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Diffstat (limited to 'plugins/dumb/dumb-kode54/src/it/readasy.c')
-rw-r--r--plugins/dumb/dumb-kode54/src/it/readasy.c331
1 files changed, 331 insertions, 0 deletions
diff --git a/plugins/dumb/dumb-kode54/src/it/readasy.c b/plugins/dumb/dumb-kode54/src/it/readasy.c
new file mode 100644
index 00000000..4c1c09f8
--- /dev/null
+++ b/plugins/dumb/dumb-kode54/src/it/readasy.c
@@ -0,0 +1,331 @@
+/* _______ ____ __ ___ ___
+ * \ _ \ \ / \ / \ \ / / ' ' '
+ * | | \ \ | | || | \/ | . .
+ * | | | | | | || ||\ /| |
+ * | | | | | | || || \/ | | ' ' '
+ * | | | | | | || || | | . .
+ * | |_/ / \ \__// || | |
+ * /_______/ynamic \____/niversal /__\ /____\usic /| . . ibliotheque
+ * / \
+ * / . \
+ * readasy.c - Code to read an ASYLUM Music Format / / \ \
+ * module from an open file. | < / \_
+ * | \/ /\ /
+ * By Chris Moeller. \_ / > /
+ * | \ / /
+ * | ' /
+ * \__/
+ */
+
+#include <stdlib.h>
+#include <string.h>
+#include <math.h>
+
+#include "dumb.h"
+#include "internal/it.h"
+
+
+
+static int it_asy_read_pattern( IT_PATTERN *pattern, DUMBFILE *f, unsigned char *buffer )
+{
+ int pos;
+ int channel;
+ int row;
+ IT_ENTRY *entry;
+
+ pattern->n_rows = 64;
+
+ if ( dumbfile_getnc( buffer, 64 * 8 * 4, f ) != 64 * 8 * 4 )
+ return -1;
+
+ /* compute number of entries */
+ pattern->n_entries = 64; /* Account for the row end markers */
+ pos = 0;
+ for ( row = 0; row < 64; ++row ) {
+ for ( channel = 0; channel < 8; ++channel ) {
+ if ( buffer[ pos + 0 ] | buffer[ pos + 1 ] | buffer[ pos + 2 ] | buffer[ pos + 3 ] )
+ ++pattern->n_entries;
+ pos += 4;
+ }
+ }
+
+ pattern->entry = malloc( pattern->n_entries * sizeof( *pattern->entry ) );
+ if ( !pattern->entry )
+ return -1;
+
+ entry = pattern->entry;
+ pos = 0;
+ for ( row = 0; row < 64; ++row ) {
+ for ( channel = 0; channel < 8; ++channel ) {
+ if ( buffer[ pos + 0 ] | buffer[ pos + 1 ] | buffer[ pos + 2 ] | buffer[ pos + 3 ] ) {
+ entry->channel = channel;
+ entry->mask = 0;
+
+ if ( buffer[ pos + 0 ] && buffer[ pos + 0 ] < 96 ) {
+ entry->note = buffer[ pos + 0 ];
+ entry->mask |= IT_ENTRY_NOTE;
+ }
+
+ if ( buffer[ pos + 1 ] && buffer[ pos + 1 ] <= 64 ) {
+ entry->instrument = buffer[ pos + 1 ];
+ entry->mask |= IT_ENTRY_INSTRUMENT;
+ }
+
+ _dumb_it_xm_convert_effect( buffer[ pos + 2 ] & 0x0F, buffer[ pos + 3 ], entry, 1 );
+
+ if ( entry->mask ) ++entry;
+ }
+ pos += 4;
+ }
+ IT_SET_END_ROW( entry );
+ ++entry;
+ }
+
+ pattern->n_entries = entry - pattern->entry;
+
+ return 0;
+}
+
+
+
+static int it_asy_read_sample_header( IT_SAMPLE *sample, DUMBFILE *f )
+{
+ int finetune;
+
+/**
+ 21 22 Chars Sample 1 name. If the name is not a full
+ 22 chars in length, it will be null
+ terminated.
+
+If
+the sample name begins with a '#' character (ASCII $23 (35)) then this is
+assumed not to be an instrument name, and is probably a message.
+*/
+ dumbfile_getnc( sample->name, 22, f );
+ sample->name[22] = 0;
+
+ sample->filename[0] = 0;
+
+/** Each finetune step changes the note 1/8th of a semitone. */
+ finetune = ( signed char ) ( dumbfile_getc( f ) << 4 ) >> 4; /* signed nibble */
+ sample->default_volume = dumbfile_getc( f ); // Should we be setting global_volume to this instead?
+ sample->global_volume = 64;
+ if ( sample->default_volume > 64 ) sample->default_volume = 64;
+ dumbfile_skip( f, 1 ); /* XXX unknown */
+ sample->length = dumbfile_igetl( f );
+ sample->loop_start = dumbfile_igetl( f );
+ sample->loop_end = sample->loop_start + dumbfile_igetl( f );
+
+ if ( sample->length <= 0 ) {
+ sample->flags = 0;
+ return 0;
+ }
+
+ sample->flags = IT_SAMPLE_EXISTS;
+
+ sample->default_pan = 0;
+ sample->C5_speed = (int)( AMIGA_CLOCK / 214.0 );//( long )( 16726.0 * pow( DUMB_PITCH_BASE, finetune * 32 ) );
+ sample->finetune = finetune * 32;
+ // the above line might be wrong
+
+ if ( ( sample->loop_end - sample->loop_start > 2 ) && ( sample->loop_end <= sample->length ) )
+ sample->flags |= IT_SAMPLE_LOOP;
+
+ sample->vibrato_speed = 0;
+ sample->vibrato_depth = 0;
+ sample->vibrato_rate = 0;
+ sample->vibrato_waveform = 0; // do we have to set _all_ these?
+ sample->max_resampling_quality = -1;
+
+ return dumbfile_error(f);
+}
+
+
+
+static int it_asy_read_sample_data( IT_SAMPLE *sample, DUMBFILE *f )
+{
+ long truncated_size;
+
+ /* let's get rid of the sample data coming after the end of the loop */
+ if ( ( sample->flags & IT_SAMPLE_LOOP ) && sample->loop_end < sample->length ) {
+ truncated_size = sample->length - sample->loop_end;
+ sample->length = sample->loop_end;
+ } else {
+ truncated_size = 0;
+ }
+
+ sample->data = malloc( sample->length );
+
+ if ( !sample->data )
+ return -1;
+
+ if ( sample->length )
+ dumbfile_getnc( sample->data, sample->length, f );
+
+ dumbfile_skip( f, truncated_size );
+
+ return dumbfile_error( f );
+}
+
+
+
+static DUMB_IT_SIGDATA *it_asy_load_sigdata(DUMBFILE *f)
+{
+ DUMB_IT_SIGDATA *sigdata;
+ int i;
+
+ static const char sig_part[] = "ASYLUM Music Format";
+ static const char sig_rest[] = " V1.0"; /* whee, string space optimization with format type below */
+
+ char signature [32];
+
+ if ( dumbfile_getnc( signature, 32, f ) != 32 ||
+ memcmp( signature, sig_part, 19 ) ||
+ memcmp( signature + 19, sig_rest, 5 ) ) {
+ return NULL;
+ }
+
+ sigdata = malloc(sizeof(*sigdata));
+ if (!sigdata) {
+ return NULL;
+ }
+
+ sigdata->speed = dumbfile_getc( f ); /* XXX seems to fit the files I have */
+ sigdata->tempo = dumbfile_getc( f ); /* ditto */
+ sigdata->n_samples = dumbfile_getc( f ); /* ditto */
+ sigdata->n_patterns = dumbfile_getc( f );
+ sigdata->n_orders = dumbfile_getc( f );
+ sigdata->restart_position = dumbfile_getc( f );
+
+ if ( dumbfile_error( f ) || !sigdata->n_samples || sigdata->n_samples > 64 || !sigdata->n_patterns ||
+ !sigdata->n_orders ) {
+ free( sigdata );
+ return NULL;
+ }
+
+ if ( sigdata->restart_position > sigdata->n_orders ) /* XXX */
+ sigdata->restart_position = 0;
+
+ sigdata->order = malloc( sigdata->n_orders );
+ if ( !sigdata->order ) {
+ free( sigdata );
+ return NULL;
+ }
+
+ if ( dumbfile_getnc( sigdata->order, sigdata->n_orders, f ) != sigdata->n_orders ||
+ dumbfile_skip( f, 256 - sigdata->n_orders ) ) {
+ free( sigdata->order );
+ free( sigdata );
+ return NULL;
+ }
+
+ sigdata->sample = malloc( sigdata->n_samples * sizeof( *sigdata->sample ) );
+ if ( !sigdata->sample ) {
+ free( sigdata->order );
+ free( sigdata );
+ return NULL;
+ }
+
+ sigdata->song_message = NULL;
+ sigdata->instrument = NULL;
+ sigdata->pattern = NULL;
+ sigdata->midi = NULL;
+ sigdata->checkpoint = NULL;
+
+ sigdata->n_instruments = 0;
+
+ for ( i = 0; i < sigdata->n_samples; ++i )
+ sigdata->sample[i].data = NULL;
+
+ for ( i = 0; i < sigdata->n_samples; ++i ) {
+ if ( it_asy_read_sample_header( &sigdata->sample[i], f ) ) {
+ _dumb_it_unload_sigdata( sigdata );
+ return NULL;
+ }
+ }
+
+ if ( dumbfile_skip( f, 37 * ( 64 - sigdata->n_samples ) ) ) {
+ _dumb_it_unload_sigdata( sigdata );
+ return NULL;
+ }
+
+ sigdata->pattern = malloc( sigdata->n_patterns * sizeof( *sigdata->pattern ) );
+ if ( !sigdata->pattern ) {
+ _dumb_it_unload_sigdata( sigdata );
+ return NULL;
+ }
+ for (i = 0; i < sigdata->n_patterns; ++i)
+ sigdata->pattern[i].entry = NULL;
+
+ /* Read in the patterns */
+ {
+ unsigned char *buffer = malloc( 64 * 8 * 4 ); /* 64 rows * 8 channels * 4 bytes */
+ if ( !buffer ) {
+ _dumb_it_unload_sigdata( sigdata );
+ return NULL;
+ }
+ for ( i = 0; i < sigdata->n_patterns; ++i ) {
+ if ( it_asy_read_pattern( &sigdata->pattern[i], f, buffer ) != 0 ) {
+ free( buffer );
+ _dumb_it_unload_sigdata( sigdata );
+ return NULL;
+ }
+ }
+ free( buffer );
+ }
+
+ /* And finally, the sample data */
+ for ( i = 0; i < sigdata->n_samples; ++i ) {
+ if ( it_asy_read_sample_data( &sigdata->sample[i], f ) ) {
+ _dumb_it_unload_sigdata( sigdata );
+ return NULL;
+ }
+ }
+
+ /* Now let's initialise the remaining variables, and we're done! */
+ sigdata->flags = IT_WAS_AN_XM | IT_WAS_A_MOD | IT_OLD_EFFECTS | IT_COMPATIBLE_GXX | IT_STEREO;
+
+ sigdata->global_volume = 128;
+ sigdata->mixing_volume = 48;
+ sigdata->pan_separation = 128;
+
+ sigdata->n_pchannels = 8;
+
+ sigdata->name[0] = 0;
+
+ memset(sigdata->channel_volume, 64, DUMB_IT_N_CHANNELS);
+
+ for (i = 0; i < DUMB_IT_N_CHANNELS; i += 4) {
+ sigdata->channel_pan[i+0] = 16;
+ sigdata->channel_pan[i+1] = 48;
+ sigdata->channel_pan[i+2] = 48;
+ sigdata->channel_pan[i+3] = 16;
+ }
+
+ _dumb_it_fix_invalid_orders(sigdata);
+
+ return sigdata;
+}
+
+
+
+DUH *dumb_read_asy_quick(DUMBFILE *f)
+{
+ sigdata_t *sigdata;
+
+ DUH_SIGTYPE_DESC *descptr = &_dumb_sigtype_it;
+
+ sigdata = it_asy_load_sigdata(f);
+
+ if (!sigdata)
+ return NULL;
+
+ {
+ const char *tag[2][2];
+ tag[0][0] = "TITLE";
+ tag[0][1] = ((DUMB_IT_SIGDATA *)sigdata)->name;
+ tag[1][0] = "FORMAT";
+ tag[1][1] = "ASYLUM Music Format";
+ return make_duh(-1, 2, (const char *const (*)[2])tag, 1, &descptr, &sigdata);
+ }
+}