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rockbox/apps/plugins/midi/synth.c

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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Stepan Moskovchenko
*
* 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.
*
****************************************************************************/
#include "plugin.h"
#include "guspat.h"
#include "midiutil.h"
#include "synth.h"
static 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);
}
void resetControllers()
{
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=false;
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 */
chPBNoteOffset[a]=0; /* No offset */
chPBFractBend[a]=65536; /* Center.. no bend */
chLastCtrlMSB[a]=0; /* Set to pitch bend depth */
chLastCtrlLSB[a]=0; /* Set to pitch bend depth */
}
}
/* 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;
resetControllers();
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)
{
midi_debug("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)
{
midi_debug("");
midi_debug("No MIDI patchset found.");
midi_debug("Please install the instruments.");
midi_debug("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';
midi_debug("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);
/* midi_debug("\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];
midi_debug("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;
}
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)
{
midi_debug("Crap... null waveform...");
exit(1);
}
if(so->wf->envRate==NULL)
{
midi_debug("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;
}
static inline void synthVoice(struct SynthObject * so, int32_t * out, unsigned int samples)
{
struct GWaveform * wf;
register int s1;
register int s2;
register unsigned int cp_temp = so->cp;
wf = so->wf;
const int16_t *sample_data = wf->data;
const unsigned int pan = chPan[so->ch];
const int volscale = so->volscale;
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;
bool rampdown = (so->state == STATE_RAMPDOWN);
const bool ch_9 = (so->ch == 9);
while(LIKELY(samples-- > 0))
{
/* Is voice being ramped? */
if(UNLIKELY(rampdown))
{
if(so->decay != 0) /* Ramp has been started */
{
so->decay = so->decay / 2;
if(so->decay < 10 && so->decay > -10)
so->isUsed = false;
s1 = so->decay;
s2 = s1 * pan;
s1 = (s1 << 7) -s2;
*(out++) += s1;
*(out++) += s2;
continue;
}
} else /* OK to advance voice */
{
cp_temp += so->delta;
}
s2 = sample_data[(cp_temp >> FRACTSIZE)+1];
if(LIKELY(mode_mask28))
{
/* LOOP_REVERSE|LOOP_PINGPONG = 24 */
if(UNLIKELY(mode_mask24 && so->loopState == STATE_LOOPING && (cp_temp < start_loop)))
{
if(UNLIKELY(mode_mask_looprev))
{
cp_temp += diff_loop;
s2 = sample_data[cp_temp >> FRACTSIZE];
}
else
{
so->delta = -so->delta; /* At this point cp_temp is wrong. We need to take a step */
}
}
if(UNLIKELY(cp_temp >= end_loop))
{
so->loopState = STATE_LOOPING;
if(UNLIKELY(!mode_mask24))
{
cp_temp -= diff_loop;
s2 = sample_data[cp_temp >> FRACTSIZE];
}
else
{
so->delta = -so->delta;
}
}
}
/* Have we overrun? */
if(UNLIKELY(cp_temp >= num_samples))
{
cp_temp -= so->delta;
s2 = sample_data[(cp_temp >> FRACTSIZE)+1];
if (!rampdown) /* stop voice */
{
rampdown = true;
so->decay = 0;
}
}
/* Better, working, linear interpolation */
s1 = sample_data[cp_temp >> FRACTSIZE];
s1 +=((signed)((s2 - s1) * (cp_temp & ((1<<FRACTSIZE)-1)))>>FRACTSIZE);
if(UNLIKELY(so->curRate == 0))
{
if (!rampdown) /* stop voice */
{
rampdown = true;
so->decay = 0;
}
// so->isUsed = false;
}
if(LIKELY(!ch_9 && !rampdown)) /* Stupid ADSR code... and don't do ADSR for drums */
{
if(UNLIKELY(so->curOffset < so->targetOffset))
{
so->curOffset += (so->curRate);
if(UNLIKELY(so -> curOffset > so->targetOffset && so->curPoint != 2))
{
if(UNLIKELY(so->curPoint != 5))
{
setPoint(so, so->curPoint+1);
}
else if (!rampdown) /* stop voice */
{
rampdown = true;
so->decay = 0;
}
}
} else
{
so->curOffset -= (so->curRate);
if(UNLIKELY(so -> curOffset < so->targetOffset && so->curPoint != 2))
{
if(UNLIKELY(so->curPoint != 5))
{
setPoint(so, so->curPoint+1);
}
else if (!rampdown) /* stop voice */
{
rampdown = true;
so->decay = 0;
}
}
}
}
if(UNLIKELY(so->curOffset < 0))
{
so->curOffset = so->targetOffset;
if (!rampdown)
{
rampdown = true;
so->decay = 0;
}
}
s1 = s1 * (so->curOffset >> 22) >> 8;
/* Scaling by channel volume and note volume is done in sequencer.c */
/* That saves us some multiplication and pointer operations */
s1 = s1 * volscale >> 14;
/* need to set ramp beginning */
if(UNLIKELY(rampdown && so->decay == 0))
{
so->decay = s1;
if(so->decay == 0)
so->decay = 1; /* stupid junk.. */
}
s2 = s1 * pan;
s1 = (s1 << 7) - s2;
*(out++) += s1;
*(out++) += s2;
}
/* store these again */
if (rampdown)
so->state = STATE_RAMPDOWN;
so->cp = cp_temp;
return;
}
/* synth num_samples samples and write them to the */
/* buffer pointed to by buf_ptr */
size_t synthSamples(int32_t *buf_ptr, size_t num_samples) ICODE_ATTR;
size_t synthSamples(int32_t *buf_ptr, size_t num_samples)
{
unsigned int i;
struct SynthObject *voicept;
size_t nsamples = MIN(num_samples, MAX_SAMPLES);
rb->memset(buf_ptr, 0, nsamples * 2 * sizeof(int32_t));
for(i=0; i < MAX_VOICES; i++)
{
voicept=&voices[i];
if(voicept->isUsed)
{
synthVoice(voicept, buf_ptr, nsamples);
}
}
/* TODO: Automatic Gain Control, anyone? */
/* Or, should this be implemented on the DSP's output volume instead? */
return nsamples; /* No more ghetto lowpass filter. Linear interpolation works well. */
}
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