Temporarily reverted r64524 until I (or someone else) find the time to sort out the stuff.

git-svn-id: https://pcsxr.svn.codeplex.com/svn/pcsxr@64536 e17a0e51-4ae3-4d35-97c3-1a29b211df97

1 files changed, 367 insertions, 367 deletions

diff --git a/libpcsxcore/decode_xa.c b/libpcsxcore/decode_xa.c
index abca4cc7..9198659e 100644
--- a/libpcsxcore/decode_xa.c
+++ b/libpcsxcore/decode_xa.c
@@ -1,367 +1,367 @@
-/***************************************************************************
- *   Copyright (C) 2007 Ryan Schultz, PCSX-df Team, PCSX team              *
- *                                                                         *
- *   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.,                                       *
- *   51 Franklin Street, Fifth Floor, Boston, MA 02111-1307 USA.           *
- ***************************************************************************/
-
-/* 
-* XA audio decoding functions (Kazzuya).
-*/
-
-#include "decode_xa.h"
-
-#define FIXED
-
-#define NOT(_X_)				(!(_X_))
-#define XACLAMP(_X_,_MI_,_MA_)	{if(_X_<_MI_)_X_=_MI_;if(_X_>_MA_)_X_=_MA_;}
-
-#define SH	4
-#define SHC	10
-
-//============================================
-//===  ADPCM DECODING ROUTINES
-//============================================
-
-#ifndef FIXED
-static double K0[4] = {
-    0.0,
-    0.9375,
-    1.796875,
-    1.53125
-};
-
-static double K1[4] = {
-    0.0,
-    0.0,
-    -0.8125,
-    -0.859375
-};
-#else
-static int K0[4] = {
-	0.0       * (1<<SHC),
-	0.9375    * (1<<SHC),
-	1.796875  * (1<<SHC),
-	1.53125   * (1<<SHC)
-};
- 
-static int K1[4] = {
-	0.0       * (1<<SHC),
-	0.0       * (1<<SHC),
-	-0.8125   * (1<<SHC),
-	-0.859375 * (1<<SHC)
-};
-#endif
-
-#define BLKSIZ 28       /* block size (32 - 4 nibbles) */
-
-//===========================================
-static void ADPCM_InitDecode(ADPCM_Decode_t *decp) {
-	decp->y0 = 0;
-	decp->y1 = 0;
-}
-
-//===========================================
-#ifndef FIXED
-#define IK0(fid)	((int)((-K0[fid]) * (1<<SHC)))
-#define IK1(fid)	((int)((-K1[fid]) * (1<<SHC)))
-#else
-#define IK0(fid)	(-K0[fid])
-#define IK1(fid)	(-K1[fid])
-#endif
-
-static __inline void ADPCM_DecodeBlock16( ADPCM_Decode_t *decp, u8 filter_range, const void *vblockp, short *destp, int inc ) {
-	int i;
-	int range, filterid;
-	s32 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) {
-		s32 y;
-		s32 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;
-
-		XACLAMP( x0, -32768<<SH, 32767<<SH ); *destp = x0 >> SH; destp += inc;
-		XACLAMP( x1, -32768<<SH, 32767<<SH ); *destp = x1 >> SH; destp += inc;
-		XACLAMP( x2, -32768<<SH, 32767<<SH ); *destp = x2 >> SH; destp += inc;
-		XACLAMP( 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
-		if ((xdp->nbits == 8) && (xdp->freq == 37800)) { // level A
-			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;
-
-					for (k=0; k < 14; k++, sound_datap2 += 8) {
-        	   				*(datap++) = (u16)sound_datap2[0] |
-        	               			     (u16)(sound_datap2[4] << 8);
-					}
-
-    				ADPCM_DecodeBlock16( &xdp->left,  sound_groupsp[headtable[i]+0], data,
-        	           				    destp+0, 2 );
-
-        			datap = data;
-        			sound_datap2 = sound_datap + i;
-        			for (k=0; k < 14; k++, sound_datap2 += 8) {
-           					*(datap++) = (u16)sound_datap2[0] |
-            	          			     (u16)(sound_datap2[4] << 8);
-					}
-					ADPCM_DecodeBlock16( &xdp->right,  sound_groupsp[headtable[i]+1], data,
-                           			    destp+1, 2 );
-
-	        		destp += 28*2;
-				}
-    		}
-		} else { // level B/C
-			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;
-
-        			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;
-        			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
-		if ((xdp->nbits == 8) && (xdp->freq == 37800)) { // level A
-			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;
-        			for (k=0; k < 14; k++, sound_datap2 += 8) {
-           					*(datap++) = (u16)sound_datap2[0] |
-                       				     (u16)(sound_datap2[4] << 8);
-					}
-	        		ADPCM_DecodeBlock16( &xdp->left,  sound_groupsp[headtable[i]+0], data,
-                           			    destp, 1 );
-
-	        		destp += 28;
-
-	        		datap = data;
-	        		sound_datap2 = sound_datap + i;
-        			for (k=0; k < 14; k++, sound_datap2 += 8) {
-           					*(datap++) = (u16)sound_datap2[0] |
-                       				     (u16)(sound_datap2[4] << 8);
-					}
-	       			ADPCM_DecodeBlock16( &xdp->left,  sound_groupsp[headtable[i]+1], data,
-                           			    destp, 1 );
-
-					destp += 28;
-				}
-	    	}
-		} else { // level B/C
-			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;
-        			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;
-        			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
-//================================================================
-s32 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 );
-	}
-*/
+/***************************************************************************

+ *   Copyright (C) 2007 Ryan Schultz, PCSX-df Team, PCSX team              *

+ *                                                                         *

+ *   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.,                                       *

+ *   51 Franklin Street, Fifth Floor, Boston, MA 02111-1307 USA.           *

+ ***************************************************************************/

+

+/* 

+* XA audio decoding functions (Kazzuya).

+*/

+

+#include "decode_xa.h"

+

+#define FIXED

+

+#define NOT(_X_)				(!(_X_))

+#define XACLAMP(_X_,_MI_,_MA_)	{if(_X_<_MI_)_X_=_MI_;if(_X_>_MA_)_X_=_MA_;}

+

+#define SH	4

+#define SHC	10

+

+//============================================

+//===  ADPCM DECODING ROUTINES

+//============================================

+

+#ifndef FIXED

+static double K0[4] = {

+    0.0,

+    0.9375,

+    1.796875,

+    1.53125

+};

+

+static double K1[4] = {

+    0.0,

+    0.0,

+    -0.8125,

+    -0.859375

+};

+#else

+static int K0[4] = {

+	0.0       * (1<<SHC),

+	0.9375    * (1<<SHC),

+	1.796875  * (1<<SHC),

+	1.53125   * (1<<SHC)

+};

+ 

+static int K1[4] = {

+	0.0       * (1<<SHC),

+	0.0       * (1<<SHC),

+	-0.8125   * (1<<SHC),

+	-0.859375 * (1<<SHC)

+};

+#endif

+

+#define BLKSIZ 28       /* block size (32 - 4 nibbles) */

+

+//===========================================

+void ADPCM_InitDecode(ADPCM_Decode_t *decp) {

+	decp->y0 = 0;

+	decp->y1 = 0;

+}

+

+//===========================================

+#ifndef FIXED

+#define IK0(fid)	((int)((-K0[fid]) * (1<<SHC)))

+#define IK1(fid)	((int)((-K1[fid]) * (1<<SHC)))

+#else

+#define IK0(fid)	(-K0[fid])

+#define IK1(fid)	(-K1[fid])

+#endif

+

+static __inline void ADPCM_DecodeBlock16( ADPCM_Decode_t *decp, u8 filter_range, const void *vblockp, short *destp, int inc ) {

+	int i;

+	int range, filterid;

+	s32 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) {

+		s32 y;

+		s32 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;

+

+		XACLAMP( x0, -32768<<SH, 32767<<SH ); *destp = x0 >> SH; destp += inc;

+		XACLAMP( x1, -32768<<SH, 32767<<SH ); *destp = x1 >> SH; destp += inc;

+		XACLAMP( x2, -32768<<SH, 32767<<SH ); *destp = x2 >> SH; destp += inc;

+		XACLAMP( 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

+		if ((xdp->nbits == 8) && (xdp->freq == 37800)) { // level A

+			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;

+

+					for (k=0; k < 14; k++, sound_datap2 += 8) {

+        	   				*(datap++) = (u16)sound_datap2[0] |

+        	               			     (u16)(sound_datap2[4] << 8);

+					}

+

+    				ADPCM_DecodeBlock16( &xdp->left,  sound_groupsp[headtable[i]+0], data,

+        	           				    destp+0, 2 );

+

+        			datap = data;

+        			sound_datap2 = sound_datap + i;

+        			for (k=0; k < 14; k++, sound_datap2 += 8) {

+           					*(datap++) = (u16)sound_datap2[0] |

+            	          			     (u16)(sound_datap2[4] << 8);

+					}

+					ADPCM_DecodeBlock16( &xdp->right,  sound_groupsp[headtable[i]+1], data,

+                           			    destp+1, 2 );

+

+	        		destp += 28*2;

+				}

+    		}

+		} else { // level B/C

+			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;

+

+        			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;

+        			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

+		if ((xdp->nbits == 8) && (xdp->freq == 37800)) { // level A

+			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;

+        			for (k=0; k < 14; k++, sound_datap2 += 8) {

+           					*(datap++) = (u16)sound_datap2[0] |

+                       				     (u16)(sound_datap2[4] << 8);

+					}

+	        		ADPCM_DecodeBlock16( &xdp->left,  sound_groupsp[headtable[i]+0], data,

+                           			    destp, 1 );

+

+	        		destp += 28;

+

+	        		datap = data;

+	        		sound_datap2 = sound_datap + i;

+        			for (k=0; k < 14; k++, sound_datap2 += 8) {

+           					*(datap++) = (u16)sound_datap2[0] |

+                       				     (u16)(sound_datap2[4] << 8);

+					}

+	       			ADPCM_DecodeBlock16( &xdp->left,  sound_groupsp[headtable[i]+1], data,

+                           			    destp, 1 );

+

+					destp += 28;

+				}

+	    	}

+		} else { // level B/C

+			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;

+        			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;

+        			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

+//================================================================

+s32 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 );

+	}

+*/