/*____________________________________________________________________________ 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 #include #include #include #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 */ /*-------------------------------------------------------------------------*/