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-rw-r--r--PcsxSrc/Decode_XA.c610
1 files changed, 305 insertions, 305 deletions
diff --git a/PcsxSrc/Decode_XA.c b/PcsxSrc/Decode_XA.c
index 3bd53f4..cc7ebe7 100644
--- a/PcsxSrc/Decode_XA.c
+++ b/PcsxSrc/Decode_XA.c
@@ -1,305 +1,305 @@
-//============================================
-//=== Audio XA decoding
-//=== Kazzuya
-//============================================
-//=== Modified by linuzappz
-//============================================
-
-#include <stdio.h>
-
-#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<<SHC)))
-#define IK1(fid) ((int)((-K1[fid]) * (1<<SHC)))
-
-void ADPCM_DecodeBlock16( ADPCM_Decode_t *decp, U8 filter_range, const void *vblockp, short *destp, int inc ) {
- int i;
- int range, filterid;
- long fy0, fy1;
- const U16 *blockp;
-
- blockp = (const unsigned short *)vblockp;
- filterid = (filter_range >> 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, 32767<<SH ); *destp = x0 >> SH; destp += inc;
- CLAMP( x1, -32768<<SH, 32767<<SH ); *destp = x1 >> SH; destp += inc;
- CLAMP( x2, -32768<<SH, 32767<<SH ); *destp = x2 >> SH; destp += inc;
- CLAMP( x3, -32768<<SH, 32767<<SH ); *destp = x3 >> 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 );
- }
-*/
+//============================================
+//=== Audio XA decoding
+//=== Kazzuya
+//============================================
+//=== Modified by linuzappz
+//============================================
+
+#include <stdio.h>
+
+#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<<SHC)))
+#define IK1(fid) ((int)((-K1[fid]) * (1<<SHC)))
+
+void ADPCM_DecodeBlock16( ADPCM_Decode_t *decp, U8 filter_range, const void *vblockp, short *destp, int inc ) {
+ int i;
+ int range, filterid;
+ long fy0, fy1;
+ const U16 *blockp;
+
+ blockp = (const unsigned short *)vblockp;
+ filterid = (filter_range >> 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, 32767<<SH ); *destp = x0 >> SH; destp += inc;
+ CLAMP( x1, -32768<<SH, 32767<<SH ); *destp = x1 >> SH; destp += inc;
+ CLAMP( x2, -32768<<SH, 32767<<SH ); *destp = x2 >> SH; destp += inc;
+ CLAMP( x3, -32768<<SH, 32767<<SH ); *destp = x3 >> 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 );
+ }
+*/
generated by cgit v1.2.3 (git 2.25.1) at 2024-09-21 19:45:53 +0000