Version 0.1.1 from FTP

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.0
Mark Spencer 26 years ago
parent f1192f7be0
commit b3b040de04

@ -0,0 +1,500 @@
/*____________________________________________________________________________
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 ***************************************************
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 *m, 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 *m input, mpeg structure for multiple streams
MPEG_HEAD *h input, mpeg header info (returned by call to head_info)
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( MPEG *m, DEC_INFO *info)
MPEG *m input, mpeg structure for multiple streams
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 "L3.h"
#include "mhead.h" /* mpeg header structure */
#ifdef _MSC_VER
#pragma warning(disable: 4709)
#endif
/*-------------------------------------------------------
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)
---------------------------------------------------------*/
/*====================================================================*/
/*----------------*/
/* Read Only */
static 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 */
}; /* Okay to be global */
/* Read Only */
static int bat_bit_master[] =
{
0, 5, 7, 9, 10, 12, 15, 18, 21, 24, 27, 30, 33, 36, 39, 42, 45, 48};
void sbt_mono(MPEG *m, float *sample, short *pcm, int n);
void sbt_dual(MPEG *m, float *sample, short *pcm, int n);
IN_OUT L2audio_decode(void *mv, unsigned char *bs, signed short *pcm);
/*======================================================================*/
/*======================================================================*/
/*------------- initialize bit getter -------------*/
static void load_init(MPEG *m, unsigned char *buf)
{
m->cup.bs_ptr = buf;
m->cup.bits = 0;
m->cup.bitbuf = 0;
}
/*------------- get n bits from bitstream -------------*/
static long load(MPEG *m, int n)
{
unsigned long x;
if (m->cup.bits < n)
{ /* refill bit buf if necessary */
while (m->cup.bits <= 24)
{
m->cup.bitbuf = (m->cup.bitbuf << 8) | *m->cup.bs_ptr++;
m->cup.bits += 8;
}
}
m->cup.bits -= n;
x = m->cup.bitbuf >> m->cup.bits;
m->cup.bitbuf -= x << m->cup.bits;
return x;
}
/*------------- skip over n bits in bitstream -------------*/
static void skip(MPEG *m, int n)
{
int k;
if (m->cup.bits < n)
{
n -= m->cup.bits;
k = n >> 3;
/*--- bytes = n/8 --*/
m->cup.bs_ptr += k;
n -= k << 3;
m->cup.bitbuf = *m->cup.bs_ptr++;
m->cup.bits = 8;
}
m->cup.bits -= n;
m->cup.bitbuf -= (m->cup.bitbuf >> m->cup.bits) << m->cup.bits;
}
/*--------------------------------------------------------------*/
#define mac_load_check(n) if( m->cup.bits < (n) ) { \
while( m->cup.bits <= 24 ) { \
m->cup.bitbuf = (m->cup.bitbuf << 8) | *m->cup.bs_ptr++; \
m->cup.bits += 8; \
} \
}
/*--------------------------------------------------------------*/
#define mac_load(n) ( m->cup.bits -= n, \
m->cup.bitval = m->cup.bitbuf >> m->cup.bits, \
m->cup.bitbuf -= m->cup.bitval << m->cup.bits, \
m->cup.bitval )
/*======================================================================*/
static void unpack_ba(MPEG *m)
{
int i, j, k;
/* Read Only */
static int nbit[4] =
{4, 4, 3, 2};
int nstereo;
m->cup.bit_skip = 0;
nstereo = m->cup.stereo_sb;
k = 0;
for (i = 0; i < 4; i++)
{
for (j = 0; j < m->cup.nbat[i]; j++, k++)
{
mac_load_check(4);
m->cup.ballo[k] = m->cup.samp_dispatch[k] = m->cup.bat[i][mac_load(nbit[i])];
if (k >= m->cup.nsb_limit)
m->cup.bit_skip += bat_bit_master[m->cup.samp_dispatch[k]];
m->cup.c_value[k] = m->cup.look_c_value[m->cup.samp_dispatch[k]];
if (--nstereo < 0)
{
m->cup.ballo[k + 1] = m->cup.ballo[k];
m->cup.samp_dispatch[k] += 18; /* flag as joint */
m->cup.samp_dispatch[k + 1] = m->cup.samp_dispatch[k]; /* flag for sf */
m->cup.c_value[k + 1] = m->cup.c_value[k];
k++;
j++;
}
}
}
m->cup.samp_dispatch[m->cup.nsb_limit] = 37; /* terminate the dispatcher with skip */
m->cup.samp_dispatch[k] = 36; /* terminate the dispatcher */
}
/*-------------------------------------------------------------------------*/
static void unpack_sfs(MPEG *m) /* unpack scale factor selectors */
{
int i;
for (i = 0; i < m->cup.max_sb; i++)
{
mac_load_check(2);
if (m->cup.ballo[i])
m->cup.sf_dispatch[i] = mac_load(2);
else
m->cup.sf_dispatch[i] = 4; /* no allo */
}
m->cup.sf_dispatch[i] = 5; /* terminate dispatcher */
}
/*-------------------------------------------------------------------------*/
static void unpack_sf(MPEG *m) /* unpack scale factor */
{ /* combine dequant and scale factors */
int i;
i = -1;
dispatch:switch (m->cup.sf_dispatch[++i])
{
case 0: /* 3 factors 012 */
mac_load_check(18);
m->cup.cs_factor[0][i] = m->cup.c_value[i] * m->cup.sf_table[mac_load(6)];
m->cup.cs_factor[1][i] = m->cup.c_value[i] * m->cup.sf_table[mac_load(6)];
m->cup.cs_factor[2][i] = m->cup.c_value[i] * m->cup.sf_table[mac_load(6)];
goto dispatch;
case 1: /* 2 factors 002 */
mac_load_check(12);
m->cup.cs_factor[1][i] = m->cup.cs_factor[0][i] = m->cup.c_value[i] * m->cup.sf_table[mac_load(6)];
m->cup.cs_factor[2][i] = m->cup.c_value[i] * m->cup.sf_table[mac_load(6)];
goto dispatch;
case 2: /* 1 factor 000 */
mac_load_check(6);
m->cup.cs_factor[2][i] = m->cup.cs_factor[1][i] = m->cup.cs_factor[0][i] =
m->cup.c_value[i] * m->cup.sf_table[mac_load(6)];
goto dispatch;
case 3: /* 2 factors 022 */
mac_load_check(12);
m->cup.cs_factor[0][i] = m->cup.c_value[i] * m->cup.sf_table[mac_load(6)];
m->cup.cs_factor[2][i] = m->cup.cs_factor[1][i] = m->cup.c_value[i] * m->cup.sf_table[mac_load(6)];
goto dispatch;
case 4: /* no allo */
/*-- m->cup.cs_factor[2][i] = m->cup.cs_factor[1][i] = m->cup.cs_factor[0][i] = 0.0; --*/
goto dispatch;
case 5: /* all done */
;
} /* end switch */
}
/*-------------------------------------------------------------------------*/
#define UNPACK_N(n) s[k] = m->cup.cs_factor[i][k]*(load(m,n)-((1 << (n-1)) -1)); \
s[k+64] = m->cup.cs_factor[i][k]*(load(m,n)-((1 << (n-1)) -1)); \
s[k+128] = m->cup.cs_factor[i][k]*(load(m,n)-((1 << (n-1)) -1)); \
goto dispatch;
#define UNPACK_N2(n) mac_load_check(3*n); \
s[k] = m->cup.cs_factor[i][k]*(mac_load(n)-((1 << (n-1)) -1)); \
s[k+64] = m->cup.cs_factor[i][k]*(mac_load(n)-((1 << (n-1)) -1)); \
s[k+128] = m->cup.cs_factor[i][k]*(mac_load(n)-((1 << (n-1)) -1)); \
goto dispatch;
#define UNPACK_N3(n) mac_load_check(2*n); \
s[k] = m->cup.cs_factor[i][k]*(mac_load(n)-((1 << (n-1)) -1)); \
s[k+64] = m->cup.cs_factor[i][k]*(mac_load(n)-((1 << (n-1)) -1)); \
mac_load_check(n); \
s[k+128] = m->cup.cs_factor[i][k]*(mac_load(n)-((1 << (n-1)) -1)); \
goto dispatch;
#define UNPACKJ_N(n) tmp = (load(m,n)-((1 << (n-1)) -1)); \
s[k] = m->cup.cs_factor[i][k]*tmp; \
s[k+1] = m->cup.cs_factor[i][k+1]*tmp; \
tmp = (load(m,n)-((1 << (n-1)) -1)); \
s[k+64] = m->cup.cs_factor[i][k]*tmp; \
s[k+64+1] = m->cup.cs_factor[i][k+1]*tmp; \
tmp = (load(m,n)-((1 << (n-1)) -1)); \
s[k+128] = m->cup.cs_factor[i][k]*tmp; \
s[k+128+1] = m->cup.cs_factor[i][k+1]*tmp; \
k++; /* skip right chan dispatch */ \
goto dispatch;
/*-------------------------------------------------------------------------*/
static void unpack_samp(MPEG *m) /* unpack samples */
{
int i, j, k;
float *s;
int n;
long tmp;
s = m->cup.sample;
for (i = 0; i < 3; i++)
{ /* 3 groups of scale factors */
for (j = 0; j < 4; j++)
{
k = -1;
dispatch:switch (m->cup.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] = m->cup.cs_factor[i][k] * m->cup.group3_table[n][0];
s[k + 64] = m->cup.cs_factor[i][k] * m->cup.group3_table[n][1];
s[k + 128] = m->cup.cs_factor[i][k] * m->cup.group3_table[n][2];
goto dispatch;
case 2: /* 5 levels grouped 7 bits */
mac_load_check(7);
n = mac_load(7);
s[k] = m->cup.cs_factor[i][k] * m->cup.group5_table[n][0];
s[k + 64] = m->cup.cs_factor[i][k] * m->cup.group5_table[n][1];
s[k + 128] = m->cup.cs_factor[i][k] * m->cup.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] = m->cup.cs_factor[i][k] * m->cup.group9_table[n][0];
s[k + 64] = m->cup.cs_factor[i][k] * m->cup.group9_table[n][1];
s[k + 128] = m->cup.cs_factor[i][k] * m->cup.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(m,5);
s[k] = m->cup.cs_factor[i][k] * m->cup.group3_table[n][0];
s[k + 1] = m->cup.cs_factor[i][k + 1] * m->cup.group3_table[n][0];
s[k + 64] = m->cup.cs_factor[i][k] * m->cup.group3_table[n][1];
s[k + 64 + 1] = m->cup.cs_factor[i][k + 1] * m->cup.group3_table[n][1];
s[k + 128] = m->cup.cs_factor[i][k] * m->cup.group3_table[n][2];
s[k + 128 + 1] = m->cup.cs_factor[i][k + 1] * m->cup.group3_table[n][2];
k++; /* skip right chan dispatch */
goto dispatch;
case 18 + 2: /* 5 levels grouped 7 bits */
n = load(m,7);
s[k] = m->cup.cs_factor[i][k] * m->cup.group5_table[n][0];
s[k + 1] = m->cup.cs_factor[i][k + 1] * m->cup.group5_table[n][0];
s[k + 64] = m->cup.cs_factor[i][k] * m->cup.group5_table[n][1];
s[k + 64 + 1] = m->cup.cs_factor[i][k + 1] * m->cup.group5_table[n][1];
s[k + 128] = m->cup.cs_factor[i][k] * m->cup.group5_table[n][2];
s[k + 128 + 1] = m->cup.cs_factor[i][k + 1] * m->cup.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(m,10);
s[k] = m->cup.cs_factor[i][k] * m->cup.group9_table[n][0];
s[k + 1] = m->cup.cs_factor[i][k + 1] * m->cup.group9_table[n][0];
s[k + 64] = m->cup.cs_factor[i][k] * m->cup.group9_table[n][1];
s[k + 64 + 1] = m->cup.cs_factor[i][k + 1] * m->cup.group9_table[n][1];
s[k + 128] = m->cup.cs_factor[i][k] * m->cup.group9_table[n][2];
s[k + 128 + 1] = m->cup.cs_factor[i][k + 1] * m->cup.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(m, m->cup.bit_skip);
case 36:
s += 3 * 64;
} /* end switch */
} /* end j loop */
} /* end i loop */
}
/*-------------------------------------------------------------------------*/
IN_OUT audio_decode(MPEG *m, unsigned char *bs, signed short *pcm)
{
return m->cup.audio_decode_routine(m, bs, pcm);
}
/*-------------------------------------------------------------------------*/
IN_OUT L2audio_decode(void *mv, unsigned char *bs, signed short *pcm)
{
MPEG *m = mv;
int sync, prot;
IN_OUT in_out;
load_init(m, bs); /* initialize bit getter */
/* test sync */
in_out.in_bytes = 0; /* assume fail */
in_out.out_bytes = 0;
sync = load(m,12);
if (sync != 0xFFF)
return in_out; /* sync fail */
load(m,3); /* skip id and option (checked by init) */
prot = load(m,1); /* load prot bit */
load(m,6); /* skip to pad */
m->cup.pad = load(m,1);
load(m,1); /* skip to mode */
m->cup.stereo_sb = look_joint[load(m,4)];
if (prot)
load(m,4); /* skip to data */
else
load(m,20); /* skip crc */
unpack_ba(m); /* unpack bit allocation */
unpack_sfs(m); /* unpack scale factor selectors */
unpack_sf(m); /* unpack scale factor */
unpack_samp(m); /* unpack samples */
m->cup.sbt(m, m->cup.sample, pcm, 36);
/*-----------*/
in_out.in_bytes = m->cup.framebytes + m->cup.pad;
in_out.out_bytes = m->cup.outbytes;
return in_out;
}
/*-------------------------------------------------------------------------*/
#include "cupini.c" /* initialization */
#include "cupL1.c" /* Layer I */
/*-------------------------------------------------------------------------*/

@ -0,0 +1,319 @@
/*____________________________________________________________________________
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 ***************************************************
MPEG audio decoder Layer I mpeg1 and mpeg2
include to clup.c
******************************************************************/
/*======================================================================*/
/* Read Only */
static int bat_bit_masterL1[] =
{
0, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16};
/*======================================================================*/
static void unpack_baL1(MPEG *m)
{
int j;
int nstereo;
m->cup.bit_skip = 0;
nstereo = m->cup.stereo_sb;
for (j = 0; j < m->cup.nbatL1; j++)
{
mac_load_check(4);
m->cup.ballo[j] = m->cup.samp_dispatch[j] = mac_load(4);
if (j >= m->cup.nsb_limit)
m->cup.bit_skip += bat_bit_masterL1[m->cup.samp_dispatch[j]];
m->cup.c_value[j] = m->cup.look_c_valueL1[m->cup.samp_dispatch[j]];
if (--nstereo < 0)
{
m->cup.ballo[j + 1] = m->cup.ballo[j];
m->cup.samp_dispatch[j] += 15; /* flag as joint */
m->cup.samp_dispatch[j + 1] = m->cup.samp_dispatch[j]; /* flag for sf */
m->cup.c_value[j + 1] = m->cup.c_value[j];
j++;
}
}
/*-- terminate with bit skip and end --*/
m->cup.samp_dispatch[m->cup.nsb_limit] = 31;
m->cup.samp_dispatch[j] = 30;
}
/*-------------------------------------------------------------------------*/
static void unpack_sfL1(MPEG *m) /* unpack scale factor */
{ /* combine dequant and scale factors */
int i;
for (i = 0; i < m->cup.nbatL1; i++)
{
if (m->cup.ballo[i])
{
mac_load_check(6);
m->cup.cs_factorL1[i] = m->cup.c_value[i] * m->cup.sf_table[mac_load(6)];
}
}
/*-- done --*/
}
/*-------------------------------------------------------------------------*/
#define UNPACKL1_N(n) s[k] = m->cup.cs_factorL1[k]*(load(m,n)-((1 << (n-1)) -1)); \
goto dispatch;
#define UNPACKL1J_N(n) tmp = (load(m,n)-((1 << (n-1)) -1)); \
s[k] = m->cup.cs_factorL1[k]*tmp; \
s[k+1] = m->cup.cs_factorL1[k+1]*tmp; \
k++; \
goto dispatch;
/*-------------------------------------------------------------------------*/
static void unpack_sampL1(MPEG *m) /* unpack samples */
{
int j, k;
float *s;
long tmp;
s = m->cup.sample;
for (j = 0; j < 12; j++)
{
k = -1;
dispatch:switch (m->cup.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(m,m->cup.bit_skip);
case 30:
s += 64;
} /* end switch */
} /* end j loop */
/*-- done --*/
}
/*-------------------------------------------------------------------*/
IN_OUT L1audio_decode(void *mv, unsigned char *bs, signed short *pcm)
{
MPEG *m = mv;
int sync, prot;
IN_OUT in_out;
load_init(m, bs); /* initialize bit getter */
/* test sync */
in_out.in_bytes = 0; /* assume fail */
in_out.out_bytes = 0;
sync = load(m,12);
if (sync != 0xFFF)
return in_out; /* sync fail */
load(m,3); /* skip id and option (checked by init) */
prot = load(m,1); /* load prot bit */
load(m,6); /* skip to pad */
m->cup.pad = (load(m,1)) << 2;
load(m,1); /* skip to mode */
m->cup.stereo_sb = look_joint[load(m,4)];
if (prot)
load(m,4); /* skip to data */
else
load(m,20); /* skip crc */
unpack_baL1(m); /* unpack bit allocation */
unpack_sfL1(m); /* unpack scale factor */
unpack_sampL1(m); /* unpack samples */
m->cup.sbt(m, m->cup.sample, pcm, 12);
/*-----------*/
in_out.in_bytes = m->cup.framebytes + m->cup.pad;
in_out.out_bytes = m->cup.outbytes;
return in_out;
}
/*-------------------------------------------------------------------------*/
int L1audio_decode_init(MPEG *m, MPEG_HEAD * h, int framebytes_arg,
int reduction_code, int transform_code, int convert_code,
int freq_limit)
{
int i, k;
long samprate;
int limit;
long step;
int bit_code;
/*--- sf init done by layer II init ---*/
if (m->cup.first_pass_L1)
{
for (step = 4, i = 1; i < 16; i++, step <<= 1)
m->cup.look_c_valueL1[i] = (float) (2.0 / (step - 1));
m->cup.first_pass_L1 = 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;
m->cup.framebytes = framebytes_arg;
/* check if code handles */
if (h->option != 3)
return 0; /* layer I only */
m->cup.nbatL1 = 32;
m->cup.max_sb = m->cup.nbatL1;
/*----- compute nsb_limit --------*/
samprate = sr_table[4 * h->id + h->sr_index];
m->cup.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 (m->cup.nsb_limit > limit)
m->cup.nsb_limit = limit;
if (m->cup.nsb_limit > m->cup.max_sb)
m->cup.nsb_limit = m->cup.max_sb;
m->cup.outvalues = 384 >> reduction_code;
if (h->mode != 3)
{ /* adjust for 2 channel modes */
m->cup.nbatL1 *= 2;
m->cup.max_sb *= 2;
m->cup.nsb_limit *= 2;
}
/* set sbt function */
k = 1 + convert_code;
if (h->mode == 3)
{
k = 0;
}
m->cup.sbt = sbt_table[bit_code][reduction_code][k];
m->cup.outvalues *= out_chans[k];
if (bit_code)
m->cup.outbytes = m->cup.outvalues;
else
m->cup.outbytes = sizeof(short) * m->cup.outvalues;
m->cup.decinfo.channels = out_chans[k];
m->cup.decinfo.outvalues = m->cup.outvalues;
m->cup.decinfo.samprate = samprate >> reduction_code;
if (bit_code)
m->cup.decinfo.bits = 8;
else
m->cup.decinfo.bits = sizeof(short) * 8;
m->cup.decinfo.framebytes = m->cup.framebytes;
m->cup.decinfo.type = 0;
/* clear sample buffer, unused sub bands must be 0 */
for (i = 0; i < 768; i++)
m->cup.sample[i] = 0.0F;
/* init sub-band transform */
sbt_init();
return 1;
}
/*---------------------------------------------------------*/
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