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jedi-academy/codemp/mp3code/cupl1.c

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#pragma warning(disable:4206) // nonstandard extension used : translation unit is empty
#pragma warning(disable:4711) // function 'xxxx' selected for automatic inline expansion
#ifdef COMPILE_ME
/*____________________________________________________________________________
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: cupL1.c,v 1.3 1999/10/19 07:13:08 elrod Exp $
____________________________________________________________________________*/
/**** cupL1.c ***************************************************
MPEG audio decoder Layer I mpeg1 and mpeg2
include to clup.c
******************************************************************/
/*======================================================================*/
static const int bat_bit_masterL1[] =
{
0, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16
};
////@@@@static float *pMP3Stream->cs_factorL1 = &pMP3Stream->cs_factor[0]; // !!!!!!!!!!!!!!!!
static float look_c_valueL1[16]; // effectively constant
////@@@@static int nbatL1 = 32;
/*======================================================================*/
static void unpack_baL1()
{
int j;
int nstereo;
pMP3Stream->bit_skip = 0;
nstereo = pMP3Stream->stereo_sb;
for (j = 0; j < pMP3Stream->nbatL1; j++)
{
mac_load_check(4);
ballo[j] = samp_dispatch[j] = mac_load(4);
if (j >= pMP3Stream->nsb_limit)
pMP3Stream->bit_skip += bat_bit_masterL1[samp_dispatch[j]];
c_value[j] = look_c_valueL1[samp_dispatch[j]];
if (--nstereo < 0)
{
ballo[j + 1] = ballo[j];
samp_dispatch[j] += 15; /* flag as joint */
samp_dispatch[j + 1] = samp_dispatch[j]; /* flag for sf */
c_value[j + 1] = c_value[j];
j++;
}
}
/*-- terminate with bit skip and end --*/
samp_dispatch[pMP3Stream->nsb_limit] = 31;
samp_dispatch[j] = 30;
}
/*-------------------------------------------------------------------------*/
static void unpack_sfL1(void) /* unpack scale factor */
{ /* combine dequant and scale factors */
int i;
for (i = 0; i < pMP3Stream->nbatL1; i++)
{
if (ballo[i])
{
mac_load_check(6);
pMP3Stream->cs_factorL1[i] = c_value[i] * sf_table[mac_load(6)];
}
}
/*-- done --*/
}
/*-------------------------------------------------------------------------*/
#define UNPACKL1_N(n) s[k] = pMP3Stream->cs_factorL1[k]*(load(n)-((1 << (n-1)) -1)); \
goto dispatch;
#define UNPACKL1J_N(n) tmp = (load(n)-((1 << (n-1)) -1)); \
s[k] = pMP3Stream->cs_factorL1[k]*tmp; \
s[k+1] = pMP3Stream->cs_factorL1[k+1]*tmp; \
k++; \
goto dispatch;
/*-------------------------------------------------------------------------*/
static void unpack_sampL1() /* unpack samples */
{
int j, k;
float *s;
long tmp;
s = sample;
for (j = 0; j < 12; j++)
{
k = -1;
dispatch:switch (samp_dispatch[++k])
{
case 0:
s[k] = 0.0F;
goto dispatch;
case 1:
UNPACKL1_N(2) /* 3 levels */
case 2:
UNPACKL1_N(3) /* 7 levels */
case 3:
UNPACKL1_N(4) /* 15 levels */
case 4:
UNPACKL1_N(5) /* 31 levels */
case 5:
UNPACKL1_N(6) /* 63 levels */
case 6:
UNPACKL1_N(7) /* 127 levels */
case 7:
UNPACKL1_N(8) /* 255 levels */
case 8:
UNPACKL1_N(9) /* 511 levels */
case 9:
UNPACKL1_N(10) /* 1023 levels */
case 10:
UNPACKL1_N(11) /* 2047 levels */
case 11:
UNPACKL1_N(12) /* 4095 levels */
case 12:
UNPACKL1_N(13) /* 8191 levels */
case 13:
UNPACKL1_N(14) /* 16383 levels */
case 14:
UNPACKL1_N(15) /* 32767 levels */
/* -- joint ---- */
case 15 + 0:
s[k + 1] = s[k] = 0.0F;
k++; /* skip right chan dispatch */
goto dispatch;
/* -- joint ---- */
case 15 + 1:
UNPACKL1J_N(2) /* 3 levels */
case 15 + 2:
UNPACKL1J_N(3) /* 7 levels */
case 15 + 3:
UNPACKL1J_N(4) /* 15 levels */
case 15 + 4:
UNPACKL1J_N(5) /* 31 levels */
case 15 + 5:
UNPACKL1J_N(6) /* 63 levels */
case 15 + 6:
UNPACKL1J_N(7) /* 127 levels */
case 15 + 7:
UNPACKL1J_N(8) /* 255 levels */
case 15 + 8:
UNPACKL1J_N(9) /* 511 levels */
case 15 + 9:
UNPACKL1J_N(10) /* 1023 levels */
case 15 + 10:
UNPACKL1J_N(11) /* 2047 levels */
case 15 + 11:
UNPACKL1J_N(12) /* 4095 levels */
case 15 + 12:
UNPACKL1J_N(13) /* 8191 levels */
case 15 + 13:
UNPACKL1J_N(14) /* 16383 levels */
case 15 + 14:
UNPACKL1J_N(15) /* 32767 levels */
/* -- end of dispatch -- */
case 31:
skip(pMP3Stream->bit_skip);
case 30:
s += 64;
} /* end switch */
} /* end j loop */
/*-- done --*/
}
/*-------------------------------------------------------------------*/
IN_OUT L1audio_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)) << 2;
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_baL1(); /* unpack bit allocation */
unpack_sfL1(); /* unpack scale factor */
unpack_sampL1(); /* unpack samples */
pMP3Stream->sbt(sample, pcm, 12);
/*-----------*/
in_out.in_bytes = pMP3Stream->framebytes + pMP3Stream->pad;
in_out.out_bytes = pMP3Stream->outbytes;
return in_out;
}
/*-------------------------------------------------------------------------*/
int L1audio_decode_init(MPEG_HEAD * h, int framebytes_arg,
int reduction_code, int transform_code, int convert_code,
int freq_limit)
{
int i, k;
static int first_pass = 1;
long samprate;
int limit;
long step;
int bit_code;
/*--- sf init done by layer II init ---*/
if (first_pass)
{
for (step = 4, i = 1; i < 16; i++, step <<= 1)
look_c_valueL1[i] = (float) (2.0 / (step - 1));
first_pass = 0;
}
pMP3Stream->cs_factorL1 = pMP3Stream->cs_factor[0];
transform_code = transform_code; /* not used, asm compatability */
bit_code = 0;
if (convert_code & 8)
bit_code = 1;
convert_code = convert_code & 3; /* higher bits used by dec8 freq cvt */
if (reduction_code < 0)
reduction_code = 0;
if (reduction_code > 2)
reduction_code = 2;
if (freq_limit < 1000)
freq_limit = 1000;
pMP3Stream->framebytes = framebytes_arg;
/* check if code handles */
if (h->option != 3)
return 0; /* layer I only */
pMP3Stream->nbatL1 = 32;
pMP3Stream->max_sb = pMP3Stream->nbatL1;
/*----- compute pMP3Stream->nsb_limit --------*/
samprate = sr_table[4 * h->id + h->sr_index];
pMP3Stream->nsb_limit = (freq_limit * 64L + samprate / 2) / samprate;
/*- caller limit -*/
/*---- limit = 0.94*(32>>reduction_code); ----*/
limit = (32 >> reduction_code);
if (limit > 8)
limit--;
if (pMP3Stream->nsb_limit > limit)
pMP3Stream->nsb_limit = limit;
if (pMP3Stream->nsb_limit > pMP3Stream->max_sb)
pMP3Stream->nsb_limit = pMP3Stream->max_sb;
pMP3Stream->outvalues = 384 >> reduction_code;
if (h->mode != 3)
{ /* adjust for 2 channel modes */
pMP3Stream->nbatL1 *= 2;
pMP3Stream->max_sb *= 2;
pMP3Stream->nsb_limit *= 2;
}
/* set sbt function */
k = 1 + convert_code;
if (h->mode == 3)
{
k = 0;
}
pMP3Stream->sbt = sbt_table[bit_code][reduction_code][k];
pMP3Stream->outvalues *= out_chans[k];
if (bit_code)
pMP3Stream->outbytes = pMP3Stream->outvalues;
else
pMP3Stream->outbytes = sizeof(short) * pMP3Stream->outvalues;
decinfo.channels = out_chans[k];
decinfo.outvalues = pMP3Stream->outvalues;
decinfo.samprate = samprate >> reduction_code;
if (bit_code)
decinfo.bits = 8;
else
decinfo.bits = sizeof(short) * 8;
decinfo.framebytes = pMP3Stream->framebytes;
decinfo.type = 0;
/* clear sample buffer, unused sub bands must be 0 */
for (i = 0; i < 768; i++)
sample[i] = 0.0F;
/* init sub-band transform */
sbt_init();
return 1;
}
/*---------------------------------------------------------*/
#endif // #ifdef COMPILE_ME
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