3 Ann Hell Ex Machina - Music Software
4 Copyright (C) 2003/2007 Angel Ortega <angel@triptico.com>
6 ss_input.c - Code to load softsynth sounds in different formats
8 This program is free software; you can redistribute it and/or
9 modify it under the terms of the GNU General Public License
10 as published by the Free Software Foundation; either version 2
11 of the License, or (at your option) any later version.
13 This program is distributed in the hope that it will be useful,
14 but WITHOUT ANY WARRANTY; without even the implied warranty of
15 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
16 GNU General Public License for more details.
18 You should have received a copy of the GNU General Public License
19 along with this program; if not, write to the Free Software
20 Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
22 http://www.triptico.com
38 /* maximum page size */
39 int ss_page_size
= 441000;
45 static int fget16(FILE * f
)
46 /* Reads a 16 bit integer from a file in big endian byte ordering */
51 c
+= (fgetc(f
) * 256);
57 static int fget32(FILE * f
)
58 /* Reads a 32 bit integer from a file in big endian byte ordering */
63 c
+= (fgetc(f
) * 256);
64 c
+= (fgetc(f
) * 65536);
65 c
+= (fgetc(f
) * 16777216);
71 static sample_t
load_sample(FILE * f
, int bits
, int sign
)
72 /* loads one sample from a file */
76 /* if on eof, return silence */
86 s
= fget16(f
) - 32768;
88 s
= (short int) fget16(f
);
91 return ((sample_t
) s
) / 32768.0;
95 void load_pcm_wave(FILE * f
, struct ss_wave
*w
)
96 /* loads an interleaved stream from a file */
100 /* fills the channels */
101 for (m
= 0; m
< w
->p_size
; m
++) {
102 for (n
= 0; n
< w
->n_channels
; n
++)
103 w
->wave
[n
][m
] = load_sample(f
, w
->bits
, w
->sign
);
109 * ss_load_wav_file - Loads a file in .WAV format.
110 * @file: name of the file
111 * @base_freq: base frequency
112 * @min_freq: minimum frequency
113 * @max_freq: maximum frequency
114 * @loop_start: frame number of loop start (-1, no loop)
115 * @loop_end: frame number of loop end (-1, end of wave)
116 * @first_channel: first channel to start spreading
117 * @skip_channels: channels to skip when spreading
119 * Loads a file in .WAV format.
121 struct ss_wave
*ss_load_wav_file(const char *file
,
122 double base_freq
, double min_freq
, double max_freq
,
123 double loop_start
, double loop_end
,
124 int first_channel
, int skip_channels
)
129 short int b_per_sec
, n_channels
;
130 char riffid
[5], waveid
[5], fmtid
[5], dataid
[5];
137 file
= transconv_pipe(file
+ 1, ".wav", "ahxm");
139 /* find the file in the library path */
140 if ((file
= libpath_locate(file
)) == NULL
)
144 if ((file
= transconv(file
, ".wav", "ahxm")) == NULL
)
148 if ((f
= fopen(file
, "r")) == NULL
)
151 fread(riffid
, 1, 4, f
);
153 fread(&rlen
, 1, 4, f
);
154 fread(waveid
, 1, 4, f
);
157 if (strcmp(waveid
, "WAVE")) {
162 fread(fmtid
, 1, 4, f
);
169 if (fget16(f
) != 1) {
170 /* wicked compressed format? fail */
175 n_channels
= fget16(f
);
177 b_per_sec
= fget32(f
);
179 bits
= fget16(f
) / n_channels
;
182 fread(dummydata
, 1, (size_t) flen
- 14, f
);
183 fread(dataid
, 1, 4, f
);
186 size
= (double) fget32(f
);
189 size
/= (double) n_channels
;
191 p
= size
> ss_page_size
? ss_page_size
: size
;
193 if ((w
= ss_alloc_wave(size
, n_channels
, s_rate
, p
)) != NULL
) {
194 w
->base_freq
= base_freq
;
195 w
->min_freq
= min_freq
;
196 w
->max_freq
= max_freq
;
198 w
->loop_start
= loop_start
;
203 w
->loop_end
= loop_end
;
205 w
->first_channel
= first_channel
;
206 w
->skip_channels
= skip_channels
;
208 /* fill the info needed for paging */
209 w
->filename
= strdup(file
);
214 /* set the page offset further the end, to
215 force a page reading the first time it's used */
216 w
->p_offset
= (int) size
;
226 * ss_load_pat_file - Loads an instrument in .PAT format.
228 * @filename: filename holding the instrument
230 * Loads data from a Gravis Ultrasound patch (.PAT) format and
231 * stores it as layers for an instrument.
233 * Returns -100 if the file could not be read, -101 or -102
234 * if the file is not recognized as a .PAT file, or 0 if
235 * everything went OK.
237 int ss_load_pat_file(struct ss_ins
*i
, const char *file
)
243 int flags
, bits
, sign
, loop
, pingpong
;
246 if ((f
= libpath_fopen(file
, "r")) == NULL
)
249 /* read signatures */
250 fread(buffer
, 12, 1, f
);
251 if (strcmp(buffer
, "GF1PATCH110") != 0) {
256 fread(buffer
, 10, 1, f
);
257 if (strcmp(buffer
, "ID#000002") != 0) {
262 /* skip description */
263 fread(buffer
, 65, 1, f
);
269 fread(buffer
, 109, 1, f
);
275 fread(buffer
, 40, 1, f
);
277 for (n
= 0; n
< n_layers
; n
++) {
279 double loop_start
, loop_end
;
280 double min_freq
, max_freq
, base_freq
;
283 fread(buffer
, 8, 1, f
);
285 size
= (double) fget32(f
);
286 loop_start
= (double) fget32(f
);
287 loop_end
= (double) fget32(f
);
290 min_freq
= ((double) fget32(f
)) / 1000.0;
291 max_freq
= ((double) fget32(f
)) / 1000.0;
292 base_freq
= ((double) fget32(f
)) / 1000.0;
297 /* ignore fine-tune */
300 /* ignore pan position */
303 /* skip envelope rate, value, tremolo and vibrato */
304 fread(buffer
, 18, 1, f
);
308 bits
= flags
& 0x01 ? 16 : 8;
309 sign
= flags
& 0x02 ? 0 : 1;
310 loop
= flags
& 0x04 ? 1 : 0;
311 pingpong
= flags
& 0x08 ? 1 : 0;
319 /* skip frequency scale data */
324 fread(buffer
, 36, 1, f
);
326 if ((w
= ss_alloc_wave(size
, 1, s_rate
, -1)) == NULL
)
329 /* set the rest of values */
330 w
->loop_start
= loop_start
;
331 w
->loop_end
= loop_end
;
332 w
->base_freq
= base_freq
;
333 w
->min_freq
= min_freq
;
334 w
->max_freq
= max_freq
;
342 if (pingpong
&& loop
) {
346 /* if ping-pong, realloc space for a reverse
347 version of the loop */
348 loop_size
= (int) (loop_end
- loop_start
);
350 ptr
= (sample_t
*) malloc((size
+ loop_size
+ 1)
353 /* transfer start and loop */
354 for (m
= 0; m
<= (int) loop_end
; m
++)
355 ptr
[m
] = w
->wave
[0][m
];
357 /* transfer a reversed version of the loop */
358 for (o
= m
- 1; o
>= loop_start
; o
--, m
++)
359 ptr
[m
] = w
->wave
[0][o
];
361 /* transfer the end */
362 for (o
= loop_end
+ 1; o
< size
; o
++, m
++)
363 ptr
[m
] = w
->wave
[0][o
];
365 w
->loop_end
+= (double) loop_size
;
366 w
->size
+= (double) loop_size
;
367 w
->p_size
+= loop_size
;
377 /* finally add layer to instrument */
378 ss_ins_add_layer(i
, w
);