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added support for transition-frames for monophonic instruments

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git-svn-id: https://lmms.svn.sf.net/svnroot/lmms/trunk/lmms@521 0778d3d1-df1d-0410-868b-ea421aaaa00d
This commit is contained in:
Tobias Doerffel
2007-08-08 00:25:18 +00:00
parent 8cdcbfcc05
commit 0b799fc3b1
4 changed files with 225 additions and 72 deletions
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9
include/instrument.h
View File

@@ -105,6 +105,15 @@ public:
return( FALSE );
}
// some monophonic instruments need a certain number of frames for a
// transition between notes - such instruments can re-implement this
// method for returning how many frames they at least like to have for
// transition
virtual f_cnt_t desiredTransitionFrames( void ) const
{
return( 0 );
}
inline virtual bool notePlayHandleBased( void ) const
{
return( TRUE );

246
plugins/lb302/lb302.cpp
View File

@@ -61,11 +61,12 @@
// New config
//
#define LB_24_IGNORE_ENVELOPE
#define LB_FILTERED
//#define LB_FILTERED
//#define LB_24_RES_TRICK
#define LB_DIST_RATIO 4.0
#define LB_24_VOL_ADJUST 3.0
//#define LB_DECAY_NOTES
//
// Old config
@@ -410,7 +411,8 @@ lb302Synth::lb302Synth( instrumentTrack * _channel_track ) :
vcf_envpos = ENVINC;
vco_detune = 0;
vca_mode = 2; vca_a = 0;
vca_mode = 0; vca_a = 0; // Start VCA on an attack.
//vca_attack = 1.0 - 0.94406088;
vca_attack = 1.0 - 0.96406088;
vca_decay = 0.99897516;
@@ -424,6 +426,11 @@ lb302Synth::lb302Synth( instrumentTrack * _channel_track ) :
recalcFilter();
lastFramesPlayed = 0;
last_offset = 0;
period_states = NULL;
period_states_cnt = 0;
note_count = 0;
filterChanged(0.0);
detuneChanged(0.0);
@@ -576,6 +583,7 @@ int lb302Synth::process(sampleFrame *outbuf, const Uint32 size)
unsigned int i;
float w;
float samp;
for(i=0;i<size;i++) {
// update vcf
@@ -595,22 +603,29 @@ int lb302Synth::process(sampleFrame *outbuf, const Uint32 size)
sample_cnt++;
vcf_envpos++;
// 01/21/07 Changed to VCF -> VCA instead of VCA -> VCF
#ifdef LB_FILTERED
samp = vcf->process(vco_k)*2.0*vca_a;
#else
samp = vco_k*vca_a;
#endif
for(int c=0; c<DEFAULT_CHANNELS; c++) {
outbuf[i][c]=samp;
}
float old_vco_k = vco_k;
bool looking;
// update vco
vco_c += vco_inc;
vco_c += vco_inc;
if(vco_c > 0.5) vco_c -= 1.0;
if (catch_decay > 0) {
if (catch_decay < desiredReleaseFrames())
catch_decay++;
else if (use_hold_note) {
printf("RETRIGGER\n");
// We would retrigger the actual note we *should* be playing.
// I removed this for now, to help detect the true problem.
/*initNote(&hold_note);
catch_decay=0;
*/
use_hold_note = 0;
}
}
switch(int(rint(wave_knob->value()))) {
case 0: vco_shape = SAWTOOTH; break;
case 1: vco_shape = INVERTED_SAWTOOTH; break;
@@ -658,15 +673,37 @@ int lb302Synth::process(sampleFrame *outbuf, const Uint32 size)
break;
}
// Make it louder. For the better?
//vco_k*=2.0;
// Write out samples.
#ifdef LB_FILTERED
samp = vcf->process(vco_k)*2.0*vca_a;
#else
samp = vco_k*vca_a;
#endif
float releaseFrames = desiredReleaseFrames();
samp *= (releaseFrames - catch_decay)/releaseFrames;
for(int c=0; c<DEFAULT_CHANNELS; c++) {
outbuf[i][c]=samp;
}
// Store state
period_states[i].vco_c = vco_c;
period_states[i].vca_a = vca_a; // Doesn't change anything (currently)
period_states[i].sample_cnt = sample_cnt; // Doesn't change anything (currently)
// Handle Envelope
// TODO: Add decay once I figure out how to extend past the end of a note.
if(sample_cnt>=0.5*44100 /*sz/2*/) vca_mode = 2;
if(vca_mode==0) vca_a+=(vca_a0-vca_a)*vca_attack;
if(vca_mode==0) {
vca_a+=(vca_a0-vca_a)*vca_attack;
if(sample_cnt>=0.5*44100)
vca_mode = 2;
}
else if(vca_mode == 1) {
vca_a *= vca_decay;
//printf("VCA: %d %f\n", dbgshit++, vca_a);
// the following line actually speeds up processing
if(vca_a < (1/65536.0)) { vca_a = 0; vca_mode = 2; }
}
@@ -675,77 +712,149 @@ int lb302Synth::process(sampleFrame *outbuf, const Uint32 size)
return 1;
}
/* Prepares the active LB302 note. I separated this into a function because it
* needs to be called on playNote() when a new note is started. It also needs
* to be called from process() when a prior edge-to-edge note is done releasing.
*/
void lb302Synth::initNote( lb302Note *n)
{
catch_decay = 0;
vco_inc = n->vco_inc;
// TODO: Try moving to the if() below
if(n->dead == 0) {
sample_cnt = 0;
vca_mode = 0; vca_a = 0;
}
// Initiate Slide
// TODO: Break out into function, should be called again on detuneChanged
if (vco_slideinc) {
vco_slide = vco_inc-vco_slideinc;
vco_slidebase = vco_inc;
vco_slideinc = 0;
}
else {
vco_slide = 0;
}
// End break-out
// Slide note, save inc for next note
if (slideToggle->value()) {
vco_slideinc = vco_inc; // May need to equal vco_slidebase+vco_slide if last note slid
}
recalcFilter();
if(n->dead ==0){
// Swap next two blocks??
vcf->playNote();
// Ensure envelope is recalculated
vcf_envpos = ENVINC;
// Double Check
vca_mode = 0;
vca_a = 0.0;
}
}
void lb302Synth::playNote( notePlayHandle * _n, bool )
{
//int nidx = _n->index();
int framesPerPeriod = engine::getMixer()->framesPerPeriod();
//if( _n->nphsOfInstrumentTrack(_n->getInstrumentTrack()).first() != _n )
//if( _n->released() && _n->nphsOfInstrumentTrack( _n->getInstrumentTrack() ).count() > 1 )
// return;
/*
if (_n->released() ) {
if( notePlayHandle::nphsOfInstrumentTrack( getInstrumentTrack() ).size() > 0
&& notePlayHandle::nphsOfInstrumentTrack( getInstrumentTrack(),
TRUE ).last() == _n )
{
return;
}
}
*/
// Malloc our period history buffer
if (period_states == NULL)
period_states = new lb302State[framesPerPeriod];
if ( _n->totalFramesPlayed() <= lastFramesPlayed ) {
// TODO: Try moving to the if() below
if(deadToggle->value()==0) {
sample_cnt = 0;
vca_mode = 0; vca_a = 0;
}
// Print debug garbage
constNotePlayHandleVector v =
notePlayHandle::nphsOfInstrumentTrack( getInstrumentTrack() );
// Adjust inc on SampRate change or detuning change
vco_inc = _n->frequency()*vco_detune/LB_HZ; // TODO: Use actual sampling rate.
if ( _n->totalFramesPlayed() <= 0 )
printf("New Note count: %d nphs: %d\n", note_count, v.size());
// Initiate Slide
// TODO: Break out into function, should be called again on detuneChanged
if (vco_slideinc) {
vco_slide = vco_inc-vco_slideinc;
vco_slidebase = vco_inc;
vco_slideinc = 0;
}
else {
vco_slide = 0;
}
// End break-out
// Slide note, save inc for next note
if (slideToggle->value()) {
vco_slideinc = vco_inc; // May need to equal vco_slidebase+vco_slide if last note slid
}
// OH HOLY FUCK!! These get interleaved between two different notes.
// is LMMS properly handling monophonic? I assumed that the youngest
// note would kill all earlier notes, even if they are released.
if (_n->released())
printf(" RELEASED!!! %ld\n", _n);
else
printf("not released... %ld\n", _n);
// end debug garbage
recalcFilter();
if(deadToggle->value()==0){
// Swap next two blocks??
vcf->playNote();
// Ensure envelope is recalculated
vcf_envpos = ENVINC;
// Start the release decay if this is the first release period.
if (_n->released() && catch_decay == 0)
catch_decay = 1;
// Double Check
vca_mode = 0;
vca_a = 0.0;
// I don't know if and when we will ever need this code again.
/*
if (_n->totalFramesPlayed() > 0) {
period_states_cnt = _n->framesLeftForCurrentPeriod();
}
else {
// New note (not overlapping), restore
if (period_states_cnt > 0) {
lb302State *st = &period_states[period_states_cnt-1];
//printf("Restoring from: %d\n", period_states_cnt-1);
vco_c = st->vco_c;
}
//vca_a = st->vca_a;
//sample_cnt = st->sample_cnt;
}
*/
if ( _n->totalFramesPlayed() <= 0 ) {
// Existing note. Allow it to decay.
if(note_count) {
// BEGIN NOT SURE OF...
//lb302State *st = &period_states[period_states_cnt-1];
//vca_a = st->vca_a;
//sample_cnt = st->sample_cnt;
// END NOT SURE OF
// Reserve this note for retrigger in process()
hold_note.vco_inc = _n->frequency()*vco_detune/LB_HZ; // TODO: Use actual sampling rate.
hold_note.dead = deadToggle->value();
use_hold_note = true;
}
// Start a new note
else {
lb302Note note;
note.vco_inc = _n->frequency()*vco_detune/LB_HZ; // TODO: Use actual sampling rate.
note.dead = deadToggle->value();
initNote(&note);
use_hold_note = false;
}
note_count=1;
}
const fpp_t frames = _n->framesLeftForCurrentPeriod();
sampleFrame *buf = new sampleFrame[frames];
bool dec_count;
if (_n->willFinishThisPeriod())
dec_count = true;
else
dec_count = false;
process(buf, frames);
getInstrumentTrack()->processAudioBuffer( buf, frames, _n );
delete[] buf;
if (dec_count)
note_count=0;
lastFramesPlayed = _n->totalFramesPlayed();
}
@@ -754,7 +863,6 @@ void lb302Synth::playNote( notePlayHandle * _n, bool )
void lb302Synth::deleteNotePluginData( notePlayHandle * _n )
{
}

35
plugins/lb302/lb302.h
View File

@@ -120,6 +120,21 @@ class lb302Filter3Pole : public lb302Filter
value;
};
class lb302State
{
public:
float vco_c;
float vca_a;
int sample_cnt;
};
class lb302Note
{
public:
float vco_inc;
bool dead;
};
class lb302Synth : public instrument
{
@@ -143,8 +158,16 @@ public:
return true;
}
virtual f_cnt_t desiredReleaseFrames( void ) const
{
return 2000;
}
private:
void initNote(lb302Note *note);
private:
knob * vcf_cut_knob;
@@ -192,7 +215,8 @@ private:
// User settings
lb302FilterState fs;
lb302Filter *vcf;
lb302Note hold_note;
bool use_hold_note;
float lastFramesPlayed;
@@ -210,6 +234,15 @@ private:
// My hacks
int sample_cnt;
int last_offset;
lb302State *period_states;
int period_states_cnt;
int catch_frame;
int catch_decay;
int note_count;
void recalcFilter();

7
src/tracks/instrument_track.cpp
View File

@@ -781,6 +781,8 @@ void instrumentTrack::playNote( notePlayHandle * _n, bool _try_parallelizing )
{
processInEvent( midiEvent( NOTE_OFF, 0,
_n->key(), 0 ), midiTime() );
_n->noteOff( ( *youngest_note )->offset()+m_instrument->desiredTransitionFrames() );
#if 0
if( ( *youngest_note )->offset() >
_n->offset() )
{
@@ -797,12 +799,13 @@ void instrumentTrack::playNote( notePlayHandle * _n, bool _try_parallelizing )
// in last period and have
// to clear parts of it
_n->noteOff();
engine::getMixer()->clearAudioBuffer( m_audioPort->firstBuffer(),
/* engine::getMixer()->clearAudioBuffer( m_audioPort->firstBuffer(),
engine::getMixer()->framesPerPeriod() -
( *youngest_note )->offset(),
( *youngest_note )->offset() );
return;
return;*/
}
#endif
}
}
}