blob: f0fa93d60e4016365f8b025c1454ebe6699c8c29 [file] [log] [blame]
/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Stepan Moskovchenko
*
* All files in this archive are subject to the GNU General Public License.
* See the file COPYING in the source tree root for full license agreement.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include "plugin.h"
#include "guspat.h"
#include "midiutil.h"
#include "synth.h"
extern struct plugin_api * rb;
void readTextBlock(int file, char * buf)
{
char c = 0;
do
{
c = readChar(file);
} while(c == '\n' || c == ' ' || c=='\t');
rb->lseek(file, -1, SEEK_CUR);
int cp = 0;
do
{
c = readChar(file);
buf[cp] = c;
cp++;
} while (c != '\n' && c != ' ' && c != '\t' && !eof(file));
buf[cp-1]=0;
rb->lseek(file, -1, SEEK_CUR);
}
/* Filename is the name of the config file */
/* The MIDI file should have been loaded at this point */
int initSynth(struct MIDIfile * mf, char * filename, char * drumConfig)
{
char patchUsed[128];
char drumUsed[128];
int a=0;
for(a=0; a<MAX_VOICES; a++)
{
voices[a].cp=0;
voices[a].vol=0;
voices[a].ch=0;
voices[a].isUsed=0;
voices[a].note=0;
}
for(a=0; a<16; a++)
{
chVol[a]=100; /* Default, not quite full blast.. */
chPan[a]=64; /* Center */
chPat[a]=0; /* Ac Gr Piano */
chPW[a]=256; /* .. not .. bent ? */
chPBDepth[a]=2; /* Default bend value is 2 */
}
for(a=0; a<128; a++)
{
patchSet[a]=NULL;
drumSet[a]=NULL;
patchUsed[a]=0;
drumUsed[a]=0;
}
/*
* Always load the piano.
* Some files will assume its loaded without specifically
* issuing a Patch command... then we wonder why we can't hear anything
*/
patchUsed[0]=1;
/* Scan the file to see what needs to be loaded */
if(mf != NULL)
{
for(a=0; a<mf->numTracks; a++)
{
unsigned int ts=0;
if(mf->tracks[a] == NULL)
{
printf("NULL TRACK !!!");
rb->splash(HZ*2, "Null Track in loader.");
return -1;
}
for(ts=0; ts<mf->tracks[a]->numEvents; ts++)
{
if((getEvent(mf->tracks[a], ts)->status) == (MIDI_NOTE_ON+9))
drumUsed[getEvent(mf->tracks[a], ts)->d1]=1;
if( (getEvent(mf->tracks[a], ts)->status & 0xF0) == MIDI_PRGM)
patchUsed[getEvent(mf->tracks[a], ts)->d1]=1;
}
}
} else
{
/* Initialize the whole drum set */
for(a=0; a<128; a++)
drumUsed[a]=1;
}
int file = rb->open(filename, O_RDONLY);
if(file < 0)
{
printf("");
printf("No MIDI patchset found.");
printf("Please install the instruments.");
printf("See Rockbox page for more info.");
rb->splash(HZ*2, "No Instruments");
return -1;
}
char name[40];
char fn[40];
/* Scan our config file and load the right patches as needed */
int c = 0;
name[0] = '\0';
printf("Loading instruments");
for(a=0; a<128; a++)
{
while(readChar(file)!=' ' && !eof(file));
readTextBlock(file, name);
rb->snprintf(fn, 40, ROCKBOX_DIR "/patchset/%s.pat", name);
/* printf("\nLOADING: <%s> ", fn); */
if(patchUsed[a]==1)
{
patchSet[a]=gusload(fn);
if(patchSet[a] == NULL) /* There was an error loading it */
return -1;
}
while((c != '\n'))
c = readChar(file);
}
rb->close(file);
file = rb->open(drumConfig, O_RDONLY);
if(file < 0)
{
rb->splash(HZ*2, "Bad drum config. Did you install the patchset?");
return -1;
}
/* Scan our config file and load the drum data */
int idx=0;
char number[30];
printf("Loading drums");
while(!eof(file))
{
readTextBlock(file, number);
readTextBlock(file, name);
rb->snprintf(fn, 40, ROCKBOX_DIR "/patchset/%s.pat", name);
idx = rb->atoi(number);
if(idx == 0)
break;
if(drumUsed[idx]==1)
{
drumSet[idx]=gusload(fn);
if(drumSet[idx] == NULL) /* Error loading patch */
return -1;
}
while((c != '\n') && (c != 255) && (!eof(file)))
c = readChar(file);
}
rb->close(file);
return 0;
}
#define getSample(s,wf) ((short *)(wf)->data)[s]
void setPoint(struct SynthObject * so, int pt) ICODE_ATTR;
void setPoint(struct SynthObject * so, int pt)
{
if(so->ch==9) /* Drums, no ADSR */
{
so->curOffset = 1<<27;
so->curRate = 1;
return;
}
if(so->wf==NULL)
{
printf("Crap... null waveform...");
exit(1);
}
if(so->wf->envRate==NULL)
{
printf("Waveform has no envelope set");
exit(1);
}
so->curPoint = pt;
int r;
int rate = so->wf->envRate[pt];
r=3-((rate>>6) & 0x3); /* Some blatant Timidity code for rate conversion... */
r*=3;
r = (rate & 0x3f) << r;
/*
* Okay. This is the rate shift. Timidity defaults to 9, and sets
* it to 10 if you use the fast decay option. Slow decay sounds better
* on some files, except on some other files... you get chords that aren't
* done decaying yet.. and they dont harmonize with the next chord and it
* sounds like utter crap. Yes, even Timitidy does that. So I'm going to
* default this to 10, and maybe later have an option to set it to 9
* for longer decays.
*/
so->curRate = r<<10;
/*
* Do this here because the patches assume a 44100 sampling rate
* We've halved our sampling rate, ergo the ADSR code will be
* called half the time. Ergo, double the rate to keep stuff
* sounding right.
*
* Or just move the 1 up one line to optimize a tiny bit.
*/
/* so->curRate = so->curRate << 1; */
so->targetOffset = so->wf->envOffset[pt]<<(20);
if(pt==0)
so->curOffset = 0;
}
inline void stopVoice(struct SynthObject * so)
{
if(so->state == STATE_RAMPDOWN)
return;
so->state = STATE_RAMPDOWN;
so->decay = 0;
}
static inline void synthVoice(struct SynthObject * so, int32_t * out, unsigned int samples)
{
struct GWaveform * wf;
register int s;
register int s1;
register int s2;
register unsigned int cp_temp = so->cp;
wf = so->wf;
const int mode_mask24 = wf->mode&24;
const int mode_mask28 = wf->mode&28;
const int mode_mask_looprev = wf->mode&LOOP_REVERSE;
const unsigned int num_samples = (wf->numSamples-1) << FRACTSIZE;
const unsigned int end_loop = wf->endLoop << FRACTSIZE;
const unsigned int start_loop = wf->startLoop << FRACTSIZE;
const int diff_loop = end_loop-start_loop;
while(samples > 0)
{
samples--;
/* Is voice being ramped? */
if(so->state == STATE_RAMPDOWN)
{
if(so->decay != 0) /* Ramp has been started */
{
so->decay = so->decay / 2;
if(so->decay < 10 && so->decay > -10)
so->isUsed = 0;
s1=so->decay;
s2 = s1*chPan[so->ch];
s1 = (s1<<7) -s2;
*(out++)+=(((s1&0x7FFF80) << 9) | ((s2&0x7FFF80) >> 7));
continue;
}
} else /* OK to advance voice */
{
cp_temp += so->delta;
}
s2 = getSample((cp_temp >> FRACTSIZE)+1, wf);
/* LOOP_REVERSE|LOOP_PINGPONG = 24 */
if(mode_mask24 && so->loopState == STATE_LOOPING && (cp_temp < start_loop))
{
if(mode_mask_looprev)
{
cp_temp += diff_loop;
s2=getSample((cp_temp >> FRACTSIZE), wf);
}
else
{
so->delta = -so->delta; /* At this point cp_temp is wrong. We need to take a step */
so->loopDir = LOOPDIR_FORWARD;
}
}
if(mode_mask28 && (cp_temp >= end_loop))
{
so->loopState = STATE_LOOPING;
if(!mode_mask24)
{
cp_temp -= diff_loop;
s2=getSample((cp_temp >> FRACTSIZE), wf);
}
else
{
so->delta = -so->delta;
so->loopDir = LOOPDIR_REVERSE;
}
}
/* Have we overrun? */
if(cp_temp >= num_samples)
{
cp_temp -= so->delta;
s2 = getSample((cp_temp >> FRACTSIZE)+1, wf);
stopVoice(so);
}
/* Better, working, linear interpolation */
s1=getSample((cp_temp >> FRACTSIZE), wf);
s = s1 + ((signed)((s2 - s1) * (cp_temp & ((1<<FRACTSIZE)-1)))>>FRACTSIZE);
if(so->curRate == 0)
{
stopVoice(so);
// so->isUsed = 0;
}
if(so->ch != 9 && so->state != STATE_RAMPDOWN) /* Stupid ADSR code... and don't do ADSR for drums */
{
if(so->curOffset < so->targetOffset)
{
so->curOffset += (so->curRate);
if(so -> curOffset > so->targetOffset && so->curPoint != 2)
{
if(so->curPoint != 5)
{
setPoint(so, so->curPoint+1);
}
else
{
stopVoice(so);
}
}
} else
{
so->curOffset -= (so->curRate);
if(so -> curOffset < so->targetOffset && so->curPoint != 2)
{
if(so->curPoint != 5)
{
setPoint(so, so->curPoint+1);
}
else
{
stopVoice(so);
}
}
}
}
if(so->curOffset < 0)
{
so->curOffset = so->targetOffset;
stopVoice(so);
}
s = (s * (so->curOffset >> 22) >> 8);
/* need to set ramp beginning */
if(so->state == STATE_RAMPDOWN && so->decay == 0)
{
so->decay = s*so->volscale>>14;
if(so->decay == 0)
so->decay = 1; /* stupid junk.. */
}
/* Scaling by channel volume and note volume is done in sequencer.c */
/* That saves us some multiplication and pointer operations */
s1=s*so->volscale>>14;
s2 = s1*chPan[so->ch];
s1 = (s1<<7) - s2;
*(out++)+=(((s1&0x7FFF80) << 9) | ((s2&0x7FFF80) >> 7));
}
so->cp=cp_temp; /* store this again */
return;
}
/* buffer to hold all the samples for the current tick, this is a hack
neccesary for coldfire targets as pcm_play_data uses the dma which cannot
access iram */
int32_t samp_buf[256] IBSS_ATTR;
/* synth num_samples samples and write them to the */
/* buffer pointed to by buf_ptr */
void synthSamples(int32_t *buf_ptr, unsigned int num_samples) ICODE_ATTR;
void synthSamples(int32_t *buf_ptr, unsigned int num_samples)
{
int i;
struct SynthObject *voicept;
rb->memset(samp_buf, 0, num_samples*4);
for(i=0; i < MAX_VOICES; i++)
{
voicept=&voices[i];
if(voicept->isUsed==1)
{
synthVoice(voicept, samp_buf, num_samples);
}
}
rb->memcpy(buf_ptr, samp_buf, num_samples*4);
/* TODO: Automatic Gain Control, anyone? */
/* Or, should this be implemented on the DSP's output volume instead? */
return; /* No more ghetto lowpass filter. Linear interpolation works well. */
}