xa variable fix & RXW support added

git-svn-id: https://vgmstream.svn.sourceforge.net/svnroot/vgmstream@102 51a99a44-fe44-0410-b1ba-c3e57ba2b86b

src/layout/xa_blocked.c

@@ -8,14 +8,13 @@ void xa_block_update(off_t block_offset, VGMSTREAM * vgmstream) {
 	uint8_t currentChannel=0;
 	uint8_t subAudio=0;
 
-	// i used interleave_block_size to check sector read length
 	if(vgmstream->samples_into_block!=0)
 		// don't change this variable in the init process
-		vgmstream->interleave_block_size+=128;
+		vgmstream->xa_sector_length+=128;
 
 	// We get to the end of a sector ?
-	if(vgmstream->interleave_block_size==(18*128)) {
+	if(vgmstream->xa_sector_length==(18*128)) {
-		vgmstream->interleave_block_size=0;
+		vgmstream->xa_sector_length=0;
 
 		// 0x30 of unused bytes/sector :(
 		block_offset+=0x30;

src/libvgmstream.vcproj

@@ -250,6 +250,10 @@
 					RelativePath=".\meta\ps2_npsf.c"
 					>
 				</File>
+				<File
+					RelativePath=".\meta\ps2_rxw.c"
+					>
+				</File>
 				<File
 					RelativePath=".\meta\psx_cdxa.c"
 					>

src/meta/meta.h

@@ -37,4 +37,6 @@ VGMSTREAM * init_vgmstream_rwsd(const char * const filename);
 
 VGMSTREAM * init_vgmstream_cdxa(const char * const filename);
 
+VGMSTREAM * init_vgmstream_ps2_rxw(const char * const filename);
+
 #endif

src/meta/ps2_rxw.c

@@ -0,0 +1,79 @@
+#include "meta.h"
+#include "../util.h"
+
+/* RXW file (Arc the Lad) */
+
+VGMSTREAM * init_vgmstream_ps2_rxw(const char * const filename) {
+    VGMSTREAM * vgmstream = NULL;
+    STREAMFILE * infile = NULL;
+
+    int loop_flag=0;
+	int channel_count;
+    off_t start_offset;
+    int i;
+
+    /* check extension, case insensitive */
+    if (strcasecmp("rxw",filename_extension(filename))) goto fail;
+
+    /* try to open the file for header reading */
+    infile = open_streamfile(filename);
+    if (!infile) goto fail;
+
+    /* check NPSF Header */
+    if (!((read_32bitBE(0x00,infile) == 0x52585753) && 
+	      (read_32bitBE(0x10,infile) == 0x464F524D)))
+        goto fail;
+
+	/* check loop */
+	loop_flag = (read_32bitLE(0x3C,infile)!=0xFFFFFFFF);
+    
+	/* Always stereo files */
+	channel_count=2;
+    
+	/* build the VGMSTREAM */
+    vgmstream = allocate_vgmstream(channel_count,loop_flag);
+    if (!vgmstream) goto fail;
+
+	/* fill in the vital statistics */
+	vgmstream->channels = channel_count;
+    vgmstream->sample_rate = read_32bitLE(0x2E,infile);
+
+	/* Check for Compression Scheme */
+	vgmstream->coding_type = coding_PSX;
+    vgmstream->num_samples = (read_32bitLE(0x38,infile)*28/16)/2;
+
+	/* Get loop point values */
+	if(vgmstream->loop_flag) {
+		vgmstream->loop_start_sample = read_32bitLE(0x3C,infile)/16*14;
+		vgmstream->loop_end_sample = read_32bitLE(0x38,infile)/16*14;
+	}
+
+	vgmstream->interleave_block_size = read_32bitLE(0x1c,infile)+0x10;
+    vgmstream->layout_type = layout_interleave;
+    vgmstream->meta_type = meta_PS2_RXW;
+
+	start_offset = 0x40;
+
+    close_streamfile(infile); infile=NULL;
+
+    /* open the file for reading by each channel */
+    {
+        for (i=0;i<channel_count;i++) {
+            vgmstream->ch[i].streamfile = open_streamfile_buffer(filename,0x8000);
+
+            if (!vgmstream->ch[i].streamfile) goto fail;
+
+            vgmstream->ch[i].channel_start_offset=
+                vgmstream->ch[i].offset=
+                start_offset+vgmstream->interleave_block_size*i;
+        }
+    }
+
+    return vgmstream;
+
+    /* clean up anything we may have opened */
+fail:
+    if (infile) close_streamfile(infile);
+    if (vgmstream) close_vgmstream(vgmstream);
+    return NULL;
+}

src/vgmstream.c

@@ -15,7 +15,7 @@
  * List of functions that will recognize files. These should correspond pretty
  * directly to the metadata types
  */
-#define INIT_VGMSTREAM_FCNS 17
+#define INIT_VGMSTREAM_FCNS 18
 VGMSTREAM * (*init_vgmstream_fcns[INIT_VGMSTREAM_FCNS])(const char * const) = {
     init_vgmstream_adx,
     init_vgmstream_brstm,
@@ -34,6 +34,7 @@ VGMSTREAM * (*init_vgmstream_fcns[INIT_VGMSTREAM_FCNS])(const char * const) = {
 	init_vgmstream_ps2_npsf,
     init_vgmstream_rwsd,
 	init_vgmstream_cdxa,
+	init_vgmstream_ps2_rxw,
 };
 
 
@@ -217,7 +218,7 @@ int get_vgmstream_frame_size(VGMSTREAM * vgmstream) {
         case coding_PSX:
             return 16;
 		case coding_XA:
-			return 28;
+			return 14*vgmstream->channels;
         default:
             return 0;
     }
@@ -362,6 +363,8 @@ int vgmstream_do_loop(VGMSTREAM * vgmstream) {
                 for (i=0;i<vgmstream->channels;i++) {
                     vgmstream->loop_ch[i].adpcm_history1_16 = vgmstream->ch[i].adpcm_history1_16;
                     vgmstream->loop_ch[i].adpcm_history2_16 = vgmstream->ch[i].adpcm_history2_16;
+                    vgmstream->loop_ch[i].adpcm_history1_32 = vgmstream->ch[i].adpcm_history1_32;
+                    vgmstream->loop_ch[i].adpcm_history2_32 = vgmstream->ch[i].adpcm_history2_32;
                 }
             }
 #if DEBUG
@@ -583,6 +586,9 @@ void describe_vgmstream(VGMSTREAM * vgmstream, char * desc, int length) {
             break;
         case meta_PSX_XA:
             snprintf(temp,TEMPSIZE,"RIFF/CDXA Header");
+            break;
+		case meta_PS2_RXW:
+            snprintf(temp,TEMPSIZE,"RXWS File (Arc The Lad)");
             break;
         default:
             snprintf(temp,TEMPSIZE,"THEY SHOULD HAVE SENT A POET");

src/vgmstream.h

@@ -76,8 +76,10 @@ typedef enum {
 
 	meta_PS2_SShd,			/* .ADS with SShd header */
 	meta_PS2_NPSF,			/* Namco Production Sound File */
+	meta_PS2_RXW,			/* Sony Arc The Lad Sound File */
+
+	meta_PSX_XA,			/* CD-XA with RIFF header */
 
-	meta_PSX_XA,
 } meta_t;
 
 typedef struct {
@@ -156,6 +158,7 @@ typedef struct {
     off_t loop_next_block_offset;   /* saved from next_block_offset */
 
 	uint8_t xa_channel;				/* Selected XA Channel */
+	int32_t xa_sector_length;		/* XA block */
 } VGMSTREAM;
 
 /* do format detection, return pointer to a usable VGMSTREAM, or NULL on failure */

winamp/in_vgmstream.c

@@ -45,7 +45,7 @@ int fade_samples = 0;
 
 #define EXTENSION_LIST_SIZE 1024
 char working_extension_list[EXTENSION_LIST_SIZE] = {0};
-#define EXTENSION_COUNT 16
+#define EXTENSION_COUNT 17
 char * extension_list[EXTENSION_COUNT] = {
     "adx\0ADX Audio File (*.ADX)\0",
     "afc\0AFC Audio File (*.AFC)\0",
@@ -63,6 +63,7 @@ char * extension_list[EXTENSION_COUNT] = {
 	"npsf\0PS2 NPSF Audio File (*.NPSF)\0",
     "rwsd\0RWSD Audio File (*.RWSD)\0",
 	"xa\0PSX CD-XA File (*.XA)\0",
+	"rxw\0PS2 RXWS File (*.RXW)\0",
 };
 
 /* stubs, we don't do anything fancy yet */
@@ -139,7 +140,7 @@ int play(char *fn)
     decode_pos_samples = 0;
     paused = 0;
     stream_length_samples = get_vgmstream_play_samples(loop_count,fade_seconds,vgmstream);
-    fade_samples = fade_seconds * vgmstream->sample_rate;
+    fade_samples = (int)(fade_seconds * vgmstream->sample_rate);
 
     decode_thread_handle = CreateThread(
             NULL,   /* handle cannot be inherited */
@@ -296,7 +297,7 @@ DWORD WINAPI __stdcall decode(void *arg) {
                             double fadedness = (double)(fade_samples-samples_into_fade)/fade_samples;
                             for (k=0;k<vgmstream->channels;k++) {
                                 sample_buffer[j*vgmstream->channels+k] =
-                                    sample_buffer[j*vgmstream->channels+k]*fadedness;
+                                    (short)(sample_buffer[j*vgmstream->channels+k]*fadedness);
                             }
                         }
                     }