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rockbox/apps/codecs/mod.c
Daniel Stenberg 2acc0ac542 Updated our source code header to explicitly mention that we are GPL v2 or 
later. We still need to hunt down snippets used that are not. 1324 modified
files...
http://www.rockbox.org/mail/archive/rockbox-dev-archive-2008-06/0060.shtml


git-svn-id: svn://svn.rockbox.org/rockbox/trunk@17847 a1c6a512-1295-4272-9138-f99709370657
2008-06-28 18:10:04 +00:00

1317 lines
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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* MOD Codec for rockbox
*
* Written from scratch by Rainer Sinsch
* exclusivly for Rockbox in February 2008
*
* 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 software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
/**************
* This version supports large files directly from internal memory management.
* There is a drawback however: It may happen that a song is not completely
* loaded when the internal rockbox-ringbuffer (approx. 28MB) is filled up
* As a workaround make sure you don't have directories with mods larger
* than a total of 28MB
*************/
#include "debug.h"
#include "codeclib.h"
#include <inttypes.h>
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include <ctype.h>
CODEC_HEADER
#define CHUNK_SIZE (1024*2)
/* This codec supports MOD Files:
*
*/
static int32_t samples[CHUNK_SIZE] IBSS_ATTR; /* The sample buffer */
/* Instrument Data */
struct s_instrument {
/* Sample name / description */
/*char description[22];*/
/* Sample length in bytes */
unsigned short length;
/* Sample finetuning (-8 - +7) */
signed char finetune;
/* Sample volume (0 - 64) */
signed char volume;
/* Sample Repeat Position */
unsigned short repeatoffset;
/* Sample Repeat Length */
unsigned short repeatlength;
/* Offset to sample data */
unsigned int sampledataoffset;
};
/* Song Data */
struct s_song {
/* Song name / title description */
/*char szTitle[20];*/
/* No. of channels in song */
unsigned char noofchannels;
/* No. of instruments used (either 15 or 31) */
unsigned char noofinstruments;
/* How many patterns are beeing played? */
unsigned char songlength;
/* Where to jump after the song end? */
unsigned char songendjumpposition;
/* Pointer to the Pattern Order Table */
unsigned char *patternordertable;
/* Pointer to the pattern data */
void *patterndata;
/* Pointer to the sample buffer */
signed char *sampledata;
/* Instrument data */
struct s_instrument instrument[31];
};
struct s_modchannel {
/* Current Volume */
signed char volume;
/* Current Offset to period in PeriodTable of notebeeing played
(can be temporarily negative) */
short periodtableoffset;
/* Current Period beeing played */
short period;
/* Current effect */
unsigned char effect;
/* Current parameters of effect */
unsigned char effectparameter;
/* Current Instrument beeing played */
unsigned char instrument;
/* Current Vibrato Speed */
unsigned char vibratospeed;
/* Current Vibrato Depth */
unsigned char vibratodepth;
/* Current Position for Vibrato in SinTable */
unsigned char vibratosinpos;
/* Current Tremolo Speed */
unsigned char tremolospeed;
/* Current Tremolo Depth */
unsigned char tremolodepth;
/* Current Position for Tremolo in SinTable */
unsigned char tremolosinpos;
/* Current Speed of Effect "Slide Note up" */
unsigned char slideupspeed;
/* Current Speed of Effect "Slide Note down" */
unsigned char slidedownspeed;
/* Current Speed of the "Slide to Note" effect */
unsigned char slidetonotespeed;
/* Current Period of the "Slide to Note" effect */
unsigned short slidetonoteperiod;
};
struct s_modplayer {
/* Ticks per Line */
unsigned char ticksperline;
/* Beats per Minute */
unsigned char bpm;
/* Position of the Song in the Pattern Table (0-127) */
unsigned char patterntableposition;
/* Current Line (may be temporarily < 0) */
signed char currentline;
/* Current Tick */
signed char currenttick;
/* How many samples are required to calculate for each tick? */
unsigned int samplespertick;
/* Information about the channels */
struct s_modchannel modchannel[8];
/* The Amiga Period Table
(+1 because we use index 0 for period 0 = no new note) */
unsigned short periodtable[37*8+1];
/* The sinus table [-255,255] */
signed short sintable[0x40];
/* Is the glissando effect enabled? */
bool glissandoenabled;
/* Is the Amiga Filter enabled? */
bool amigafilterenabled;
/* The pattern-line where the loop is carried out (set with e6 command) */
unsigned char loopstartline;
/* Number of times to loop */
unsigned char looptimes;
};
struct s_channel {
/* Panning (0 = left, 16 = right) */
unsigned char panning;
/* Sample frequency of the channel */
unsigned short frequency;
/* Position of the sample currently played */
unsigned int samplepos;
/* Fractual Position of the sample currently player */
unsigned int samplefractpos;
/* Loop Sample */
bool loopsample;
/* Loop Position Start */
unsigned int loopstart;
/* Loop Position End */
unsigned int loopend;
/* Is The channel beeing played? */
bool channelactive;
/* The Volume (0..64) */
signed char volume;
/* The last sampledata beeing played (required for interpolation) */
signed short lastsampledata;
};
struct s_mixer {
/* The channels */
struct s_channel channel[32];
};
struct s_song modsong IDATA_ATTR; /* The Song */
struct s_modplayer modplayer IDATA_ATTR; /* The Module Player */
struct s_mixer mixer IDATA_ATTR;
const unsigned short mixingrate = 44100;
STATICIRAM void mixer_playsample(int channel, int instrument) ICODE_ATTR;
void mixer_playsample(int channel, int instrument)
{
struct s_channel *p_channel = &mixer.channel[channel];
struct s_instrument *p_instrument = &modsong.instrument[instrument];
p_channel->channelactive = true;
p_channel->samplepos = p_instrument->sampledataoffset;
p_channel->samplefractpos = 0;
p_channel->loopsample = (p_instrument->repeatlength > 2) ? true : false;
if (p_channel->loopsample) {
p_channel->loopstart = p_instrument->repeatoffset +
p_instrument->sampledataoffset;
p_channel->loopend = p_channel->loopstart +
p_instrument->repeatlength;
}
else p_channel->loopend = p_instrument->length +
p_instrument->sampledataoffset;
/* Remember the instrument */
modplayer.modchannel[channel].instrument = instrument;
}
inline void mixer_stopsample(int channel)
{
mixer.channel[channel].channelactive = false;
}
inline void mixer_continuesample(int channel)
{
mixer.channel[channel].channelactive = true;
}
inline void mixer_setvolume(int channel, int volume)
{
mixer.channel[channel].volume = volume;
}
inline void mixer_setpanning(int channel, int panning)
{
mixer.channel[channel].panning = panning;
}
inline void mixer_setamigaperiod(int channel, int amigaperiod)
{
/* Just to make sure we don't devide by zero
* amigaperiod shouldn't 0 anyway - if it is the case
* then something terribly went wrong */
if (amigaperiod == 0)
return;
mixer.channel[channel].frequency = 3579546 / amigaperiod;
}
/* Initialize the MOD Player with default values and precalc tables */
STATICIRAM void initmodplayer(void) ICODE_ATTR;
void initmodplayer(void)
{
unsigned int i,c;
/* Calculate Amiga Period Values
* Start with Period 907 (= C-1 with Finetune -8) and work upwards */
double f = 907.0f;
/* Index 0 stands for no note (and therefore no period) */
modplayer.periodtable[0] = 0;
for (i=1;i<297;i++)
{
modplayer.periodtable[i] = (unsigned short) f;
f /= 1.0072464122237039; /* = pow(2.0f, 1.0f/(12.0f*8.0f)); */
}
/*
* This is a more accurate but also time more consuming approach
* to calculate the amiga period table
* Commented out for speed purposes
const int finetuning = 8;
const int octaves = 3;
for (int halftone=0;halftone<=finetuning*octaves*12+7;halftone++)
{
float e = pow(2.0f, halftone/(12.0f*8.0f));
float f = 906.55f/e;
modplayer.periodtable[halfetone+1] = (int)(f+0.5f);
}
*/
/* Calculate Protracker Vibrato sine table
* The routine makes use of the Harmonical Oscillator Approach
* for calculating sine tables
* (see http://membres.lycos.fr/amycoders/tutorials/sintables.html)
* The routine presented here calculates a complete sine wave
* with 64 values in range [-255,255]
*/
float a, b, d, dd;
d = 0.09817475f; /* = 2*PI/64 */
dd = d*d;
a = 0;
b = d;
for (i=0;i<0x40;i++)
{
modplayer.sintable[i] = (int)(255*a);
a = a+b;
b = b-dd*a;
}
/* Set Default Player Values */
modplayer.currentline = 0;
modplayer.currenttick = 0;
modplayer.patterntableposition = 0;
modplayer.bpm = 125;
modplayer.ticksperline = 6;
modplayer.glissandoenabled = false; /* Disable glissando */
modplayer.amigafilterenabled = false; /* Disable the Amiga Filter */
/* Default Panning Values */
int panningvalues[8] = {4,12,12,4,4,12,12,4};
for (c=0;c<8;c++)
{
/* Set Default Panning */
mixer_setpanning(c, panningvalues[c]);
/* Reset channels in the MOD Player */
memset(&modplayer.modchannel[c], 0, sizeof(struct s_modchannel));
/* Don't play anything */
mixer.channel[c].channelactive = false;
}
}
/* Load the MOD File from memory */
STATICIRAM bool loadmod(void *modfile) ICODE_ATTR;
bool loadmod(void *modfile)
{
int i;
unsigned char *periodsconverted;
/* We don't support PowerPacker 2.0 Files */
if (memcmp((char*) modfile, "PP20", 4) == 0) return false;
/* Get the File Format Tag */
char *fileformattag = (char*)modfile + 1080;
/* Find out how many channels and instruments are used */
if (memcmp(fileformattag, "2CHN", 4) == 0)
{modsong.noofchannels = 2; modsong.noofinstruments = 31;}
else if (memcmp(fileformattag, "M.K.", 4) == 0)
{modsong.noofchannels = 4; modsong.noofinstruments = 31;}
else if (memcmp(fileformattag, "M!K!", 4) == 0)
{modsong.noofchannels = 4; modsong.noofinstruments = 31;}
else if (memcmp(fileformattag, "4CHN", 4) == 0)
{modsong.noofchannels = 4; modsong.noofinstruments = 31;}
else if (memcmp(fileformattag, "FLT4", 4) == 0)
{modsong.noofchannels = 4; modsong.noofinstruments = 31;}
else if (memcmp(fileformattag, "6CHN", 4) == 0)
{modsong.noofchannels = 6; modsong.noofinstruments = 31;}
else if (memcmp(fileformattag, "8CHN", 4) == 0)
{modsong.noofchannels = 8; modsong.noofinstruments = 31;}
else if (memcmp(fileformattag, "OKTA", 4) == 0)
{modsong.noofchannels = 8; modsong.noofinstruments = 31;}
else if (memcmp(fileformattag, "CD81", 4) == 0)
{modsong.noofchannels = 8; modsong.noofinstruments = 31;}
else {
/* The file has no format tag, so most likely soundtracker */
modsong.noofchannels = 4;
modsong.noofinstruments = 15;
}
/* Get the Song title
* Skipped here
* strncpy(modsong.szTitle, (char*)pMODFile, 20); */
/* Get the Instrument information */
for (i=0;i<modsong.noofinstruments;i++)
{
struct s_instrument *instrument = &modsong.instrument[i];
unsigned char *p = (unsigned char *)modfile + 20 + i*30;
/*strncpy(instrument->description, (char*)p, 22); */
p += 22;
instrument->length = (((p[0])<<8) + p[1]) << 1; p+=2;
instrument->finetune = *p++ & 0x0f;
/* Treat finetuning as signed nibble */
if (instrument->finetune > 7) instrument->finetune -= 16;
instrument->volume = *p++;
instrument->repeatoffset = (((p[0])<<8) + p[1]) << 1; p+= 2;
instrument->repeatlength = (((p[0])<<8) + p[1]) << 1;
}
/* Get the pattern information */
unsigned char *p = (unsigned char *)modfile + 20 +
modsong.noofinstruments*30;
modsong.songlength = *p++;
modsong.songendjumpposition = *p++;
modsong.patternordertable = p;
/* Find out how many patterns are used within this song */
int maxpatterns = 0;
for (i=0;i<128;i++)
if (modsong.patternordertable[i] > maxpatterns)
maxpatterns = modsong.patternordertable[i];
maxpatterns++;
/* use 'restartposition' (historically set to 127) which is not used here
as a marker that periods have already been converted */
periodsconverted = (char*)modfile + 20 + modsong.noofinstruments*30 + 1;
/* Get the pattern data; ST doesn't have fileformattag, so 4 bytes less */
modsong.patterndata = periodsconverted +
(modsong.noofinstruments==15 ? 129 : 133);
/* Convert the period values in the mod file to offsets
* in our periodtable (but only, if we haven't done this yet) */
p = (unsigned char *) modsong.patterndata;
if (*periodsconverted != 0xfe)
{
int note, note2, channel;
for (note=0;note<maxpatterns*64;note++)
for (channel=0;channel<modsong.noofchannels;channel++)
{
int period = ((p[0] & 0x0f) << 8) | p[1];
int periodoffset = 0;
/* Find the offset of the current period */
for (note2 = 1; note2 < 12*3+1; note2++)
if (abs(modplayer.periodtable[note2*8+1]-period) < 4)
{
periodoffset = note2*8+1;
break;
}
/* Write back the period offset */
p[0] = (periodoffset >> 8) | (p[0] & 0xf0);
p[1] = periodoffset & 0xff;
p += 4;
}
/* Remember that we already converted the periods,
* in case the file gets reloaded by rewinding
* with 0xfe (arbitary magic value > 127) */
*periodsconverted = 0xfe;
}
/* Get the samples
* Calculation: The Samples come after the pattern data
* We know that there are nMaxPatterns and each pattern requires
* 4 bytes per note and per channel.
* And of course there are always lines in each channel */
modsong.sampledata = (signed char*) modsong.patterndata +
maxpatterns*4*modsong.noofchannels*64;
int sampledataoffset = 0;
for (i=0;i<modsong.noofinstruments;i++)
{
modsong.instrument[i].sampledataoffset = sampledataoffset;
sampledataoffset += modsong.instrument[i].length;
}
return true;
}
/* Apply vibrato to channel */
STATICIRAM void vibrate(int channel) ICODE_ATTR;
void vibrate(int channel)
{
struct s_modchannel *p_modchannel = &modplayer.modchannel[channel];
/* Apply Vibrato
* >> 7 is used in the original protracker source code */
mixer_setamigaperiod(channel, p_modchannel->period+
((p_modchannel->vibratodepth *
modplayer.sintable[p_modchannel->vibratosinpos])>>7));
/* Foward in Sine Table */
p_modchannel->vibratosinpos += p_modchannel->vibratospeed;
p_modchannel->vibratosinpos &= 0x3f;
}
/* Apply tremolo to channel
* (same as vibrato, but only apply on volume instead of pitch) */
STATICIRAM void tremolo(int channel) ICODE_ATTR;
void tremolo(int channel)
{
struct s_modchannel *p_modchannel = &modplayer.modchannel[channel];
/* Apply Tremolo
* >> 6 is used in the original protracker source code */
int volume = (p_modchannel->volume *
modplayer.sintable[p_modchannel->tremolosinpos])>>6;
if (volume > 64) volume = 64;
else if (volume < 0) volume = 0;
mixer_setvolume(channel, volume);
/* Foward in Sine Table */
p_modchannel->tremolosinpos += p_modchannel->tremolosinpos;
p_modchannel->tremolosinpos &= 0x3f;
}
/* Apply Slide to Note effect to channel */
STATICIRAM void slidetonote(int channel) ICODE_ATTR;
void slidetonote(int channel)
{
struct s_modchannel *p_modchannel = &modplayer.modchannel[channel];
/* If there hasn't been any slide-to note set up, then return */
if (p_modchannel->slidetonoteperiod == 0) return;
/* Slide note up */
if (p_modchannel->slidetonoteperiod > p_modchannel->period)
{
p_modchannel->period += p_modchannel->slidetonotespeed;
if (p_modchannel->period > p_modchannel->slidetonoteperiod)
p_modchannel->period = p_modchannel->slidetonoteperiod;
}
/* Slide note down */
else if (p_modchannel->slidetonoteperiod < p_modchannel->period)
{
p_modchannel->period -= p_modchannel->slidetonotespeed;
if (p_modchannel->period < p_modchannel->slidetonoteperiod)
p_modchannel->period = p_modchannel->slidetonoteperiod;
}
mixer_setamigaperiod(channel, p_modchannel->period);
}
/* Apply Slide to Note effect on channel,
* but this time with glissando enabled */
STATICIRAM void slidetonoteglissando(int channel) ICODE_ATTR;
void slidetonoteglissando(int channel)
{
struct s_modchannel *p_modchannel = &modplayer.modchannel[channel];
/* Slide note up */
if (p_modchannel->slidetonoteperiod > p_modchannel->period)
{
p_modchannel->period =
modplayer.periodtable[p_modchannel->periodtableoffset+=8];
if (p_modchannel->period > p_modchannel->slidetonoteperiod)
p_modchannel->period = p_modchannel->slidetonoteperiod;
}
/* Slide note down */
else
{
p_modchannel->period =
modplayer.periodtable[p_modchannel->periodtableoffset-=8];
if (p_modchannel->period < p_modchannel->slidetonoteperiod)
p_modchannel->period = p_modchannel->slidetonoteperiod;
}
mixer_setamigaperiod(channel, p_modchannel->period);
}
/* Apply Volume Slide */
STATICIRAM void volumeslide(int channel, int effectx, int effecty) ICODE_ATTR;
void volumeslide(int channel, int effectx, int effecty)
{
struct s_modchannel *p_modchannel = &modplayer.modchannel[channel];
/* If both X and Y Parameters are non-zero, then the y value is ignored */
if (effectx > 0) {
p_modchannel->volume += effectx;
if (p_modchannel->volume > 64) p_modchannel->volume = 64;
}
else {
p_modchannel->volume -= effecty;
if (p_modchannel->volume < 0) p_modchannel->volume = 0;
}
mixer_setvolume(channel, p_modchannel->volume);
}
/* Play the current line (at tick 0) */
STATICIRAM void playline(int pattern, int line) ICODE_ATTR;
void playline(int pattern, int line)
{
int c;
/* Get pointer to the current pattern */
unsigned char *p_line = (unsigned char*)modsong.patterndata;
p_line += pattern*64*4*modsong.noofchannels;
p_line += line*4*modsong.noofchannels;
/* Only allow one Patternbreak Commando per Line */
bool patternbreakdone = false;
for (c=0;c<modsong.noofchannels;c++)
{
struct s_modchannel *p_modchannel = &modplayer.modchannel[c];
unsigned char *p_note = p_line + c*4;
unsigned char samplenumber = (p_note[0] & 0xf0) | (p_note[2] >> 4);
short periodtableoffset = ((p_note[0] & 0x0f) << 8) | p_note[1];
p_modchannel->effect = p_note[2] & 0x0f;
p_modchannel->effectparameter = p_note[3];
/* Remember Instrument and set Volume if new Instrument triggered */
if (samplenumber > 0)
{
/* And trigger new sample, if new instrument was set */
if (samplenumber-1 != p_modchannel->instrument)
{
/* Advance the new sample to the same offset
* the old sample was beeing played */
int oldsampleoffset = mixer.channel[c].samplepos -
modsong.instrument[
p_modchannel->instrument].sampledataoffset;
mixer_playsample(c, samplenumber-1);
mixer.channel[c].samplepos += oldsampleoffset;
}
/* Remember last played instrument on channel */
p_modchannel->instrument = samplenumber-1;
/* Set Volume to standard instrument volume,
* if not overwritten by volume effect */
if (p_modchannel->effect != 0x0c)
{
p_modchannel->volume = modsong.instrument[
p_modchannel->instrument].volume;
mixer_setvolume(c, p_modchannel->volume);
}
}
/* Trigger new sample if note available */
if (periodtableoffset > 0)
{
/* Restart instrument only when new sample triggered */
if (samplenumber != 0)
mixer_playsample(c, (samplenumber > 0) ?
samplenumber-1 : p_modchannel->instrument);
/* Set the new amiga period
* (but only, if there is no slide to note effect) */
if ((p_modchannel->effect != 0x3) &&
(p_modchannel->effect != 0x5))
{
/* Apply finetuning to sample */
p_modchannel->periodtableoffset = periodtableoffset +
modsong.instrument[p_modchannel->instrument].finetune;
p_modchannel->period = modplayer.periodtable[
p_modchannel->periodtableoffset];
mixer_setamigaperiod(c, p_modchannel->period);
/* When a new note is played without slide to note setup,
* then disable slide to note */
modplayer.modchannel[c].slidetonoteperiod =
p_modchannel->period;
}
}
int effectx = p_modchannel->effectparameter>>4;
int effecty = p_modchannel->effectparameter&0x0f;
switch (p_modchannel->effect)
{
/* Effect 0: Arpeggio */
case 0x00:
/* Set the base period on tick 0 */
if (p_modchannel->effectparameter > 0)
mixer_setamigaperiod(c,
modplayer.periodtable[
p_modchannel->periodtableoffset]);
break;
/* Slide up (Portamento up) */
case 0x01:
if (p_modchannel->effectparameter > 0)
p_modchannel->slideupspeed =
p_modchannel->effectparameter;
break;
/* Slide down (Portamento down) */
case 0x02:
if (p_modchannel->effectparameter > 0)
p_modchannel->slidedownspeed =
p_modchannel->effectparameter;
break;
/* Slide to Note */
case 0x03:
if (p_modchannel->effectparameter > 0)
p_modchannel->slidetonotespeed =
p_modchannel->effectparameter;
/* Get the slide to note directly from the pattern buffer */
if (periodtableoffset > 0)
p_modchannel->slidetonoteperiod =
modplayer.periodtable[periodtableoffset +
modsong.instrument[
p_modchannel->instrument].finetune];
/* If glissando is enabled apply the effect directly here */
if (modplayer.glissandoenabled)
slidetonoteglissando(c);
break;
/* Set Vibrato */
case 0x04:
if (effectx > 0) p_modchannel->vibratospeed = effectx;
if (effecty > 0) p_modchannel->vibratodepth = effecty;
break;
/* Effect 0x06: Slide to note */
case 0x05:
/* Get the slide to note directly from the pattern buffer */
if (periodtableoffset > 0)
p_modchannel->slidetonoteperiod =
modplayer.periodtable[periodtableoffset +
modsong.instrument[
p_modchannel->instrument].finetune];
break;
/* Effect 0x06 is "Continue Effects" */
/* It is not processed on tick 0 */
case 0x06:
break;
/* Set Tremolo */
case 0x07:
if (effectx > 0) p_modchannel->tremolodepth = effectx;
if (effecty > 0) p_modchannel->tremolospeed = effecty;
break;
/* Set fine panning */
case 0x08:
/* Internal panning goes from 0..15
* Scale the fine panning value to that range */
mixer.channel[c].panning = p_modchannel->effectparameter>>4;
break;
/* Set Sample Offset */
case 0x09:
{
struct s_instrument *p_instrument =
&modsong.instrument[p_modchannel->instrument];
int sampleoffset = p_instrument->sampledataoffset;
if (sampleoffset > p_instrument->length)
sampleoffset = p_instrument->length;
/* Forward the new offset to the mixer */
mixer.channel[c].samplepos =
p_instrument->sampledataoffset +
(p_modchannel->effectparameter<<8);
mixer.channel[c].samplefractpos = 0;
break;
}
/* Effect 0x0a (Volume slide) is not processed on tick 0 */
/* Position Jump */
case 0x0b:
modplayer.currentline = -1;
modplayer.patterntableposition = (effectx<<4)+effecty;
break;
/* Set Volume */
case 0x0c:
p_modchannel->volume = p_modchannel->effectparameter;
mixer_setvolume(c, p_modchannel->volume);
break;
/* Pattern break */
case 0x0d:
modplayer.currentline = effectx*10 + effecty - 1;
if (!patternbreakdone)
{
patternbreakdone = true;
modplayer.patterntableposition++;
}
break;
/* Extended Effects */
case 0x0e:
switch (effectx)
{
/* Set Filter */
case 0x0:
modplayer.amigafilterenabled =
(effecty>0) ? false : true;
break;
/* Fineslide up */
case 0x1:
mixer_setamigaperiod(c, p_modchannel->period -=
effecty);
if (p_modchannel->period <
modplayer.periodtable[37*8]) p_modchannel->period = 100;
/* Find out the new offset in the period table */
if (p_modchannel->periodtableoffset < 36*8)
while (modplayer.periodtable[
p_modchannel->periodtableoffset+8] >= p_modchannel->period)
p_modchannel->periodtableoffset+=8;
break;
/* Fineslide down */
case 0x2:
mixer_setamigaperiod(c,
p_modchannel->period += effecty);
if (p_modchannel->periodtableoffset > 8)
while (modplayer.periodtable[
p_modchannel->periodtableoffset-8]
<= p_modchannel->period)
p_modchannel->periodtableoffset-=8;
break;
/* Set glissando on/off */
case 0x3:
modplayer.glissandoenabled =
(effecty > 0) ? true:false;
break;
/* Set Vibrato waveform */
case 0x4:
/* Currently not implemented */
break;
/* Set Finetune value */
case 0x5:
/* Treat as signed nibble */
if (effecty > 7) effecty -= 16;
p_modchannel->periodtableoffset +=
effecty -
modsong.instrument[
p_modchannel->instrument].finetune;
p_modchannel->period =
modplayer.periodtable[
p_modchannel->periodtableoffset];
modsong.instrument[
p_modchannel->instrument].finetune = effecty;
break;
/* Pattern loop */
case 0x6:
if (effecty == 0)
modplayer.loopstartline = line-1;
else
{
if (modplayer.looptimes == 0)
{
modplayer.currentline =
modplayer.loopstartline;
modplayer.looptimes = effecty;
}
else modplayer.looptimes--;
if (modplayer.looptimes > 0)
modplayer.currentline =
modplayer.loopstartline;
}
break;
/* Set Tremolo waveform */
case 0x7:
/* Not yet implemented */
break;
/* Enhanced Effect 8 is not used */
case 0x8:
break;
/* Retrigger sample */
case 0x9:
/* Only processed on subsequent ticks */
break;
/* Fine volume slide up */
case 0xa:
p_modchannel->volume += effecty;
if (p_modchannel->volume > 64)
p_modchannel->volume = 64;
mixer_setvolume(c, p_modchannel->volume);
break;
/* Fine volume slide down */
case 0xb:
p_modchannel->volume -= effecty;
if (p_modchannel->volume < 0)
p_modchannel->volume = 0;
mixer_setvolume(c, p_modchannel->volume);
break;
/* Cut sample */
case 0xc:
/* Continue sample */
mixer_continuesample(c);
break;
/* Note delay (Usage: $ED + ticks to delay note.) */
case 0xd:
/* We stop the sample here on tick 0
* and restart it later in the effect */
if (effecty > 0)
mixer.channel[c].channelactive = false;
break;
}
break;
/* Set Speed */
case 0x0f:
if (p_modchannel->effectparameter < 32)
modplayer.ticksperline = p_modchannel->effectparameter;
else
modplayer.bpm = p_modchannel->effectparameter;
break;
}
}
}
/* Play the current effect of the note (ticks 1..speed) */
STATICIRAM void playeffect(int currenttick) ICODE_ATTR;
void playeffect(int currenttick)
{
int c;
for (c=0;c<modsong.noofchannels;c++)
{
struct s_modchannel *p_modchannel = &modplayer.modchannel[c];
/* If there is no note active then there are no effects to play */
if (p_modchannel->period == 0) continue;
unsigned char effectx = p_modchannel->effectparameter>>4;
unsigned char effecty = p_modchannel->effectparameter&0x0f;
switch (p_modchannel->effect)
{
/* Effect 0: Arpeggio */
case 0x00:
if (p_modchannel->effectparameter > 0)
{
unsigned short newperiodtableoffset;
switch (currenttick % 3)
{
case 0:
mixer_setamigaperiod(c,
modplayer.periodtable[
p_modchannel->periodtableoffset]);
break;
case 1:
newperiodtableoffset =
p_modchannel->periodtableoffset+(effectx<<3);
if (newperiodtableoffset < 37*8)
mixer_setamigaperiod(c,
modplayer.periodtable[
newperiodtableoffset]);
break;
case 2:
newperiodtableoffset =
p_modchannel->periodtableoffset+(effecty<<3);
if (newperiodtableoffset < 37*8)
mixer_setamigaperiod(c,
modplayer.periodtable[
newperiodtableoffset]);
break;
}
}
break;
/* Effect 1: Slide Up */
case 0x01:
mixer_setamigaperiod(c,
p_modchannel->period -= p_modchannel->slideupspeed);
/* Find out the new offset in the period table */
if (p_modchannel->periodtableoffset <= 37*8)
while (modplayer.periodtable[
p_modchannel->periodtableoffset] >
p_modchannel->period)
{
p_modchannel->periodtableoffset++;
/* Make sure we don't go out of range */
if (p_modchannel->periodtableoffset > 37*8)
{
p_modchannel->periodtableoffset = 37*8;
break;
}
}
break;
/* Effect 2: Slide Down */
case 0x02:
mixer_setamigaperiod(c, p_modchannel->period +=
p_modchannel->slidedownspeed);
/* Find out the new offset in the period table */
if (p_modchannel->periodtableoffset > 8)
while (modplayer.periodtable[
p_modchannel->periodtableoffset] <
p_modchannel->period)
{
p_modchannel->periodtableoffset--;
/* Make sure we don't go out of range */
if (p_modchannel->periodtableoffset < 1)
{
p_modchannel->periodtableoffset = 1;
break;
}
}
break;
/* Effect 3: Slide to Note */
case 0x03:
/* Apply smooth sliding, if no glissando is enabled */
if (modplayer.glissandoenabled == 0)
slidetonote(c);
break;
/* Effect 4: Vibrato */
case 0x04:
vibrate(c);
break;
/* Effect 5: Continue effect 3:'Slide to note',
* but also do Volume slide */
case 0x05:
slidetonote(c);
volumeslide(c, effectx, effecty);
break;
/* Effect 6: Continue effect 4:'Vibrato',
* but also do Volume slide */
case 0x06:
vibrate(c);
volumeslide(c, effectx, effecty);
break;
/* Effect 7: Tremolo */
case 0x07:
tremolo(c);
break;
/* Effect 8 (Set fine panning) is only processed at tick 0 */
/* Effect 9 (Set sample offset) is only processed at tick 0 */
/* Effect A: Volume slide */
case 0x0a:
volumeslide(c, effectx, effecty);
break;
/* Effect B (Position jump) is only processed at tick 0 */
/* Effect C (Set Volume) is only processed at tick 0 */
/* Effect D (Pattern Preak) is only processed at tick 0 */
/* Effect E (Enhanced Effect) */
case 0x0e:
switch (effectx)
{
/* Retrigger sample ($E9 + Tick to Retrig note at) */
case 0x9:
/* Don't device by zero */
if (effecty == 0) effecty = 1;
/* Apply retrig */
if (currenttick % effecty == 0)
mixer_playsample(c, p_modchannel->instrument);
break;
/* Cut note (Usage: $EC + Tick to Cut note at) */
case 0xc:
if (currenttick == effecty)
mixer_stopsample(c);
break;
/* Delay note (Usage: $ED + ticks to delay note) */
case 0xd:
/* If this is the correct tick,
* we start playing the sample now */
if (currenttick == effecty)
mixer.channel[c].channelactive = true;
break;
}
break;
/* Effect F (Set Speed) is only processed at tick 0 */
}
}
}
inline int clip(int i)
{
if (i > 32767) return(32767);
else if (i < -32768) return(-32768);
else return(i);
}
STATICIRAM void synthrender(void *renderbuffer, int samplecount) ICODE_ATTR;
void synthrender(void *renderbuffer, int samplecount)
{
/* 125bpm equals to 50Hz (= 0.02s)
* => one tick = mixingrate/50,
* samples passing in one tick:
* mixingrate/(bpm/2.5) = 2.5*mixingrate/bpm */
int *p_left = (int *) renderbuffer; /* int in rockbox */
int *p_right = p_left+1;
signed short s;
int qf_distance, qf_distance2;
int i;
int c, left, right;
for (i=0;i<samplecount;i++)
{
/* New Tick? */
if ((modplayer.samplespertick-- <= 0) &&
(modplayer.patterntableposition < 127))
{
if (modplayer.currenttick == 0)
playline(modsong.patternordertable[
modplayer.patterntableposition], modplayer.currentline);
else playeffect(modplayer.currenttick);
modplayer.currenttick++;
if (modplayer.currenttick >= modplayer.ticksperline)
{
modplayer.currentline++;
modplayer.currenttick = 0;
if (modplayer.currentline == 64)
{
modplayer.patterntableposition++;
if (modplayer.patterntableposition >= modsong.songlength)
/* This is for Noise Tracker
* modplayer.patterntableposition =
* modsong.songendjumpposition;
* More compatible approach is restart from 0 */
modplayer.patterntableposition=0;
modplayer.currentline = 0;
}
}
modplayer.samplespertick = (20*mixingrate/modplayer.bpm)>>3;
}
/* Mix buffers from here
* Walk through all channels */
left=0, right=0;
/* If song has not stopped playing */
if (modplayer.patterntableposition < 127)
/* Loop through all channels */
for (c=0;c<modsong.noofchannels;c++)
{
/* Only mix the sample,
* if channel there is something played on the channel */
if (!mixer.channel[c].channelactive) continue;
/* Loop the sample, if requested? */
if (mixer.channel[c].samplepos >= mixer.channel[c].loopend)
{
if (mixer.channel[c].loopsample)
mixer.channel[c].samplepos -=
(mixer.channel[c].loopend-
mixer.channel[c].loopstart);
else mixer.channel[c].channelactive = false;
}
/* If the sample has stopped playing don't mix it */
if (!mixer.channel[c].channelactive) continue;
/* Get the sample */
s = (signed short)(modsong.sampledata[
mixer.channel[c].samplepos]*mixer.channel[c].volume);
/* Interpolate if the sample-frequency is lower
* than the mixing rate
* If you don't want interpolation simply skip this part */
if (mixer.channel[c].frequency < mixingrate)
{
/* Low precision linear interpolation
* (fast integer based) */
qf_distance = mixer.channel[c].samplefractpos<<16 /
mixingrate;
qf_distance2 = (1<<16)-qf_distance;
s = (qf_distance*s + qf_distance2*
mixer.channel[c].lastsampledata)>>16;
}
/* Save the last played sample for interpolation purposes */
mixer.channel[c].lastsampledata = s;
/* Pan the sample */
left += s*(16-mixer.channel[c].panning)>>3;
right += s*mixer.channel[c].panning>>3;
/* Advance sample */
mixer.channel[c].samplefractpos += mixer.channel[c].frequency;
while (mixer.channel[c].samplefractpos > mixingrate)
{
mixer.channel[c].samplefractpos -= mixingrate;
mixer.channel[c].samplepos++;
}
}
/* If we have more than 4 channels
* we have to make sure that we apply clipping */
if (modsong.noofchannels > 4) {
*p_left = clip(left)<<13;
*p_right = clip(right)<<13;
}
else {
*p_left = left<<13;
*p_right = right<<13;
}
p_left+=2;
p_right+=2;
}
}
enum codec_status codec_main(void)
{
size_t n;
unsigned char *modfile;
int old_patterntableposition;
int bytesdone;
ci->configure(CODEC_SET_FILEBUF_WATERMARK, 1024*512);
next_track:
if (codec_init()) {
return CODEC_ERROR;
}
while (!*ci->taginfo_ready && !ci->stop_codec)
ci->sleep(1);
codec_set_replaygain(ci->id3);
/* Load MOD file */
/*
* This is the save way
size_t bytesfree;
unsigned int filesize;
p = modfile;
bytesfree=sizeof(modfile);
while ((n = ci->read_filebuf(p, bytesfree)) > 0) {
p += n;
bytesfree -= n;
if (bytesfree == 0)
return CODEC_ERROR;
}
filesize = p-modfile;
if (filesize == 0)
return CODEC_ERROR;
*/
/* Directly use mod in buffer */
ci->seek_buffer(0);
modfile = ci->request_buffer(&n, ci->filesize);
if (!modfile || n < (size_t)ci->filesize) {
return CODEC_ERROR;
}
initmodplayer();
loadmod(modfile);
/* Make use of 44.1khz */
ci->configure(DSP_SET_FREQUENCY, 44100);
/* Sample depth is 28 bit host endian */
ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
/* Stereo output */
ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
/* The main decoder loop */
ci->set_elapsed(0);
bytesdone = 0;
old_patterntableposition = 0;
while (1) {
ci->yield();
if (ci->stop_codec || ci->new_track)
break;
if (ci->seek_time) {
/* New time is ready in ci->seek_time */
modplayer.patterntableposition = ci->seek_time/1000;
modplayer.currentline = 0;
ci->seek_complete();
}
if(old_patterntableposition != modplayer.patterntableposition) {
ci->set_elapsed(modplayer.patterntableposition*1000+500);
old_patterntableposition=modplayer.patterntableposition;
}
synthrender(samples, CHUNK_SIZE/2);
bytesdone += CHUNK_SIZE;
ci->pcmbuf_insert(samples, NULL, CHUNK_SIZE/2);
}
if (ci->request_next_track())
goto next_track;
return CODEC_OK;
}
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