removed from mpeg bitstream.
Returns 0 if sync loss.
Number bytes of pcm output.
*******************************************************************/
#include <stdlib.h>
#include <stdio.h>
#include <float.h>
#include <math.h>
#include "mhead.h" /* mpeg header structure */
#include "mp3struct.h"
/*-------------------------------------------------------
NOTE: Decoder may read up to three bytes beyond end of
frame. Calling application must ensure that this does
not cause a memory access violation (protection fault)
---------------------------------------------------------*/
/*====================================================================*/
/*----------------*/
//@@@@ This next one (decinfo) is ok:
DEC_INFO decinfo; /* global for Layer III */ // only written into by decode init funcs, then copied to stack struct higher up
/*----------------*/
static float look_c_value[18]; /* built by init */ // effectively constant
/*----------------*/
////@@@@static int pMP3Stream->outbytes; // !!!!!!!!!!!!!!?
////@@@@static int pMP3Stream->framebytes; // !!!!!!!!!!!!!!!!
////@@@@static int pMP3Stream->outvalues; // !!!!!!!!!!!!?
////@@@@static int pad;
static const int look_joint[16] =
{ /* lookup stereo sb's by mode+ext */
64, 64, 64, 64, /* stereo */
2 * 4, 2 * 8, 2 * 12, 2 * 16, /* joint */
64, 64, 64, 64, /* dual */
32, 32, 32, 32, /* mono */
};
/*----------------*/
////@@@@static int max_sb; // !!!!!!!!! L1, 2 3
////@@@@static int stereo_sb;
/*----------------*/
////@@@@static int pMP3Stream->nsb_limit = 6;
////@@@@static int bit_skip;
static const int bat_bit_master[] =
{
0, 5, 7, 9, 10, 12, 15, 18, 21, 24, 27, 30, 33, 36, 39, 42, 45, 48};
/*----------------*/
////@@@@static int nbat[4] = {3, 8, 12, 7}; // !!!!!!!!!!!!! not constant!!!!
////@@@@static int bat[4][16]; // built as constant, but built according to header type (sigh)
static int ballo[64]; /* set by unpack_ba */ // scratchpad
static unsigned int samp_dispatch[66]; /* set by unpack_ba */ // scratchpad?
static float c_value[64]; /* set by unpack_ba */ // scratchpad
/*----------------*/
static unsigned int sf_dispatch[66]; /* set by unpack_ba */ // scratchpad?
static float sf_table[64]; // effectively constant
////@@@@ static float cs_factor[3][64];
/*----------------*/
////@@@@FINDME - groan.... (I shoved a *2 in just in case it needed it for stereo. This whole thing is crap now
float sample[2304*2]; /* global for use by Later 3 */ // !!!!!!!!!!!!!!!!!!!!!! // scratchpad?
static signed char group3_table[32][3]; // effectively constant
static signed char group5_table[128][3]; // effectively constant
static signed short group9_table[1024][3]; // effectively constant
/*----------------*/
////@@@@typedef void (*SBT_FUNCTION) (float *sample, short *pcm, int n);
void sbt_mono(float *sample, short *pcm, int n);
void sbt_dual(float *sample, short *pcm, int n);
////@@@@static SBT_FUNCTION sbt = sbt_mono;
typedef IN_OUT(*AUDIO_DECODE_ROUTINE) (unsigned char *bs, signed short *pcm);
IN_OUT L2audio_decode(unsigned char *bs, signed short *pcm);
static AUDIO_DECODE_ROUTINE audio_decode_routine = L2audio_decode;
/*======================================================================*/
/*======================================================================*/
/* get bits from bitstream in endian independent way */
////@@@@ FINDME - this stuff doesn't appear to be used by any of our samples (phew)
static unsigned char *bs_ptr;
static unsigned long bitbuf;
static int bits;
static long bitval;
/*------------- initialize bit getter -------------*/
static void load_init(unsigned char *buf)
{
bs_ptr = buf;
bits = 0;
bitbuf = 0;
}
/*------------- get n bits from bitstream -------------*/
static long load(int n)
{
unsigned long x;
if (bits < n)
{ /* refill bit buf if necessary */
while (bits <= 24)
{
bitbuf = (bitbuf << 8) | *bs_ptr++;
bits += 8;
}
}
bits -= n;
x = bitbuf >> bits;
bitbuf -= x << bits;
return x;
}
/*------------- skip over n bits in bitstream -------------*/
static void skip(int n)
{
int k;
if (bits < n)
{
n -= bits;
k = n >> 3;
/*--- bytes = n/8 --*/
bs_ptr += k;
n -= k << 3;
bitbuf = *bs_ptr++;
bits = 8;
}
bits -= n;
bitbuf -= (bitbuf >> bits) << bits;
}
/*--------------------------------------------------------------*/
#define mac_load_check(n) if( bits < (n) ) { \
while( bits <= 24 ) { \
bitbuf = (bitbuf << 8) | *bs_ptr++; \
bits += 8; \
} \
}
/*--------------------------------------------------------------*/
#define mac_load(n) ( bits -= n, \
bitval = bitbuf >> bits, \
bitbuf -= bitval << bits, \
bitval )
/*======================================================================*/
static void unpack_ba()
{
int i, j, k;
static int nbit[4] =
{4, 4, 3, 2};
int nstereo;
pMP3Stream->bit_skip = 0;
nstereo = pMP3Stream->stereo_sb;
k = 0;
for (i = 0; i < 4; i++)
{
for (j = 0; j < pMP3Stream->nbat[i]; j++, k++)
{
mac_load_check(4);
ballo[k] = samp_dispatch[k] = pMP3Stream->bat[i][mac_load(nbit[i])];
if (k >= pMP3Stream->nsb_limit)
pMP3Stream->bit_skip += bat_bit_master[samp_dispatch[k]];
c_value[k] = look_c_value[samp_dispatch[k]];
if (--nstereo < 0)
{
ballo[k + 1] = ballo[k];
samp_dispatch[k] += 18; /* flag as joint */
samp_dispatch[k + 1] = samp_dispatch[k]; /* flag for sf */
c_value[k + 1] = c_value[k];
k++;
j++;
}
}
}
samp_dispatch[pMP3Stream->nsb_limit] = 37; /* terminate the dispatcher with skip */
samp_dispatch[k] = 36; /* terminate the dispatcher */
}
/*-------------------------------------------------------------------------*/
static void unpack_sfs() /* unpack scale factor selectors */
{
int i;
for (i = 0; i < pMP3Stream->max_sb; i++)
{
mac_load_check(2);
if (ballo[i])
sf_dispatch[i] = mac_load(2);
else
sf_dispatch[i] = 4; /* no allo */
}
sf_dispatch[i] = 5; /* terminate dispatcher */
}
/*-------------------------------------------------------------------------*/
static void unpack_sf() /* unpack scale factor */
{ /* combine dequant and scale factors */
int i;
i = -1;
dispatch:switch (sf_dispatch[++i])
{
case 0: /* 3 factors 012 */
mac_load_check(18);
pMP3Stream->cs_factor[0][i] = c_value[i] * sf_table[mac_load(6)];
pMP3Stream->cs_factor[1][i] = c_value[i] * sf_table[mac_load(6)];
pMP3Stream->cs_factor[2][i] = c_value[i] * sf_table[mac_load(6)];
goto dispatch;
case 1: /* 2 factors 002 */
mac_load_check(12);
pMP3Stream->cs_factor[1][i] = pMP3Stream->cs_factor[0][i] = c_value[i] * sf_table[mac_load(6)];
pMP3Stream->cs_factor[2][i] = c_value[i] * sf_table[mac_load(6)];
goto dispatch;
case 2: /* 1 factor 000 */
mac_load_check(6);
pMP3Stream->cs_factor[2][i] = pMP3Stream->cs_factor[1][i] = pMP3Stream->cs_factor[0][i] =
c_value[i] * sf_table[mac_load(6)];
goto dispatch;
case 3: /* 2 factors 022 */
mac_load_check(12);
pMP3Stream->cs_factor[0][i] = c_value[i] * sf_table[mac_load(6)];
pMP3Stream->cs_factor[2][i] = pMP3Stream->cs_factor[1][i] = c_value[i] * sf_table[mac_load(6)];
goto dispatch;
case 4: /* no allo */
/*-- pMP3Stream->cs_factor[2][i] = pMP3Stream->cs_factor[1][i] = pMP3Stream->cs_factor[0][i] = 0.0; --*/
goto dispatch;
case 5: /* all done */
;
} /* end switch */
}
/*-------------------------------------------------------------------------*/
#define UNPACK_N(n) s[k] = pMP3Stream->cs_factor[i][k]*(load(n)-((1 << (n-1)) -1)); \
s[k+64] = pMP3Stream->cs_factor[i][k]*(load(n)-((1 << (n-1)) -1)); \
s[k+128] = pMP3Stream->cs_factor[i][k]*(load(n)-((1 << (n-1)) -1)); \
goto dispatch;
#define UNPACK_N2(n) mac_load_check(3*n); \
s[k] = pMP3Stream->cs_factor[i][k]*(mac_load(n)-((1 << (n-1)) -1)); \
s[k+64] = pMP3Stream->cs_factor[i][k]*(mac_load(n)-((1 << (n-1)) -1)); \
s[k+128] = pMP3Stream->cs_factor[i][k]*(mac_load(n)-((1 << (n-1)) -1)); \
goto dispatch;
#define UNPACK_N3(n) mac_load_check(2*n); \
s[k] = pMP3Stream->cs_factor[i][k]*(mac_load(n)-((1 << (n-1)) -1)); \
s[k+64] = pMP3Stream->cs_factor[i][k]*(mac_load(n)-((1 << (n-1)) -1)); \
mac_load_check(n); \
s[k+128] = pMP3Stream->cs_factor[i][k]*(mac_load(n)-((1 << (n-1)) -1)); \
goto dispatch;
#define UNPACKJ_N(n) tmp = (load(n)-((1 << (n-1)) -1)); \
s[k] = pMP3Stream->cs_factor[i][k]*tmp; \
s[k+1] = pMP3Stream->cs_factor[i][k+1]*tmp; \
tmp = (load(n)-((1 << (n-1)) -1)); \
s[k+64] = pMP3Stream->cs_factor[i][k]*tmp; \
s[k+64+1] = pMP3Stream->cs_factor[i][k+1]*tmp; \
tmp = (load(n)-((1 << (n-1)) -1)); \
s[k+128] = pMP3Stream->cs_factor[i][k]*tmp; \
s[k+128+1] = pMP3Stream->cs_factor[i][k+1]*tmp; \
k++; /* skip right chan dispatch */ \
goto dispatch;
/*-------------------------------------------------------------------------*/
static void unpack_samp() /* unpack samples */
{
int i, j, k;
float *s;
int n;
long tmp;
s = sample;
for (i = 0; i < 3; i++)
{ /* 3 groups of scale factors */
for (j = 0; j < 4; j++)
{
k = -1;
dispatch:switch (samp_dispatch[++k])
{
case 0:
s[k + 128] = s[k + 64] = s[k] = 0.0F;
goto dispatch;
case 1: /* 3 levels grouped 5 bits */
mac_load_check(5);
n = mac_load(5);
s[k] = pMP3Stream->cs_factor[i][k] * group3_table[n][0];
s[k + 64] = pMP3Stream->cs_factor[i][k] * group3_table[n][1];
s[k + 128] = pMP3Stream->cs_factor[i][k] * group3_table[n][2];
goto dispatch;
case 2: /* 5 levels grouped 7 bits */
mac_load_check(7);
n = mac_load(7);
s[k] = pMP3Stream->cs_factor[i][k] * group5_table[n][0];
s[k + 64] = pMP3Stream->cs_factor[i][k] * group5_table[n][1];
s[k + 128] = pMP3Stream->cs_factor[i][k] * group5_table[n][2];
goto dispatch;
case 3:
UNPACK_N2(3) /* 7 levels */
case 4: /* 9 levels grouped 10 bits */
mac_load_check(10);
n = mac_load(10);
s[k] = pMP3Stream->cs_factor[i][k] * group9_table[n][0];
s[k + 64] = pMP3Stream->cs_factor[i][k] * group9_table[n][1];
s[k + 128] = pMP3Stream->cs_factor[i][k] * group9_table[n][2];
goto dispatch;
case 5:
UNPACK_N2(4) /* 15 levels */
case 6:
UNPACK_N2(5) /* 31 levels */
case 7:
UNPACK_N2(6) /* 63 levels */
case 8:
UNPACK_N2(7) /* 127 levels */
case 9:
UNPACK_N2(8) /* 255 levels */
case 10:
UNPACK_N3(9) /* 511 levels */
case 11:
UNPACK_N3(10) /* 1023 levels */
case 12:
UNPACK_N3(11) /* 2047 levels */
case 13:
UNPACK_N3(12) /* 4095 levels */
case 14:
UNPACK_N(13) /* 8191 levels */
case 15:
UNPACK_N(14) /* 16383 levels */
case 16:
UNPACK_N(15) /* 32767 levels */
case 17:
UNPACK_N(16) /* 65535 levels */
/* -- joint ---- */
case 18 + 0:
s[k + 128 + 1] = s[k + 128] = s[k + 64 + 1] = s[k + 64] = s[k + 1] = s[k] = 0.0F;
k++; /* skip right chan dispatch */
goto dispatch;
case 18 + 1: /* 3 levels grouped 5 bits */
n = load(5);
s[k] = pMP3Stream->cs_factor[i][k] * group3_table[n][0];
s[k + 1] = pMP3Stream->cs_factor[i][k + 1] * group3_table[n][0];
s[k + 64] = pMP3Stream->cs_factor[i][k] * group3_table[n][1];
s[k + 64 + 1] = pMP3Stream->cs_factor[i][k + 1] * group3_table[n][1];
s[k + 128] = pMP3Stream->cs_factor[i][k] * group3_table[n][2];
s[k + 128 + 1] = pMP3Stream->cs_factor[i][k + 1] * group3_table[n][2];
k++; /* skip right chan dispatch */
goto dispatch;
case 18 + 2: /* 5 levels grouped 7 bits */
n = load(7);
s[k] = pMP3Stream->cs_factor[i][k] * group5_table[n][0];
s[k + 1] = pMP3Stream->cs_factor[i][k + 1] * group5_table[n][0];
s[k + 64] = pMP3Stream->cs_factor[i][k] * group5_table[n][1];
s[k + 64 + 1] = pMP3Stream->cs_factor[i][k + 1] * group5_table[n][1];
s[k + 128] = pMP3Stream->cs_factor[i][k] * group5_table[n][2];
s[k + 128 + 1] = pMP3Stream->cs_factor[i][k + 1] * group5_table[n][2];
k++; /* skip right chan dispatch */
goto dispatch;
case 18 + 3:
UNPACKJ_N(3) /* 7 levels */
case 18 + 4: /* 9 levels grouped 10 bits */
n = load(10);
s[k] = pMP3Stream->cs_factor[i][k] * group9_table[n][0];
s[k + 1] = pMP3Stream->cs_factor[i][k + 1] * group9_table[n][0];
s[k + 64] = pMP3Stream->cs_factor[i][k] * group9_table[n][1];
s[k + 64 + 1] = pMP3Stream->cs_factor[i][k + 1] * group9_table[n][1];
s[k + 128] = pMP3Stream->cs_factor[i][k] * group9_table[n][2];
s[k + 128 + 1] = pMP3Stream->cs_factor[i][k + 1] * group9_table[n][2];
k++; /* skip right chan dispatch */
goto dispatch;
case 18 + 5:
UNPACKJ_N(4) /* 15 levels */
case 18 + 6:
UNPACKJ_N(5) /* 31 levels */
case 18 + 7:
UNPACKJ_N(6) /* 63 levels */
case 18 + 8:
UNPACKJ_N(7) /* 127 levels */
case 18 + 9:
UNPACKJ_N(8) /* 255 levels */
case 18 + 10:
UNPACKJ_N(9) /* 511 levels */
case 18 + 11:
UNPACKJ_N(10) /* 1023 levels */
case 18 + 12:
UNPACKJ_N(11) /* 2047 levels */
case 18 + 13:
UNPACKJ_N(12) /* 4095 levels */
case 18 + 14:
UNPACKJ_N(13) /* 8191 levels */
case 18 + 15:
UNPACKJ_N(14) /* 16383 levels */
case 18 + 16:
UNPACKJ_N(15) /* 32767 levels */
case 18 + 17:
UNPACKJ_N(16) /* 65535 levels */
/* -- end of dispatch -- */
case 37:
skip(pMP3Stream->bit_skip);
case 36:
s += 3 * 64;
} /* end switch */
} /* end j loop */
} /* end i loop */
}
/*-------------------------------------------------------------------------*/
unsigned char *gpNextByteAfterData = NULL;
IN_OUT audio_decode(unsigned char *bs, signed short *pcm, unsigned char *pNextByteAfterData)
{
gpNextByteAfterData = pNextByteAfterData; // sigh....
return audio_decode_routine(bs, pcm);
}
/*-------------------------------------------------------------------------*/
IN_OUT L2audio_decode(unsigned char *bs, signed short *pcm)
{
int sync, prot;
IN_OUT in_out;
load_init(bs); /* initialize bit getter */
/* test sync */
in_out.in_bytes = 0; /* assume fail */
in_out.out_bytes = 0;
sync = load(12);
if (sync != 0xFFF)
return in_out; /* sync fail */
load(3); /* skip id and option (checked by init) */
prot = load(1); /* load prot bit */
load(6); /* skip to pad */
pMP3Stream->pad = load(1);
load(1); /* skip to mode */
pMP3Stream->stereo_sb = look_joint[load(4)];
if (prot)
load(4); /* skip to data */
else
load(20); /* skip crc */
unpack_ba(); /* unpack bit allocation */
unpack_sfs(); /* unpack scale factor selectors */
unpack_sf(); /* unpack scale factor */
unpack_samp(); /* unpack samples */
pMP3Stream->sbt(sample, pcm, 36);
/*-----------*/
in_out.in_bytes = pMP3Stream->framebytes + pMP3Stream->pad;
in_out.out_bytes = pMP3Stream->outbytes;
return in_out;
}
/*-------------------------------------------------------------------------*/
#define COMPILE_ME
#include "cupini.c" /* initialization */
#include "cupl1.c" /* Layer I */
/*-------------------------------------------------------------------------*/
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