Lua fscanf: use pointer of file descriptor instead of file descriptor itself to avoid...
[kugel-rb.git] / apps / codecs / mod.c
blob3e2e4284deab784f11958d87dae4bae92e0797b1
1 /***************************************************************************
2 * __________ __ ___.
3 * Open \______ \ ____ ____ | | _\_ |__ _______ ___
4 * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
5 * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
6 * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
7 * \/ \/ \/ \/ \/
8 * $Id$
10 * MOD Codec for rockbox
12 * Written from scratch by Rainer Sinsch
13 * exclusivly for Rockbox in February 2008
15 * This program is free software; you can redistribute it and/or
16 * modify it under the terms of the GNU General Public License
17 * as published by the Free Software Foundation; either version 2
18 * of the License, or (at your option) any later version.
20 * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
21 * KIND, either express or implied.
23 ****************************************************************************/
25 /**************
26 * This version supports large files directly from internal memory management.
27 * There is a drawback however: It may happen that a song is not completely
28 * loaded when the internal rockbox-ringbuffer (approx. 28MB) is filled up
29 * As a workaround make sure you don't have directories with mods larger
30 * than a total of 28MB
31 *************/
33 #include "debug.h"
34 #include "codeclib.h"
35 #include <inttypes.h>
37 #include <stdio.h>
38 #include <string.h>
39 #include <stdlib.h>
40 #include <ctype.h>
43 CODEC_HEADER
45 #define CHUNK_SIZE (1024*2)
48 /* This codec supports MOD Files:
52 static int32_t samples[CHUNK_SIZE] IBSS_ATTR; /* The sample buffer */
54 /* Instrument Data */
55 struct s_instrument {
56 /* Sample name / description */
57 /*char description[22];*/
59 /* Sample length in bytes */
60 unsigned short length;
62 /* Sample finetuning (-8 - +7) */
63 signed char finetune;
65 /* Sample volume (0 - 64) */
66 signed char volume;
68 /* Sample Repeat Position */
69 unsigned short repeatoffset;
71 /* Sample Repeat Length */
72 unsigned short repeatlength;
74 /* Offset to sample data */
75 unsigned int sampledataoffset;
78 /* Song Data */
79 struct s_song {
80 /* Song name / title description */
81 /*char szTitle[20];*/
83 /* No. of channels in song */
84 unsigned char noofchannels;
86 /* No. of instruments used (either 15 or 31) */
87 unsigned char noofinstruments;
89 /* How many patterns are beeing played? */
90 unsigned char songlength;
92 /* Where to jump after the song end? */
93 unsigned char songendjumpposition;
95 /* Pointer to the Pattern Order Table */
96 unsigned char *patternordertable;
98 /* Pointer to the pattern data */
99 void *patterndata;
101 /* Pointer to the sample buffer */
102 signed char *sampledata;
104 /* Instrument data */
105 struct s_instrument instrument[31];
108 struct s_modchannel {
109 /* Current Volume */
110 signed char volume;
112 /* Current Offset to period in PeriodTable of notebeeing played
113 (can be temporarily negative) */
114 short periodtableoffset;
116 /* Current Period beeing played */
117 short period;
119 /* Current effect */
120 unsigned char effect;
122 /* Current parameters of effect */
123 unsigned char effectparameter;
125 /* Current Instrument beeing played */
126 unsigned char instrument;
128 /* Current Vibrato Speed */
129 unsigned char vibratospeed;
131 /* Current Vibrato Depth */
132 unsigned char vibratodepth;
134 /* Current Position for Vibrato in SinTable */
135 unsigned char vibratosinpos;
137 /* Current Tremolo Speed */
138 unsigned char tremolospeed;
140 /* Current Tremolo Depth */
141 unsigned char tremolodepth;
143 /* Current Position for Tremolo in SinTable */
144 unsigned char tremolosinpos;
146 /* Current Speed of Effect "Slide Note up" */
147 unsigned char slideupspeed;
149 /* Current Speed of Effect "Slide Note down" */
150 unsigned char slidedownspeed;
152 /* Current Speed of the "Slide to Note" effect */
153 unsigned char slidetonotespeed;
155 /* Current Period of the "Slide to Note" effect */
156 unsigned short slidetonoteperiod;
159 struct s_modplayer {
160 /* Ticks per Line */
161 unsigned char ticksperline;
163 /* Beats per Minute */
164 unsigned char bpm;
166 /* Position of the Song in the Pattern Table (0-127) */
167 unsigned char patterntableposition;
169 /* Current Line (may be temporarily < 0) */
170 signed char currentline;
172 /* Current Tick */
173 signed char currenttick;
175 /* How many samples are required to calculate for each tick? */
176 unsigned int samplespertick;
178 /* Information about the channels */
179 struct s_modchannel modchannel[8];
181 /* The Amiga Period Table
182 (+1 because we use index 0 for period 0 = no new note) */
183 unsigned short periodtable[37*8+1];
185 /* The sinus table [-255,255] */
186 signed short sintable[0x40];
188 /* Is the glissando effect enabled? */
189 bool glissandoenabled;
191 /* Is the Amiga Filter enabled? */
192 bool amigafilterenabled;
194 /* The pattern-line where the loop is carried out (set with e6 command) */
195 unsigned char loopstartline;
197 /* Number of times to loop */
198 unsigned char looptimes;
201 struct s_channel {
202 /* Panning (0 = left, 16 = right) */
203 unsigned char panning;
205 /* Sample frequency of the channel */
206 unsigned short frequency;
208 /* Position of the sample currently played */
209 unsigned int samplepos;
211 /* Fractual Position of the sample currently player */
212 unsigned int samplefractpos;
214 /* Loop Sample */
215 bool loopsample;
217 /* Loop Position Start */
218 unsigned int loopstart;
220 /* Loop Position End */
221 unsigned int loopend;
223 /* Is The channel beeing played? */
224 bool channelactive;
226 /* The Volume (0..64) */
227 signed char volume;
229 /* The last sampledata beeing played (required for interpolation) */
230 signed short lastsampledata;
233 struct s_mixer {
234 /* The channels */
235 struct s_channel channel[32];
238 struct s_song modsong IDATA_ATTR; /* The Song */
239 struct s_modplayer modplayer IDATA_ATTR; /* The Module Player */
240 struct s_mixer mixer IDATA_ATTR;
242 const unsigned short mixingrate = 44100;
244 STATICIRAM void mixer_playsample(int channel, int instrument) ICODE_ATTR;
245 void mixer_playsample(int channel, int instrument)
247 struct s_channel *p_channel = &mixer.channel[channel];
248 struct s_instrument *p_instrument = &modsong.instrument[instrument];
250 p_channel->channelactive = true;
251 p_channel->samplepos = p_instrument->sampledataoffset;
252 p_channel->samplefractpos = 0;
253 p_channel->loopsample = (p_instrument->repeatlength > 2);
254 if (p_channel->loopsample) {
255 p_channel->loopstart = p_instrument->repeatoffset +
256 p_instrument->sampledataoffset;
257 p_channel->loopend = p_channel->loopstart +
258 p_instrument->repeatlength;
260 else p_channel->loopend = p_instrument->length +
261 p_instrument->sampledataoffset;
263 /* Remember the instrument */
264 modplayer.modchannel[channel].instrument = instrument;
267 static inline void mixer_stopsample(int channel)
269 mixer.channel[channel].channelactive = false;
272 static inline void mixer_continuesample(int channel)
274 mixer.channel[channel].channelactive = true;
277 static inline void mixer_setvolume(int channel, int volume)
279 mixer.channel[channel].volume = volume;
282 static inline void mixer_setpanning(int channel, int panning)
284 mixer.channel[channel].panning = panning;
287 static inline void mixer_setamigaperiod(int channel, int amigaperiod)
289 /* Just to make sure we don't devide by zero
290 * amigaperiod shouldn't 0 anyway - if it is the case
291 * then something terribly went wrong */
292 if (amigaperiod == 0)
293 return;
295 mixer.channel[channel].frequency = 3579546 / amigaperiod;
298 /* Initialize the MOD Player with default values and precalc tables */
299 STATICIRAM void initmodplayer(void) ICODE_ATTR;
300 void initmodplayer(void)
302 unsigned int i,c;
304 /* Calculate Amiga Period Values
305 * Start with Period 907 (= C-1 with Finetune -8) and work upwards */
306 double f = 907.0f;
307 /* Index 0 stands for no note (and therefore no period) */
308 modplayer.periodtable[0] = 0;
309 for (i=1;i<297;i++)
311 modplayer.periodtable[i] = (unsigned short) f;
312 f /= 1.0072464122237039; /* = pow(2.0f, 1.0f/(12.0f*8.0f)); */
316 * This is a more accurate but also time more consuming approach
317 * to calculate the amiga period table
318 * Commented out for speed purposes
319 const int finetuning = 8;
320 const int octaves = 3;
321 for (int halftone=0;halftone<=finetuning*octaves*12+7;halftone++)
323 float e = pow(2.0f, halftone/(12.0f*8.0f));
324 float f = 906.55f/e;
325 modplayer.periodtable[halfetone+1] = (int)(f+0.5f);
329 /* Calculate Protracker Vibrato sine table
330 * The routine makes use of the Harmonical Oscillator Approach
331 * for calculating sine tables
332 * (see http://membres.lycos.fr/amycoders/tutorials/sintables.html)
333 * The routine presented here calculates a complete sine wave
334 * with 64 values in range [-255,255]
336 float a, b, d, dd;
338 d = 0.09817475f; /* = 2*PI/64 */
339 dd = d*d;
340 a = 0;
341 b = d;
343 for (i=0;i<0x40;i++)
345 modplayer.sintable[i] = (int)(255*a);
347 a = a+b;
348 b = b-dd*a;
351 /* Set Default Player Values */
352 modplayer.currentline = 0;
353 modplayer.currenttick = 0;
354 modplayer.patterntableposition = 0;
355 modplayer.bpm = 125;
356 modplayer.ticksperline = 6;
357 modplayer.glissandoenabled = false; /* Disable glissando */
358 modplayer.amigafilterenabled = false; /* Disable the Amiga Filter */
360 /* Default Panning Values */
361 int panningvalues[8] = {4,12,12,4,4,12,12,4};
362 for (c=0;c<8;c++)
364 /* Set Default Panning */
365 mixer_setpanning(c, panningvalues[c]);
366 /* Reset channels in the MOD Player */
367 memset(&modplayer.modchannel[c], 0, sizeof(struct s_modchannel));
368 /* Don't play anything */
369 mixer.channel[c].channelactive = false;
374 /* Load the MOD File from memory */
375 STATICIRAM bool loadmod(void *modfile) ICODE_ATTR;
376 bool loadmod(void *modfile)
378 int i;
379 unsigned char *periodsconverted;
381 /* We don't support PowerPacker 2.0 Files */
382 if (memcmp((char*) modfile, "PP20", 4) == 0) return false;
384 /* Get the File Format Tag */
385 char *fileformattag = (char*)modfile + 1080;
387 /* Find out how many channels and instruments are used */
388 if (memcmp(fileformattag, "2CHN", 4) == 0)
389 {modsong.noofchannels = 2; modsong.noofinstruments = 31;}
390 else if (memcmp(fileformattag, "M.K.", 4) == 0)
391 {modsong.noofchannels = 4; modsong.noofinstruments = 31;}
392 else if (memcmp(fileformattag, "M!K!", 4) == 0)
393 {modsong.noofchannels = 4; modsong.noofinstruments = 31;}
394 else if (memcmp(fileformattag, "4CHN", 4) == 0)
395 {modsong.noofchannels = 4; modsong.noofinstruments = 31;}
396 else if (memcmp(fileformattag, "FLT4", 4) == 0)
397 {modsong.noofchannels = 4; modsong.noofinstruments = 31;}
398 else if (memcmp(fileformattag, "6CHN", 4) == 0)
399 {modsong.noofchannels = 6; modsong.noofinstruments = 31;}
400 else if (memcmp(fileformattag, "8CHN", 4) == 0)
401 {modsong.noofchannels = 8; modsong.noofinstruments = 31;}
402 else if (memcmp(fileformattag, "OKTA", 4) == 0)
403 {modsong.noofchannels = 8; modsong.noofinstruments = 31;}
404 else if (memcmp(fileformattag, "CD81", 4) == 0)
405 {modsong.noofchannels = 8; modsong.noofinstruments = 31;}
406 else {
407 /* The file has no format tag, so most likely soundtracker */
408 modsong.noofchannels = 4;
409 modsong.noofinstruments = 15;
412 /* Get the Song title
413 * Skipped here
414 * strncpy(modsong.szTitle, (char*)pMODFile, 20); */
416 /* Get the Instrument information */
417 for (i=0;i<modsong.noofinstruments;i++)
419 struct s_instrument *instrument = &modsong.instrument[i];
420 unsigned char *p = (unsigned char *)modfile + 20 + i*30;
422 /*strncpy(instrument->description, (char*)p, 22); */
423 p += 22;
424 instrument->length = (((p[0])<<8) + p[1]) << 1; p+=2;
425 instrument->finetune = *p++ & 0x0f;
426 /* Treat finetuning as signed nibble */
427 if (instrument->finetune > 7) instrument->finetune -= 16;
428 instrument->volume = *p++;
429 instrument->repeatoffset = (((p[0])<<8) + p[1]) << 1; p+= 2;
430 instrument->repeatlength = (((p[0])<<8) + p[1]) << 1;
433 /* Get the pattern information */
434 unsigned char *p = (unsigned char *)modfile + 20 +
435 modsong.noofinstruments*30;
436 modsong.songlength = *p++;
437 modsong.songendjumpposition = *p++;
438 modsong.patternordertable = p;
440 /* Find out how many patterns are used within this song */
441 int maxpatterns = 0;
442 for (i=0;i<128;i++)
443 if (modsong.patternordertable[i] > maxpatterns)
444 maxpatterns = modsong.patternordertable[i];
445 maxpatterns++;
447 /* use 'restartposition' (historically set to 127) which is not used here
448 as a marker that periods have already been converted */
450 periodsconverted = (char*)modfile + 20 + modsong.noofinstruments*30 + 1;
452 /* Get the pattern data; ST doesn't have fileformattag, so 4 bytes less */
453 modsong.patterndata = periodsconverted +
454 (modsong.noofinstruments==15 ? 129 : 133);
456 /* Convert the period values in the mod file to offsets
457 * in our periodtable (but only, if we haven't done this yet) */
458 p = (unsigned char *) modsong.patterndata;
459 if (*periodsconverted != 0xfe)
461 int note, note2, channel;
462 for (note=0;note<maxpatterns*64;note++)
463 for (channel=0;channel<modsong.noofchannels;channel++)
465 int period = ((p[0] & 0x0f) << 8) | p[1];
466 int periodoffset = 0;
468 /* Find the offset of the current period */
469 for (note2 = 1; note2 < 12*3+1; note2++)
470 if (abs(modplayer.periodtable[note2*8+1]-period) < 4)
472 periodoffset = note2*8+1;
473 break;
475 /* Write back the period offset */
476 p[0] = (periodoffset >> 8) | (p[0] & 0xf0);
477 p[1] = periodoffset & 0xff;
478 p += 4;
480 /* Remember that we already converted the periods,
481 * in case the file gets reloaded by rewinding
482 * with 0xfe (arbitary magic value > 127) */
483 *periodsconverted = 0xfe;
486 /* Get the samples
487 * Calculation: The Samples come after the pattern data
488 * We know that there are nMaxPatterns and each pattern requires
489 * 4 bytes per note and per channel.
490 * And of course there are always lines in each channel */
491 modsong.sampledata = (signed char*) modsong.patterndata +
492 maxpatterns*4*modsong.noofchannels*64;
493 int sampledataoffset = 0;
494 for (i=0;i<modsong.noofinstruments;i++)
496 modsong.instrument[i].sampledataoffset = sampledataoffset;
497 sampledataoffset += modsong.instrument[i].length;
500 return true;
503 /* Apply vibrato to channel */
504 STATICIRAM void vibrate(int channel) ICODE_ATTR;
505 void vibrate(int channel)
507 struct s_modchannel *p_modchannel = &modplayer.modchannel[channel];
509 /* Apply Vibrato
510 * >> 7 is used in the original protracker source code */
511 mixer_setamigaperiod(channel, p_modchannel->period+
512 ((p_modchannel->vibratodepth *
513 modplayer.sintable[p_modchannel->vibratosinpos])>>7));
515 /* Foward in Sine Table */
516 p_modchannel->vibratosinpos += p_modchannel->vibratospeed;
517 p_modchannel->vibratosinpos &= 0x3f;
520 /* Apply tremolo to channel
521 * (same as vibrato, but only apply on volume instead of pitch) */
522 STATICIRAM void tremolo(int channel) ICODE_ATTR;
523 void tremolo(int channel)
525 struct s_modchannel *p_modchannel = &modplayer.modchannel[channel];
527 /* Apply Tremolo
528 * >> 6 is used in the original protracker source code */
529 int volume = (p_modchannel->volume *
530 modplayer.sintable[p_modchannel->tremolosinpos])>>6;
531 if (volume > 64) volume = 64;
532 else if (volume < 0) volume = 0;
533 mixer_setvolume(channel, volume);
535 /* Foward in Sine Table */
536 p_modchannel->tremolosinpos += p_modchannel->tremolosinpos;
537 p_modchannel->tremolosinpos &= 0x3f;
540 /* Apply Slide to Note effect to channel */
541 STATICIRAM void slidetonote(int channel) ICODE_ATTR;
542 void slidetonote(int channel)
544 struct s_modchannel *p_modchannel = &modplayer.modchannel[channel];
546 /* If there hasn't been any slide-to note set up, then return */
547 if (p_modchannel->slidetonoteperiod == 0) return;
549 /* Slide note up */
550 if (p_modchannel->slidetonoteperiod > p_modchannel->period)
552 p_modchannel->period += p_modchannel->slidetonotespeed;
553 if (p_modchannel->period > p_modchannel->slidetonoteperiod)
554 p_modchannel->period = p_modchannel->slidetonoteperiod;
556 /* Slide note down */
557 else if (p_modchannel->slidetonoteperiod < p_modchannel->period)
559 p_modchannel->period -= p_modchannel->slidetonotespeed;
560 if (p_modchannel->period < p_modchannel->slidetonoteperiod)
561 p_modchannel->period = p_modchannel->slidetonoteperiod;
563 mixer_setamigaperiod(channel, p_modchannel->period);
566 /* Apply Slide to Note effect on channel,
567 * but this time with glissando enabled */
568 STATICIRAM void slidetonoteglissando(int channel) ICODE_ATTR;
569 void slidetonoteglissando(int channel)
571 struct s_modchannel *p_modchannel = &modplayer.modchannel[channel];
573 /* Slide note up */
574 if (p_modchannel->slidetonoteperiod > p_modchannel->period)
576 p_modchannel->period =
577 modplayer.periodtable[p_modchannel->periodtableoffset+=8];
578 if (p_modchannel->period > p_modchannel->slidetonoteperiod)
579 p_modchannel->period = p_modchannel->slidetonoteperiod;
581 /* Slide note down */
582 else
584 p_modchannel->period =
585 modplayer.periodtable[p_modchannel->periodtableoffset-=8];
586 if (p_modchannel->period < p_modchannel->slidetonoteperiod)
587 p_modchannel->period = p_modchannel->slidetonoteperiod;
589 mixer_setamigaperiod(channel, p_modchannel->period);
592 /* Apply Volume Slide */
593 STATICIRAM void volumeslide(int channel, int effectx, int effecty) ICODE_ATTR;
594 void volumeslide(int channel, int effectx, int effecty)
596 struct s_modchannel *p_modchannel = &modplayer.modchannel[channel];
598 /* If both X and Y Parameters are non-zero, then the y value is ignored */
599 if (effectx > 0) {
600 p_modchannel->volume += effectx;
601 if (p_modchannel->volume > 64) p_modchannel->volume = 64;
603 else {
604 p_modchannel->volume -= effecty;
605 if (p_modchannel->volume < 0) p_modchannel->volume = 0;
608 mixer_setvolume(channel, p_modchannel->volume);
611 /* Play the current line (at tick 0) */
612 STATICIRAM void playline(int pattern, int line) ICODE_ATTR;
613 void playline(int pattern, int line)
615 int c;
617 /* Get pointer to the current pattern */
618 unsigned char *p_line = (unsigned char*)modsong.patterndata;
619 p_line += pattern*64*4*modsong.noofchannels;
620 p_line += line*4*modsong.noofchannels;
622 /* Only allow one Patternbreak Commando per Line */
623 bool patternbreakdone = false;
625 for (c=0;c<modsong.noofchannels;c++)
627 struct s_modchannel *p_modchannel = &modplayer.modchannel[c];
628 unsigned char *p_note = p_line + c*4;
629 unsigned char samplenumber = (p_note[0] & 0xf0) | (p_note[2] >> 4);
630 short periodtableoffset = ((p_note[0] & 0x0f) << 8) | p_note[1];
632 p_modchannel->effect = p_note[2] & 0x0f;
633 p_modchannel->effectparameter = p_note[3];
635 /* Remember Instrument and set Volume if new Instrument triggered */
636 if (samplenumber > 0)
638 /* And trigger new sample, if new instrument was set */
639 if (samplenumber-1 != p_modchannel->instrument)
641 /* Advance the new sample to the same offset
642 * the old sample was beeing played */
643 int oldsampleoffset = mixer.channel[c].samplepos -
644 modsong.instrument[
645 p_modchannel->instrument].sampledataoffset;
646 mixer_playsample(c, samplenumber-1);
647 mixer.channel[c].samplepos += oldsampleoffset;
650 /* Remember last played instrument on channel */
651 p_modchannel->instrument = samplenumber-1;
653 /* Set Volume to standard instrument volume,
654 * if not overwritten by volume effect */
655 if (p_modchannel->effect != 0x0c)
657 p_modchannel->volume = modsong.instrument[
658 p_modchannel->instrument].volume;
659 mixer_setvolume(c, p_modchannel->volume);
662 /* Trigger new sample if note available */
663 if (periodtableoffset > 0)
665 /* Restart instrument only when new sample triggered */
666 if (samplenumber != 0)
667 mixer_playsample(c, (samplenumber > 0) ?
668 samplenumber-1 : p_modchannel->instrument);
670 /* Set the new amiga period
671 * (but only, if there is no slide to note effect) */
672 if ((p_modchannel->effect != 0x3) &&
673 (p_modchannel->effect != 0x5))
675 /* Apply finetuning to sample */
676 p_modchannel->periodtableoffset = periodtableoffset +
677 modsong.instrument[p_modchannel->instrument].finetune;
678 p_modchannel->period = modplayer.periodtable[
679 p_modchannel->periodtableoffset];
680 mixer_setamigaperiod(c, p_modchannel->period);
681 /* When a new note is played without slide to note setup,
682 * then disable slide to note */
683 modplayer.modchannel[c].slidetonoteperiod =
684 p_modchannel->period;
687 int effectx = p_modchannel->effectparameter>>4;
688 int effecty = p_modchannel->effectparameter&0x0f;
690 switch (p_modchannel->effect)
692 /* Effect 0: Arpeggio */
693 case 0x00:
694 /* Set the base period on tick 0 */
695 if (p_modchannel->effectparameter > 0)
696 mixer_setamigaperiod(c,
697 modplayer.periodtable[
698 p_modchannel->periodtableoffset]);
699 break;
700 /* Slide up (Portamento up) */
701 case 0x01:
702 if (p_modchannel->effectparameter > 0)
703 p_modchannel->slideupspeed =
704 p_modchannel->effectparameter;
705 break;
707 /* Slide down (Portamento down) */
708 case 0x02:
709 if (p_modchannel->effectparameter > 0)
710 p_modchannel->slidedownspeed =
711 p_modchannel->effectparameter;
712 break;
714 /* Slide to Note */
715 case 0x03:
716 if (p_modchannel->effectparameter > 0)
717 p_modchannel->slidetonotespeed =
718 p_modchannel->effectparameter;
719 /* Get the slide to note directly from the pattern buffer */
720 if (periodtableoffset > 0)
721 p_modchannel->slidetonoteperiod =
722 modplayer.periodtable[periodtableoffset +
723 modsong.instrument[
724 p_modchannel->instrument].finetune];
725 /* If glissando is enabled apply the effect directly here */
726 if (modplayer.glissandoenabled)
727 slidetonoteglissando(c);
728 break;
730 /* Set Vibrato */
731 case 0x04:
732 if (effectx > 0) p_modchannel->vibratospeed = effectx;
733 if (effecty > 0) p_modchannel->vibratodepth = effecty;
734 break;
736 /* Effect 0x06: Slide to note */
737 case 0x05:
738 /* Get the slide to note directly from the pattern buffer */
739 if (periodtableoffset > 0)
740 p_modchannel->slidetonoteperiod =
741 modplayer.periodtable[periodtableoffset +
742 modsong.instrument[
743 p_modchannel->instrument].finetune];
744 break;
746 /* Effect 0x06 is "Continue Effects" */
747 /* It is not processed on tick 0 */
748 case 0x06:
749 break;
751 /* Set Tremolo */
752 case 0x07:
753 if (effectx > 0) p_modchannel->tremolodepth = effectx;
754 if (effecty > 0) p_modchannel->tremolospeed = effecty;
755 break;
757 /* Set fine panning */
758 case 0x08:
759 /* Internal panning goes from 0..15
760 * Scale the fine panning value to that range */
761 mixer.channel[c].panning = p_modchannel->effectparameter>>4;
762 break;
764 /* Set Sample Offset */
765 case 0x09:
767 struct s_instrument *p_instrument =
768 &modsong.instrument[p_modchannel->instrument];
769 int sampleoffset = p_instrument->sampledataoffset;
770 if (sampleoffset > p_instrument->length)
771 sampleoffset = p_instrument->length;
772 /* Forward the new offset to the mixer */
773 mixer.channel[c].samplepos =
774 p_instrument->sampledataoffset +
775 (p_modchannel->effectparameter<<8);
776 mixer.channel[c].samplefractpos = 0;
777 break;
780 /* Effect 0x0a (Volume slide) is not processed on tick 0 */
782 /* Position Jump */
783 case 0x0b:
784 modplayer.currentline = -1;
785 modplayer.patterntableposition = (effectx<<4)+effecty;
786 break;
788 /* Set Volume */
789 case 0x0c:
790 p_modchannel->volume = p_modchannel->effectparameter;
791 mixer_setvolume(c, p_modchannel->volume);
792 break;
794 /* Pattern break */
795 case 0x0d:
796 modplayer.currentline = effectx*10 + effecty - 1;
797 if (!patternbreakdone)
799 patternbreakdone = true;
800 modplayer.patterntableposition++;
802 break;
804 /* Extended Effects */
805 case 0x0e:
806 switch (effectx)
808 /* Set Filter */
809 case 0x0:
810 modplayer.amigafilterenabled = (effecty == 0);
811 break;
812 /* Fineslide up */
813 case 0x1:
814 mixer_setamigaperiod(c, p_modchannel->period -=
815 effecty);
816 if (p_modchannel->period <
817 modplayer.periodtable[37*8]) p_modchannel->period = 100;
818 /* Find out the new offset in the period table */
819 if (p_modchannel->periodtableoffset < 36*8)
820 while (modplayer.periodtable[
821 p_modchannel->periodtableoffset+8] >= p_modchannel->period)
822 p_modchannel->periodtableoffset+=8;
823 break;
824 /* Fineslide down */
825 case 0x2:
826 mixer_setamigaperiod(c,
827 p_modchannel->period += effecty);
828 if (p_modchannel->periodtableoffset > 8)
829 while (modplayer.periodtable[
830 p_modchannel->periodtableoffset-8]
831 <= p_modchannel->period)
832 p_modchannel->periodtableoffset-=8;
833 break;
834 /* Set glissando on/off */
835 case 0x3:
836 modplayer.glissandoenabled = (effecty > 0);
837 break;
838 /* Set Vibrato waveform */
839 case 0x4:
840 /* Currently not implemented */
841 break;
842 /* Set Finetune value */
843 case 0x5:
844 /* Treat as signed nibble */
845 if (effecty > 7) effecty -= 16;
847 p_modchannel->periodtableoffset +=
848 effecty -
849 modsong.instrument[
850 p_modchannel->instrument].finetune;
851 p_modchannel->period =
852 modplayer.periodtable[
853 p_modchannel->periodtableoffset];
854 modsong.instrument[
855 p_modchannel->instrument].finetune = effecty;
856 break;
857 /* Pattern loop */
858 case 0x6:
859 if (effecty == 0)
860 modplayer.loopstartline = line-1;
861 else
863 if (modplayer.looptimes == 0)
865 modplayer.currentline =
866 modplayer.loopstartline;
867 modplayer.looptimes = effecty;
869 else modplayer.looptimes--;
870 if (modplayer.looptimes > 0)
871 modplayer.currentline =
872 modplayer.loopstartline;
874 break;
875 /* Set Tremolo waveform */
876 case 0x7:
877 /* Not yet implemented */
878 break;
879 /* Enhanced Effect 8 is not used */
880 case 0x8:
881 break;
882 /* Retrigger sample */
883 case 0x9:
884 /* Only processed on subsequent ticks */
885 break;
886 /* Fine volume slide up */
887 case 0xa:
888 p_modchannel->volume += effecty;
889 if (p_modchannel->volume > 64)
890 p_modchannel->volume = 64;
891 mixer_setvolume(c, p_modchannel->volume);
892 break;
893 /* Fine volume slide down */
894 case 0xb:
895 p_modchannel->volume -= effecty;
896 if (p_modchannel->volume < 0)
897 p_modchannel->volume = 0;
898 mixer_setvolume(c, p_modchannel->volume);
899 break;
900 /* Cut sample */
901 case 0xc:
902 /* Continue sample */
903 mixer_continuesample(c);
904 break;
905 /* Note delay (Usage: $ED + ticks to delay note.) */
906 case 0xd:
907 /* We stop the sample here on tick 0
908 * and restart it later in the effect */
909 if (effecty > 0)
910 mixer.channel[c].channelactive = false;
911 break;
913 break;
915 /* Set Speed */
916 case 0x0f:
917 if (p_modchannel->effectparameter < 32)
918 modplayer.ticksperline = p_modchannel->effectparameter;
919 else
920 modplayer.bpm = p_modchannel->effectparameter;
921 break;
926 /* Play the current effect of the note (ticks 1..speed) */
927 STATICIRAM void playeffect(int currenttick) ICODE_ATTR;
928 void playeffect(int currenttick)
930 int c;
932 for (c=0;c<modsong.noofchannels;c++)
934 struct s_modchannel *p_modchannel = &modplayer.modchannel[c];
936 /* If there is no note active then there are no effects to play */
937 if (p_modchannel->period == 0) continue;
939 unsigned char effectx = p_modchannel->effectparameter>>4;
940 unsigned char effecty = p_modchannel->effectparameter&0x0f;
942 switch (p_modchannel->effect)
944 /* Effect 0: Arpeggio */
945 case 0x00:
946 if (p_modchannel->effectparameter > 0)
948 unsigned short newperiodtableoffset;
949 switch (currenttick % 3)
951 case 0:
952 mixer_setamigaperiod(c,
953 modplayer.periodtable[
954 p_modchannel->periodtableoffset]);
955 break;
956 case 1:
957 newperiodtableoffset =
958 p_modchannel->periodtableoffset+(effectx<<3);
959 if (newperiodtableoffset < 37*8)
960 mixer_setamigaperiod(c,
961 modplayer.periodtable[
962 newperiodtableoffset]);
963 break;
964 case 2:
965 newperiodtableoffset =
966 p_modchannel->periodtableoffset+(effecty<<3);
967 if (newperiodtableoffset < 37*8)
968 mixer_setamigaperiod(c,
969 modplayer.periodtable[
970 newperiodtableoffset]);
971 break;
974 break;
976 /* Effect 1: Slide Up */
977 case 0x01:
978 mixer_setamigaperiod(c,
979 p_modchannel->period -= p_modchannel->slideupspeed);
980 /* Find out the new offset in the period table */
981 if (p_modchannel->periodtableoffset <= 37*8)
982 while (modplayer.periodtable[
983 p_modchannel->periodtableoffset] >
984 p_modchannel->period)
986 p_modchannel->periodtableoffset++;
987 /* Make sure we don't go out of range */
988 if (p_modchannel->periodtableoffset > 37*8)
990 p_modchannel->periodtableoffset = 37*8;
991 break;
994 break;
996 /* Effect 2: Slide Down */
997 case 0x02:
998 mixer_setamigaperiod(c, p_modchannel->period +=
999 p_modchannel->slidedownspeed);
1000 /* Find out the new offset in the period table */
1001 if (p_modchannel->periodtableoffset > 8)
1002 while (modplayer.periodtable[
1003 p_modchannel->periodtableoffset] <
1004 p_modchannel->period)
1006 p_modchannel->periodtableoffset--;
1007 /* Make sure we don't go out of range */
1008 if (p_modchannel->periodtableoffset < 1)
1010 p_modchannel->periodtableoffset = 1;
1011 break;
1014 break;
1016 /* Effect 3: Slide to Note */
1017 case 0x03:
1018 /* Apply smooth sliding, if no glissando is enabled */
1019 if (modplayer.glissandoenabled == 0)
1020 slidetonote(c);
1021 break;
1023 /* Effect 4: Vibrato */
1024 case 0x04:
1025 vibrate(c);
1026 break;
1028 /* Effect 5: Continue effect 3:'Slide to note',
1029 * but also do Volume slide */
1030 case 0x05:
1031 slidetonote(c);
1032 volumeslide(c, effectx, effecty);
1033 break;
1035 /* Effect 6: Continue effect 4:'Vibrato',
1036 * but also do Volume slide */
1037 case 0x06:
1038 vibrate(c);
1039 volumeslide(c, effectx, effecty);
1040 break;
1042 /* Effect 7: Tremolo */
1043 case 0x07:
1044 tremolo(c);
1045 break;
1047 /* Effect 8 (Set fine panning) is only processed at tick 0 */
1048 /* Effect 9 (Set sample offset) is only processed at tick 0 */
1050 /* Effect A: Volume slide */
1051 case 0x0a:
1052 volumeslide(c, effectx, effecty);
1053 break;
1055 /* Effect B (Position jump) is only processed at tick 0 */
1056 /* Effect C (Set Volume) is only processed at tick 0 */
1057 /* Effect D (Pattern Preak) is only processed at tick 0 */
1058 /* Effect E (Enhanced Effect) */
1059 case 0x0e:
1060 switch (effectx)
1062 /* Retrigger sample ($E9 + Tick to Retrig note at) */
1063 case 0x9:
1064 /* Don't device by zero */
1065 if (effecty == 0) effecty = 1;
1066 /* Apply retrig */
1067 if (currenttick % effecty == 0)
1068 mixer_playsample(c, p_modchannel->instrument);
1069 break;
1070 /* Cut note (Usage: $EC + Tick to Cut note at) */
1071 case 0xc:
1072 if (currenttick == effecty)
1073 mixer_stopsample(c);
1074 break;
1075 /* Delay note (Usage: $ED + ticks to delay note) */
1076 case 0xd:
1077 /* If this is the correct tick,
1078 * we start playing the sample now */
1079 if (currenttick == effecty)
1080 mixer.channel[c].channelactive = true;
1081 break;
1084 break;
1085 /* Effect F (Set Speed) is only processed at tick 0 */
1091 static inline int clip(int i)
1093 if (i > 32767) return(32767);
1094 else if (i < -32768) return(-32768);
1095 else return(i);
1098 STATICIRAM void synthrender(int32_t *renderbuffer, int samplecount) ICODE_ATTR;
1099 void synthrender(int32_t *renderbuffer, int samplecount)
1101 /* 125bpm equals to 50Hz (= 0.02s)
1102 * => one tick = mixingrate/50,
1103 * samples passing in one tick:
1104 * mixingrate/(bpm/2.5) = 2.5*mixingrate/bpm */
1106 int32_t *p_left = renderbuffer; /* int in rockbox */
1107 int32_t *p_right = p_left+1;
1108 signed short s;
1109 int qf_distance, qf_distance2;
1111 int i;
1113 int c, left, right;
1115 for (i=0;i<samplecount;i++)
1117 /* New Tick? */
1118 if ((modplayer.samplespertick-- <= 0) &&
1119 (modplayer.patterntableposition < 127))
1121 if (modplayer.currenttick == 0)
1122 playline(modsong.patternordertable[
1123 modplayer.patterntableposition], modplayer.currentline);
1124 else playeffect(modplayer.currenttick);
1126 modplayer.currenttick++;
1128 if (modplayer.currenttick >= modplayer.ticksperline)
1130 modplayer.currentline++;
1131 modplayer.currenttick = 0;
1132 if (modplayer.currentline == 64)
1134 modplayer.patterntableposition++;
1135 if (modplayer.patterntableposition >= modsong.songlength)
1136 /* This is for Noise Tracker
1137 * modplayer.patterntableposition =
1138 * modsong.songendjumpposition;
1139 * More compatible approach is restart from 0 */
1140 modplayer.patterntableposition=0;
1141 modplayer.currentline = 0;
1145 modplayer.samplespertick = (20*mixingrate/modplayer.bpm)>>3;
1147 /* Mix buffers from here
1148 * Walk through all channels */
1149 left=0, right=0;
1151 /* If song has not stopped playing */
1152 if (modplayer.patterntableposition < 127)
1153 /* Loop through all channels */
1154 for (c=0;c<modsong.noofchannels;c++)
1156 /* Only mix the sample,
1157 * if channel there is something played on the channel */
1158 if (!mixer.channel[c].channelactive) continue;
1160 /* Loop the sample, if requested? */
1161 if (mixer.channel[c].samplepos >= mixer.channel[c].loopend)
1163 if (mixer.channel[c].loopsample)
1164 mixer.channel[c].samplepos -=
1165 (mixer.channel[c].loopend-
1166 mixer.channel[c].loopstart);
1167 else mixer.channel[c].channelactive = false;
1170 /* If the sample has stopped playing don't mix it */
1171 if (!mixer.channel[c].channelactive) continue;
1173 /* Get the sample */
1174 s = (signed short)(modsong.sampledata[
1175 mixer.channel[c].samplepos]*mixer.channel[c].volume);
1177 /* Interpolate if the sample-frequency is lower
1178 * than the mixing rate
1179 * If you don't want interpolation simply skip this part */
1180 if (mixer.channel[c].frequency < mixingrate)
1182 /* Low precision linear interpolation
1183 * (fast integer based) */
1184 qf_distance = mixer.channel[c].samplefractpos<<16 /
1185 mixingrate;
1186 qf_distance2 = (1<<16)-qf_distance;
1187 s = (qf_distance*s + qf_distance2*
1188 mixer.channel[c].lastsampledata)>>16;
1191 /* Save the last played sample for interpolation purposes */
1192 mixer.channel[c].lastsampledata = s;
1194 /* Pan the sample */
1195 left += s*(16-mixer.channel[c].panning)>>3;
1196 right += s*mixer.channel[c].panning>>3;
1198 /* Advance sample */
1199 mixer.channel[c].samplefractpos += mixer.channel[c].frequency;
1200 while (mixer.channel[c].samplefractpos > mixingrate)
1202 mixer.channel[c].samplefractpos -= mixingrate;
1203 mixer.channel[c].samplepos++;
1206 /* If we have more than 4 channels
1207 * we have to make sure that we apply clipping */
1208 if (modsong.noofchannels > 4) {
1209 *p_left = clip(left)<<13;
1210 *p_right = clip(right)<<13;
1212 else {
1213 *p_left = left<<13;
1214 *p_right = right<<13;
1216 p_left+=2;
1217 p_right+=2;
1222 enum codec_status codec_main(void)
1224 size_t n;
1225 unsigned char *modfile;
1226 int old_patterntableposition;
1228 int bytesdone;
1230 next_track:
1231 if (codec_init()) {
1232 return CODEC_ERROR;
1235 while (!*ci->taginfo_ready && !ci->stop_codec)
1236 ci->sleep(1);
1238 codec_set_replaygain(ci->id3);
1240 /* Load MOD file */
1242 * This is the save way
1244 size_t bytesfree;
1245 unsigned int filesize;
1247 p = modfile;
1248 bytesfree=sizeof(modfile);
1249 while ((n = ci->read_filebuf(p, bytesfree)) > 0) {
1250 p += n;
1251 bytesfree -= n;
1252 if (bytesfree == 0)
1253 return CODEC_ERROR;
1255 filesize = p-modfile;
1257 if (filesize == 0)
1258 return CODEC_ERROR;
1261 /* Directly use mod in buffer */
1262 ci->seek_buffer(0);
1263 modfile = ci->request_buffer(&n, ci->filesize);
1264 if (!modfile || n < (size_t)ci->filesize) {
1265 return CODEC_ERROR;
1268 initmodplayer();
1269 loadmod(modfile);
1271 /* Make use of 44.1khz */
1272 ci->configure(DSP_SET_FREQUENCY, 44100);
1273 /* Sample depth is 28 bit host endian */
1274 ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
1275 /* Stereo output */
1276 ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
1278 /* The main decoder loop */
1279 ci->set_elapsed(0);
1280 bytesdone = 0;
1281 old_patterntableposition = 0;
1283 while (1) {
1284 ci->yield();
1285 if (ci->stop_codec || ci->new_track)
1286 break;
1288 if (ci->seek_time) {
1289 /* New time is ready in ci->seek_time */
1290 modplayer.patterntableposition = ci->seek_time/1000;
1291 modplayer.currentline = 0;
1292 ci->seek_complete();
1295 if(old_patterntableposition != modplayer.patterntableposition) {
1296 ci->set_elapsed(modplayer.patterntableposition*1000+500);
1297 old_patterntableposition=modplayer.patterntableposition;
1300 synthrender(samples, CHUNK_SIZE/2);
1302 bytesdone += CHUNK_SIZE;
1304 ci->pcmbuf_insert(samples, NULL, CHUNK_SIZE/2);
1308 if (ci->request_next_track())
1309 goto next_track;
1311 return CODEC_OK;