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rockbox/apps/codecs/mod.c
Michael Sevakis 7ad2cad173 Commit work started in FS#12153 to put timing/position information in PCM 
buffer chunks.

* Samples and position indication is closely associated with audio data
  instead of compensating by a latency constant. Alleviates problems with
  using the elapsed as a track indicator where it could be off by several
  steps.

* Timing is accurate throughout track even if resampling for pitch shift,
  whereas before it updated during transition latency at the normal 1:1 rate.

* Simpler PCM buffer with a constant chunk size, no linked lists.

In converting crossfade, a minor change was made to not change the WPS until
the fade-in of the incoming track, whereas before it would change upon the
start of the fade-out of the outgoing track possibly having the WPS change
with far too much lead time.

Codec changes are to set elapsed times *before* writing next PCM frame because
 time and position data last set are saved in the next committed PCM chunk. 


git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30366 a1c6a512-1295-4272-9138-f99709370657
2011-08-28 07:45:35 +00:00

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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 */
unsigned short *periodtable;
/* The sinus table [-255,255] */
signed short *sintable;
/* 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;
/* The Amiga Period Table (+1 because we use index 0 for period 0 = no new note) */
static unsigned short s_periodtable[37*8+1] IDATA_ATTR =
{ 0, 907, 900, 893, 887, 881, 874, 868,
862, 856, 849, 843, 837, 831, 825, 819,
813, 808, 802, 796, 790, 785, 779, 773,
768, 762, 757, 751, 746, 740, 735, 730,
725, 719, 714, 709, 704, 699, 694, 689,
684, 679, 674, 669, 664, 660, 655, 650,
645, 641, 636, 632, 627, 623, 618, 614,
609, 605, 600, 596, 592, 588, 583, 579,
575, 571, 567, 563, 559, 555, 551, 547,
543, 539, 535, 531, 527, 523, 520, 516,
512, 509, 505, 501, 498, 494, 490, 487,
483, 480, 477, 473, 470, 466, 463, 460,
456, 453, 450, 446, 443, 440, 437, 434,
431, 428, 424, 421, 418, 415, 412, 409,
406, 404, 401, 398, 395, 392, 389, 386,
384, 381, 378, 375, 373, 370, 367, 365,
362, 359, 357, 354, 352, 349, 347, 344,
342, 339, 337, 334, 332, 330, 327, 325,
322, 320, 318, 316, 313, 311, 309, 307,
304, 302, 300, 298, 296, 294, 291, 289,
287, 285, 283, 281, 279, 277, 275, 273,
271, 269, 267, 265, 263, 261, 260, 258,
256, 254, 252, 250, 249, 247, 245, 243,
241, 240, 238, 236, 235, 233, 231, 230,
228, 226, 225, 223, 221, 220, 218, 217,
215, 214, 212, 210, 209, 207, 206, 204,
203, 202, 200, 199, 197, 196, 194, 193,
192, 190, 189, 187, 186, 185, 183, 182,
181, 179, 178, 177, 176, 174, 173, 172,
171, 169, 168, 167, 166, 165, 163, 162,
161, 160, 159, 158, 156, 155, 154, 153,
152, 151, 150, 149, 148, 147, 145, 144,
143, 142, 141, 140, 139, 138, 137, 136,
135, 134, 133, 132, 131, 130, 130, 129,
128, 127, 126, 125, 124, 123, 122, 121,
120, 120, 119, 118, 117, 116, 115, 115,
114, 113, 112, 111, 110, 110, 109, 108,
107};
/* The sin table */
static signed short s_sintable[0x40] IDATA_ATTR =
{ 0, 25, 49, 74, 97, 120, 141, 162,
180, 197, 212, 225, 235, 244, 250, 254,
255, 254, 250, 244, 235, 225, 212, 197,
180, 161, 141, 120, 97, 73, 49, 24,
0, -25, -50, -74, -98, -120, -142, -162,
-180, -197, -212, -225, -236, -244, -250, -254,
-255, -254, -250, -244, -235, -224, -211, -197,
-180, -161, -141, -119, -97, -73, -49, -24};
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);
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;
}
static inline void mixer_stopsample(int channel)
{
mixer.channel[channel].channelactive = false;
}
static inline void mixer_continuesample(int channel)
{
mixer.channel[channel].channelactive = true;
}
static inline void mixer_setvolume(int channel, int volume)
{
mixer.channel[channel].volume = volume;
}
static inline void mixer_setpanning(int channel, int panning)
{
mixer.channel[channel].panning = panning;
}
static 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 c;
#if 0
/* As the calculation of periodtable and sintable uses float and double
* rockbox uses two predefined tables. This reduces the codesize by
* several KB. */
unsigned int i;
/* 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;
}
#else
/* Point to the predefined tables */
modplayer.periodtable = s_periodtable;
modplayer.sintable = s_sintable;
#endif
/* 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);
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);
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 */
}
}
}
static inline int clip(int i)
{
if (i > 32767) return(32767);
else if (i < -32768) return(-32768);
else return(i);
}
STATICIRAM void synthrender(int32_t *renderbuffer, int samplecount) ICODE_ATTR;
void synthrender(int32_t *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 */
int32_t *p_left = renderbuffer; /* int in rockbox */
int32_t *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;
}
}
/* this is the codec entry point */
enum codec_status codec_main(enum codec_entry_call_reason reason)
{
if (reason == CODEC_LOAD) {
/* 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);
}
return CODEC_OK;
}
/* this is called for each file to process */
enum codec_status codec_run(void)
{
size_t n;
unsigned char *modfile;
int old_patterntableposition;
int bytesdone;
intptr_t param;
if (codec_init()) {
return CODEC_ERROR;
}
codec_set_replaygain(ci->id3);
/* Load MOD file */
ci->seek_buffer(0);
modfile = ci->request_buffer(&n, ci->filesize);
if (!modfile || n < (size_t)ci->filesize) {
return CODEC_ERROR;
}
initmodplayer();
loadmod(modfile);
/* The main decoder loop */
ci->set_elapsed(0);
bytesdone = 0;
old_patterntableposition = 0;
while (1) {
enum codec_command_action action = ci->get_command(&param);
if (action == CODEC_ACTION_HALT)
break;
if (action == CODEC_ACTION_SEEK_TIME) {
/* New time is ready in param */
modplayer.patterntableposition = param/1000;
modplayer.currentline = 0;
ci->seek_complete();
}
if(old_patterntableposition != modplayer.patterntableposition) {
ci->set_elapsed(modplayer.patterntableposition*1000);
old_patterntableposition=modplayer.patterntableposition;
}
synthrender(samples, CHUNK_SIZE/2);
bytesdone += CHUNK_SIZE;
ci->pcmbuf_insert(samples, NULL, CHUNK_SIZE/2);
}
return CODEC_OK;
}
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