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rockbox/apps/plugins/midi/synth.c
Stepan Moskovchenko fb3e9a4073 Fix error/warning, change pitch bend to 9 bits 
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7288 a1c6a512-1295-4272-9138-f99709370657
2005-08-07 22:20:40 +00:00

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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
*
* 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.
*
****************************************************************************/
extern struct plugin_api * rb;
struct Event * getEvent(struct Track * tr, int evNum)
{
return tr->dataBlock + (evNum*sizeof(struct Event));
}
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.. */
chPanLeft[a]=64; /* Center */
chPanRight[a]=64; /* Center */
chPat[a]=0; /* Ac Gr Piano */
chPW[a]=256; /* .. not .. bent ? */
}
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 */
for(a=0; a<mf->numTracks; a++)
{
unsigned int ts=0;
if(mf->tracks[a] == NULL)
{
printf("\nNULL TRACK !!!");
rb->splash(HZ*2, true, "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)
{
if(patchUsed[getEvent(mf->tracks[a], ts)->d1]==0)
printf("\nI need to load patch %d.", getEvent(mf->tracks[a], ts)->d1);
patchUsed[getEvent(mf->tracks[a], ts)->d1]=1;
}
}
}
int file = rb->open(filename, O_RDONLY);
if(file == -1)
{
rb->splash(HZ*2, true, "Bad patch config.\nDid you install the patchset?");
return -1;
}
char name[40];
char fn[40];
/* Scan our config file and load the right patches as needed */
int c = 0;
rb->snprintf(name, 40, "");
for(a=0; a<128; a++)
{
while(readChar(file)!=' ' && !eof(file));
readTextBlock(file, name);
rb->snprintf(fn, 40, "/.rockbox/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 == -1)
{
rb->splash(HZ*2, true, "Bad drum config.\nDid you install the patchset?");
return -1;
}
/* Scan our config file and load the drum data */
int idx=0;
char number[30];
while(!eof(file))
{
readTextBlock(file, number);
readTextBlock(file, name);
rb->snprintf(fn, 40, "/.rockbox/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;
}
int currentVoice IDATA_ATTR;
struct SynthObject * so IDATA_ATTR;
struct GWaveform * wf IDATA_ATTR;
int s IDATA_ATTR;
short s1 IDATA_ATTR;
short s2 IDATA_ATTR;
short sample IDATA_ATTR; /* For synthSample */
unsigned int cpShifted IDATA_ATTR;
unsigned char b1 IDATA_ATTR;
unsigned char b2 IDATA_ATTR;
inline int getSample(int s)
{
/* Sign conversion moved to guspat.c */
/* 8bit conversion NOT YET IMPLEMENTED in guspat.c */
return ((short *) wf->data)[s];
}
inline 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("\nCrap... null waveform...");
exit(1);
}
if(so->wf->envRate==NULL)
{
printf("\nWaveform has no envelope set");
exit(1);
}
so->curPoint = pt;
int r=0;
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.
*/
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 = 255;
}
inline signed short int synthVoice(void)
{
so = &voices[currentVoice];
wf = so->wf;
if(so->state != STATE_RAMPDOWN)
{
so->cp += so->delta;
}
cpShifted = so->cp >> 10; //Was 10
if( (cpShifted > (wf->numSamples) && (so->state != STATE_RAMPDOWN)))
{
stopVoice(so);
}
s2 = getSample((cpShifted)+1);
/* LOOP_REVERSE|LOOP_PINGPONG = 24 */
if((wf->mode & (24)) && so->loopState == STATE_LOOPING && (cpShifted <= (wf->startLoop)))
{
if(wf->mode & LOOP_REVERSE)
{
so->cp = (wf->endLoop)<<10; //Was 10
cpShifted = wf->endLoop;
s2=getSample((cpShifted));
} else
{
so->delta = -so->delta;
so->loopDir = LOOPDIR_FORWARD;
}
}
if((wf->mode & 28) && (cpShifted >= wf->endLoop))
{
so->loopState = STATE_LOOPING;
if((wf->mode & (24)) == 0)
{
so->cp = (wf->startLoop)<<10; //Was 10
cpShifted = wf->startLoop;
s2=getSample((cpShifted));
} else
{
so->delta = -so->delta;
so->loopDir = LOOPDIR_REVERSE;
}
}
/* Better, working, linear interpolation */
s1=getSample((cpShifted)); //\|/ Was 1023)) >> 10
s = s1 + ((signed)((s2 - s1) * (so->cp & 1023))>>10); //Was 10
/* ADSR COMMENT WOULD GO FROM HERE.........*/
if(so->curRate == 0)
stopVoice(so);
if(so->ch != 9) /* 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->isUsed=0; /* This is OK because offset faded it out already */
s = (s * (so->curOffset >> 22) >> 8);
/* ............. TO HERE */
if(so->state == STATE_RAMPDOWN)
{
so->decay--;
if(so->decay == 0)
so->isUsed=0;
s = (s * so->decay) >> 8;
}
return s*((signed short int)so->vol*(signed short int)chVol[so->ch])>>14;
}
inline void synthSample(int * mixL, int * mixR)
{
*mixL = 0;
*mixR = 0;
for(currentVoice=0; currentVoice<MAX_VOICES; currentVoice++)
{
if(voices[currentVoice].isUsed==1)
{
sample = synthVoice();
*mixL += (sample*chPanLeft[voices[currentVoice].ch])>>7;
*mixR += (sample*chPanRight[voices[currentVoice].ch])>>7;
}
}
/* TODO: Automatic Gain Control, anyone? */
/* Or, should this be implemented on the DSP's output volume instead? */
return; /* No more ghetto lowpass filter.. linear intrpolation works well. */
}
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