326 lines
9.5 KiB
C
326 lines
9.5 KiB
C
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#pragma warning(disable:4206) // nonstandard extension used : translation unit is empty
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#pragma warning(disable:4711) // function 'xxxx' selected for automatic inline expansion
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#ifdef COMPILE_ME
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/*____________________________________________________________________________
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FreeAmp - The Free MP3 Player
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MP3 Decoder originally Copyright (C) 1995-1997 Xing Technology
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Corp. http://www.xingtech.com
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Portions Copyright (C) 1998-1999 EMusic.com
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This program is free software; you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation; either version 2 of the License, or
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(at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with this program; if not, write to the Free Software
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Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
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$Id: cupL1.c,v 1.3 1999/10/19 07:13:08 elrod Exp $
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____________________________________________________________________________*/
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/**** cupL1.c ***************************************************
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MPEG audio decoder Layer I mpeg1 and mpeg2
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include to clup.c
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******************************************************************/
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/*======================================================================*/
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static const int bat_bit_masterL1[] =
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{
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0, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16
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};
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////@@@@static float *pMP3Stream->cs_factorL1 = &pMP3Stream->cs_factor[0]; // !!!!!!!!!!!!!!!!
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static float look_c_valueL1[16]; // effectively constant
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////@@@@static int nbatL1 = 32;
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/*======================================================================*/
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static void unpack_baL1()
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{
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int j;
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int nstereo;
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pMP3Stream->bit_skip = 0;
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nstereo = pMP3Stream->stereo_sb;
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for (j = 0; j < pMP3Stream->nbatL1; j++)
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{
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mac_load_check(4);
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ballo[j] = samp_dispatch[j] = mac_load(4);
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if (j >= pMP3Stream->nsb_limit)
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pMP3Stream->bit_skip += bat_bit_masterL1[samp_dispatch[j]];
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c_value[j] = look_c_valueL1[samp_dispatch[j]];
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if (--nstereo < 0)
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{
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ballo[j + 1] = ballo[j];
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samp_dispatch[j] += 15; /* flag as joint */
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samp_dispatch[j + 1] = samp_dispatch[j]; /* flag for sf */
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c_value[j + 1] = c_value[j];
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j++;
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}
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}
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/*-- terminate with bit skip and end --*/
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samp_dispatch[pMP3Stream->nsb_limit] = 31;
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samp_dispatch[j] = 30;
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}
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/*-------------------------------------------------------------------------*/
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static void unpack_sfL1(void) /* unpack scale factor */
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{ /* combine dequant and scale factors */
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int i;
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for (i = 0; i < pMP3Stream->nbatL1; i++)
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{
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if (ballo[i])
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{
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mac_load_check(6);
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pMP3Stream->cs_factorL1[i] = c_value[i] * sf_table[mac_load(6)];
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}
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}
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/*-- done --*/
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}
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/*-------------------------------------------------------------------------*/
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#define UNPACKL1_N(n) s[k] = pMP3Stream->cs_factorL1[k]*(load(n)-((1 << (n-1)) -1)); \
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goto dispatch;
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#define UNPACKL1J_N(n) tmp = (load(n)-((1 << (n-1)) -1)); \
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s[k] = pMP3Stream->cs_factorL1[k]*tmp; \
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s[k+1] = pMP3Stream->cs_factorL1[k+1]*tmp; \
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k++; \
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goto dispatch;
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/*-------------------------------------------------------------------------*/
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static void unpack_sampL1() /* unpack samples */
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{
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int j, k;
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float *s;
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long tmp;
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s = sample;
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for (j = 0; j < 12; j++)
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{
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k = -1;
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dispatch:switch (samp_dispatch[++k])
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{
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case 0:
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s[k] = 0.0F;
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goto dispatch;
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case 1:
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UNPACKL1_N(2) /* 3 levels */
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case 2:
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UNPACKL1_N(3) /* 7 levels */
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case 3:
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UNPACKL1_N(4) /* 15 levels */
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case 4:
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UNPACKL1_N(5) /* 31 levels */
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case 5:
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UNPACKL1_N(6) /* 63 levels */
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case 6:
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UNPACKL1_N(7) /* 127 levels */
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case 7:
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UNPACKL1_N(8) /* 255 levels */
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case 8:
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UNPACKL1_N(9) /* 511 levels */
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case 9:
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UNPACKL1_N(10) /* 1023 levels */
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case 10:
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UNPACKL1_N(11) /* 2047 levels */
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case 11:
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UNPACKL1_N(12) /* 4095 levels */
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case 12:
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UNPACKL1_N(13) /* 8191 levels */
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case 13:
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UNPACKL1_N(14) /* 16383 levels */
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case 14:
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UNPACKL1_N(15) /* 32767 levels */
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/* -- joint ---- */
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case 15 + 0:
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s[k + 1] = s[k] = 0.0F;
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k++; /* skip right chan dispatch */
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goto dispatch;
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/* -- joint ---- */
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case 15 + 1:
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UNPACKL1J_N(2) /* 3 levels */
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case 15 + 2:
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UNPACKL1J_N(3) /* 7 levels */
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case 15 + 3:
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UNPACKL1J_N(4) /* 15 levels */
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case 15 + 4:
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UNPACKL1J_N(5) /* 31 levels */
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case 15 + 5:
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UNPACKL1J_N(6) /* 63 levels */
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case 15 + 6:
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UNPACKL1J_N(7) /* 127 levels */
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case 15 + 7:
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UNPACKL1J_N(8) /* 255 levels */
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case 15 + 8:
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UNPACKL1J_N(9) /* 511 levels */
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case 15 + 9:
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UNPACKL1J_N(10) /* 1023 levels */
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case 15 + 10:
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UNPACKL1J_N(11) /* 2047 levels */
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case 15 + 11:
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UNPACKL1J_N(12) /* 4095 levels */
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case 15 + 12:
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UNPACKL1J_N(13) /* 8191 levels */
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case 15 + 13:
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UNPACKL1J_N(14) /* 16383 levels */
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case 15 + 14:
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UNPACKL1J_N(15) /* 32767 levels */
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/* -- end of dispatch -- */
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case 31:
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skip(pMP3Stream->bit_skip);
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case 30:
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s += 64;
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} /* end switch */
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} /* end j loop */
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/*-- done --*/
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}
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/*-------------------------------------------------------------------*/
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IN_OUT L1audio_decode(unsigned char *bs, signed short *pcm)
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{
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int sync, prot;
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IN_OUT in_out;
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load_init(bs); /* initialize bit getter */
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/* test sync */
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in_out.in_bytes = 0; /* assume fail */
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in_out.out_bytes = 0;
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sync = load(12);
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if (sync != 0xFFF)
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return in_out; /* sync fail */
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load(3); /* skip id and option (checked by init) */
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prot = load(1); /* load prot bit */
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load(6); /* skip to pad */
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pMP3Stream->pad = (load(1)) << 2;
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load(1); /* skip to mode */
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pMP3Stream->stereo_sb = look_joint[load(4)];
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if (prot)
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load(4); /* skip to data */
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else
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load(20); /* skip crc */
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unpack_baL1(); /* unpack bit allocation */
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unpack_sfL1(); /* unpack scale factor */
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unpack_sampL1(); /* unpack samples */
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pMP3Stream->sbt(sample, pcm, 12);
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/*-----------*/
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in_out.in_bytes = pMP3Stream->framebytes + pMP3Stream->pad;
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in_out.out_bytes = pMP3Stream->outbytes;
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return in_out;
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}
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/*-------------------------------------------------------------------------*/
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int L1audio_decode_init(MPEG_HEAD * h, int framebytes_arg,
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int reduction_code, int transform_code, int convert_code,
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int freq_limit)
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{
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int i, k;
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static int first_pass = 1;
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long samprate;
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int limit;
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long step;
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int bit_code;
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/*--- sf init done by layer II init ---*/
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if (first_pass)
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{
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for (step = 4, i = 1; i < 16; i++, step <<= 1)
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look_c_valueL1[i] = (float) (2.0 / (step - 1));
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first_pass = 0;
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}
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pMP3Stream->cs_factorL1 = pMP3Stream->cs_factor[0];
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transform_code = transform_code; /* not used, asm compatability */
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bit_code = 0;
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if (convert_code & 8)
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bit_code = 1;
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convert_code = convert_code & 3; /* higher bits used by dec8 freq cvt */
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if (reduction_code < 0)
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reduction_code = 0;
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if (reduction_code > 2)
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reduction_code = 2;
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if (freq_limit < 1000)
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freq_limit = 1000;
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pMP3Stream->framebytes = framebytes_arg;
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/* check if code handles */
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if (h->option != 3)
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return 0; /* layer I only */
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pMP3Stream->nbatL1 = 32;
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pMP3Stream->max_sb = pMP3Stream->nbatL1;
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/*----- compute pMP3Stream->nsb_limit --------*/
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samprate = sr_table[4 * h->id + h->sr_index];
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pMP3Stream->nsb_limit = (freq_limit * 64L + samprate / 2) / samprate;
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/*- caller limit -*/
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/*---- limit = 0.94*(32>>reduction_code); ----*/
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limit = (32 >> reduction_code);
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if (limit > 8)
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limit--;
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if (pMP3Stream->nsb_limit > limit)
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pMP3Stream->nsb_limit = limit;
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if (pMP3Stream->nsb_limit > pMP3Stream->max_sb)
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pMP3Stream->nsb_limit = pMP3Stream->max_sb;
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pMP3Stream->outvalues = 384 >> reduction_code;
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if (h->mode != 3)
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{ /* adjust for 2 channel modes */
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pMP3Stream->nbatL1 *= 2;
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pMP3Stream->max_sb *= 2;
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pMP3Stream->nsb_limit *= 2;
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}
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/* set sbt function */
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k = 1 + convert_code;
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if (h->mode == 3)
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{
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k = 0;
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}
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pMP3Stream->sbt = sbt_table[bit_code][reduction_code][k];
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pMP3Stream->outvalues *= out_chans[k];
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if (bit_code)
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pMP3Stream->outbytes = pMP3Stream->outvalues;
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else
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pMP3Stream->outbytes = sizeof(short) * pMP3Stream->outvalues;
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decinfo.channels = out_chans[k];
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decinfo.outvalues = pMP3Stream->outvalues;
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decinfo.samprate = samprate >> reduction_code;
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if (bit_code)
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decinfo.bits = 8;
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else
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decinfo.bits = sizeof(short) * 8;
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decinfo.framebytes = pMP3Stream->framebytes;
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decinfo.type = 0;
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/* clear sample buffer, unused sub bands must be 0 */
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for (i = 0; i < 768; i++)
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sample[i] = 0.0F;
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/* init sub-band transform */
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sbt_init();
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return 1;
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}
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/*---------------------------------------------------------*/
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#endif // #ifdef COMPILE_ME
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