From 60c1003845035ad4cd0e9ea50862bad7626faf0e Mon Sep 17 00:00:00 2001 From: Pixel Date: Sat, 21 Sep 2002 18:47:46 +0000 Subject: Added the Pcsx source. --- PcsxSrc/Decode_XA.c | 305 ++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 305 insertions(+) create mode 100644 PcsxSrc/Decode_XA.c (limited to 'PcsxSrc/Decode_XA.c') diff --git a/PcsxSrc/Decode_XA.c b/PcsxSrc/Decode_XA.c new file mode 100644 index 0000000..3bd53f4 --- /dev/null +++ b/PcsxSrc/Decode_XA.c @@ -0,0 +1,305 @@ +//============================================ +//=== Audio XA decoding +//=== Kazzuya +//============================================ +//=== Modified by linuzappz +//============================================ + +#include + +#include "Decode_XA.h" + +#ifdef __WIN32__ +#pragma warning(disable:4244) +#endif + +typedef unsigned char U8; +typedef unsigned short U16; +typedef unsigned long U32; + +#define NOT(_X_) (!(_X_)) +#define CLAMP(_X_,_MI_,_MA_) {if(_X_<_MI_)_X_=_MI_;if(_X_>_MA_)_X_=_MA_;} + +//============================================ +//=== ADPCM DECODING ROUTINES +//============================================ + +static double K0[4] = { + 0.0, + 0.9375, + 1.796875, + 1.53125 +}; + +static double K1[4] = { + 0.0, + 0.0, + -0.8125, + -0.859375 +}; + +#define BLKSIZ 28 /* block size (32 - 4 nibbles) */ + +//=========================================== +void ADPCM_InitDecode( ADPCM_Decode_t *decp ) +{ + decp->y0 = 0; + decp->y1 = 0; +} + +//=========================================== +#define SH 4 +#define SHC 10 + +#define IK0(fid) ((int)((-K0[fid]) * (1<> 4) & 0x0f; + range = (filter_range >> 0) & 0x0f; + + fy0 = decp->y0; + fy1 = decp->y1; + + for (i = BLKSIZ/4; i; --i) { + long y; + long x0, x1, x2, x3; + + y = *blockp++; + x3 = (short)( y & 0xf000) >> range; x3 <<= SH; + x2 = (short)((y << 4) & 0xf000) >> range; x2 <<= SH; + x1 = (short)((y << 8) & 0xf000) >> range; x1 <<= SH; + x0 = (short)((y << 12) & 0xf000) >> range; x0 <<= SH; + + x0 -= (IK0(filterid) * fy0 + (IK1(filterid) * fy1)) >> SHC; fy1 = fy0; fy0 = x0; + x1 -= (IK0(filterid) * fy0 + (IK1(filterid) * fy1)) >> SHC; fy1 = fy0; fy0 = x1; + x2 -= (IK0(filterid) * fy0 + (IK1(filterid) * fy1)) >> SHC; fy1 = fy0; fy0 = x2; + x3 -= (IK0(filterid) * fy0 + (IK1(filterid) * fy1)) >> SHC; fy1 = fy0; fy0 = x3; + + CLAMP( x0, -32768<> SH; destp += inc; + CLAMP( x1, -32768<> SH; destp += inc; + CLAMP( x2, -32768<> SH; destp += inc; + CLAMP( x3, -32768<> SH; destp += inc; + } + decp->y0 = fy0; + decp->y1 = fy1; +} + +static int headtable[4] = {0,2,8,10}; + +//=========================================== +static void xa_decode_data( xa_decode_t *xdp, unsigned char *srcp ) { + const U8 *sound_groupsp; + const U8 *sound_datap, *sound_datap2; + int i, j, k, nbits; + U16 data[4096], *datap; + short *destp; + + destp = xdp->pcm; + nbits = xdp->nbits == 4 ? 4 : 2; + + if (xdp->stereo) { // stereo + for (j=0; j < 18; j++) { + sound_groupsp = srcp + j * 128; // sound groups header + sound_datap = sound_groupsp + 16; // sound data just after the header + + for (i=0; i < nbits; i++) { + datap = data; + sound_datap2 = sound_datap + i; + if ((xdp->nbits == 8) && (xdp->freq == 37800)) { // level A + for (k=0; k < 14; k++, sound_datap2 += 8) { + *(datap++) = (U16)sound_datap2[0] | + (U16)(sound_datap2[4] << 8); + } + } else { // level B/C + for (k=0; k < 7; k++, sound_datap2 += 16) { + *(datap++) = (U16)(sound_datap2[ 0] & 0x0f) | + ((U16)(sound_datap2[ 4] & 0x0f) << 4) | + ((U16)(sound_datap2[ 8] & 0x0f) << 8) | + ((U16)(sound_datap2[12] & 0x0f) << 12); + } + } + ADPCM_DecodeBlock16( &xdp->left, sound_groupsp[headtable[i]+0], data, + destp+0, 2 ); + + datap = data; + sound_datap2 = sound_datap + i; + if ((xdp->nbits == 8) && (xdp->freq == 37800)) { // level A + for (k=0; k < 14; k++, sound_datap2 += 8) { + *(datap++) = (U16)sound_datap2[0] | + (U16)(sound_datap2[4] << 8); + } + } else { // level B/C + for (k=0; k < 7; k++, sound_datap2 += 16) { + *(datap++) = (U16)(sound_datap2[ 0] >> 4) | + ((U16)(sound_datap2[ 4] >> 4) << 4) | + ((U16)(sound_datap2[ 8] >> 4) << 8) | + ((U16)(sound_datap2[12] >> 4) << 12); + } + } + ADPCM_DecodeBlock16( &xdp->right, sound_groupsp[headtable[i]+1], data, + destp+1, 2 ); + + destp += 28*2; + } + } + } + else { // mono + for (j=0; j < 18; j++) { + sound_groupsp = srcp + j * 128; // sound groups header + sound_datap = sound_groupsp + 16; // sound data just after the header + + for (i=0; i < nbits; i++) { + datap = data; + sound_datap2 = sound_datap + i; + if ((xdp->nbits == 8) && (xdp->freq == 37800)) { // level A + for (k=0; k < 14; k++, sound_datap2 += 8) { + *(datap++) = (U16)sound_datap2[0] | + (U16)(sound_datap2[4] << 8); + } + } else { // level B/C + for (k=0; k < 7; k++, sound_datap2 += 16) { + *(datap++) = (U16)(sound_datap2[ 0] & 0x0f) | + ((U16)(sound_datap2[ 4] & 0x0f) << 4) | + ((U16)(sound_datap2[ 8] & 0x0f) << 8) | + ((U16)(sound_datap2[12] & 0x0f) << 12); + } + } + ADPCM_DecodeBlock16( &xdp->left, sound_groupsp[headtable[i]+0], data, + destp, 1 ); + + destp += 28; + + datap = data; + sound_datap2 = sound_datap + i; + if ((xdp->nbits == 8) && (xdp->freq == 37800)) { // level A + for (k=0; k < 14; k++, sound_datap2 += 8) { + *(datap++) = (U16)sound_datap2[0] | + (U16)(sound_datap2[4] << 8); + } + } else { // level B/C + for (k=0; k < 7; k++, sound_datap2 += 16) { + *(datap++) = (U16)(sound_datap2[ 0] >> 4) | + ((U16)(sound_datap2[ 4] >> 4) << 4) | + ((U16)(sound_datap2[ 8] >> 4) << 8) | + ((U16)(sound_datap2[12] >> 4) << 12); + } + } + ADPCM_DecodeBlock16( &xdp->left, sound_groupsp[headtable[i]+1], data, + destp, 1 ); + + destp += 28; + } + } + } +} + +//============================================ +//=== XA SPECIFIC ROUTINES +//============================================ +typedef struct { +U8 filenum; +U8 channum; +U8 submode; +U8 coding; + +U8 filenum2; +U8 channum2; +U8 submode2; +U8 coding2; +} xa_subheader_t; + +#define SUB_SUB_EOF (1<<7) // end of file +#define SUB_SUB_RT (1<<6) // real-time sector +#define SUB_SUB_FORM (1<<5) // 0 form1 1 form2 +#define SUB_SUB_TRIGGER (1<<4) // used for interrupt +#define SUB_SUB_DATA (1<<3) // contains data +#define SUB_SUB_AUDIO (1<<2) // contains audio +#define SUB_SUB_VIDEO (1<<1) // contains video +#define SUB_SUB_EOR (1<<0) // end of record + +#define AUDIO_CODING_GET_STEREO(_X_) ( (_X_) & 3) +#define AUDIO_CODING_GET_FREQ(_X_) (((_X_) >> 2) & 3) +#define AUDIO_CODING_GET_BPS(_X_) (((_X_) >> 4) & 3) +#define AUDIO_CODING_GET_EMPHASIS(_X_) (((_X_) >> 6) & 1) + +#define SUB_UNKNOWN 0 +#define SUB_VIDEO 1 +#define SUB_AUDIO 2 + +//============================================ +static int parse_xa_audio_sector( xa_decode_t *xdp, + xa_subheader_t *subheadp, + unsigned char *sectorp, + int is_first_sector ) { + if ( is_first_sector ) { + switch ( AUDIO_CODING_GET_FREQ(subheadp->coding) ) { + case 0: xdp->freq = 37800; break; + case 1: xdp->freq = 18900; break; + default: xdp->freq = 0; break; + } + switch ( AUDIO_CODING_GET_BPS(subheadp->coding) ) { + case 0: xdp->nbits = 4; break; + case 1: xdp->nbits = 8; break; + default: xdp->nbits = 0; break; + } + switch ( AUDIO_CODING_GET_STEREO(subheadp->coding) ) { + case 0: xdp->stereo = 0; break; + case 1: xdp->stereo = 1; break; + default: xdp->stereo = 0; break; + } + + if ( xdp->freq == 0 ) + return -1; + + ADPCM_InitDecode( &xdp->left ); + ADPCM_InitDecode( &xdp->right ); + + xdp->nsamples = 18 * 28 * 8; + if (xdp->stereo == 1) xdp->nsamples /= 2; + } + xa_decode_data( xdp, sectorp ); + + return 0; +} + +//================================================================ +//=== THIS IS WHAT YOU HAVE TO CALL +//=== xdp - structure were all important data are returned +//=== sectorp - data in input +//=== pcmp - data in output +//=== is_first_sector - 1 if it's the 1st sector of the stream +//=== - 0 for any other successive sector +//=== return -1 if error +//================================================================ +long xa_decode_sector( xa_decode_t *xdp, + unsigned char *sectorp, int is_first_sector ) { + if (parse_xa_audio_sector(xdp, (xa_subheader_t *)sectorp, sectorp + sizeof(xa_subheader_t), is_first_sector)) + return -1; + + return 0; +} + +/* EXAMPLE: +"nsamples" is the number of 16 bit samples +every sample is 2 bytes in mono and 4 bytes in stereo + +xa_decode_t xa; + + sectorp = read_first_sector(); + xa_decode_sector( &xa, sectorp, 1 ); + play_wave( xa.pcm, xa.freq, xa.nsamples ); + + while ( --n_sectors ) + { + sectorp = read_next_sector(); + xa_decode_sector( &xa, sectorp, 0 ); + play_wave( xa.pcm, xa.freq, xa.nsamples ); + } +*/ -- cgit v1.2.3