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Added support for drumsynth files and added many ds files

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git-svn-id: https://lmms.svn.sf.net/svnroot/lmms/branches/lmms/stable-0.3@549 0778d3d1-df1d-0410-868b-ea421aaaa00d
This commit is contained in:
Paul Giblock
2007-08-28 04:45:58 +00:00
parent a252706ee2
commit 29853b8db6
8 changed files with 110 additions and 97 deletions
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2
Makefile.am
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@@ -196,6 +196,7 @@ lmms_SOURCES = \
$(srcdir)/src/lib/embed.cpp \
$(srcdir)/src/lib/mmp.cpp \
$(srcdir)/src/lib/oscillator.cpp \
$(srcdir)/src/lib/drumsynth.cpp \
$(srcdir)/src/lib/sample_buffer.cpp \
$(srcdir)/src/lib/string_pair_drag.cpp \
$(srcdir)/src/midi/midi_alsa_raw.cpp \
@@ -298,6 +299,7 @@ lmms_SOURCES = \
$(srcdir)/include/track_container.h \
$(srcdir)/include/bb_track.h \
$(srcdir)/include/sample_track.h \
$(srcdir)/include/drumsynth.h \
$(srcdir)/include/sample_buffer.h \
$(srcdir)/include/name_label.h \
$(srcdir)/include/play_handle.h \

26
configure.in
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@@ -596,6 +596,32 @@ AC_CONFIG_FILES([Makefile
data/samples/bassloopes/Makefile
data/samples/beats/Makefile
data/samples/drums/Makefile
data/samples/drumsynth/Makefile
data/samples/drumsynth/acoustic/Makefile
data/samples/drumsynth/cr78/Makefile
data/samples/drumsynth/cr8000/Makefile
data/samples/drumsynth/effects/Makefile
data/samples/drumsynth/electro/Makefile
data/samples/drumsynth/farfisa/Makefile
data/samples/drumsynth/ferraro/Makefile
data/samples/drumsynth/instrument/Makefile
data/samples/drumsynth/jorgensohn/Makefile
data/samples/drumsynth/latin/Makefile
data/samples/drumsynth/linn/Makefile
data/samples/drumsynth/magnetboy/Makefile
data/samples/drumsynth/misc/Makefile
data/samples/drumsynth/misc_bass/Makefile
data/samples/drumsynth/misc_claps/Makefile
data/samples/drumsynth/misc_electro/Makefile
data/samples/drumsynth/misc_fx/Makefile
data/samples/drumsynth/misc_hats/Makefile
data/samples/drumsynth/misc_perc/Makefile
data/samples/drumsynth/misc_synth/Makefile
data/samples/drumsynth/r_b/Makefile
data/samples/drumsynth/tr606/Makefile
data/samples/drumsynth/tr77/Makefile
data/samples/drumsynth/tr808/Makefile
data/samples/drumsynth/tr909/Makefile
data/samples/effects/Makefile
data/samples/instruments/Makefile
data/samples/latin/Makefile

2
data/samples/Makefile.am
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@@ -1,2 +1,2 @@
SUBDIRS = basses bassloopes beats drums effects instruments latin misc shapes stringsnpads
SUBDIRS = basses bassloopes beats drums drumsynth effects instruments latin misc shapes stringsnpads

5
include/sample_buffer.h
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@@ -249,6 +249,11 @@ private:
ch_cnt_t & _channels,
sample_rate_t & _sample_rate );
#endif
f_cnt_t FASTCALL decodeSampleDS( const char * _f,
int_sample_t * & _buf,
ch_cnt_t & _channels,
sample_rate_t & _sample_rate );
QString m_audioFile;
sampleFrame * m_origData;

4
plugins/audio_file_processor/audio_file_processor.cpp
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@@ -724,8 +724,8 @@ const QStringList & audioFileProcessor::subPluginFeatures::supported_extensions(
void )
{
static QStringList extensions = QStringList()
<< "wav" << "ogg" << "spx" << "au" << "voc"
<< "aif" << "aiff" << "flac" << "raw";
<< "wav" << "ogg" << "ds" << "spx" << "au"
<< "voc" << "aif" << "aiff" << "flac" << "raw";
return( extensions );
}

139
plugins/lb302/lb302.cpp
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@@ -127,7 +127,7 @@ void lb302Filter::recalc()
void lb302Filter::envRecalc()
{
vcf_c0 *= fs->envdecay; // Filter Decay. vcf_decay is adjusted for Hz and ENVINC
vcf_c0 *= fs->envdecay; // Filter Decay. vcf_decay is already adjusted for Hz and ENVINC
// vcf_rescoeff = exp(-1.20 + 3.455*(fs->reso)); moved above
};
@@ -250,9 +250,7 @@ void lb302Filter3Pole::envRecalc()
w = vcf_e0 + vcf_c0; // e0 is adjusted for Hz and doesn't need ENVINC
k = (fs->cutoff > 0.975)?0.975:fs->cutoff;
kfco = 50.f+(k)*((2300.f-1600.f*(fs->envmod))+(w)*(700.f+1500.f*(k)+(1500.f+(k)*(44100.f/2.f-6000.f))*(fs->envmod)));
//+iacc*(.3+.7*kfco*kenvmod)*kaccent*kaccurve*2000
//+iacc*(.3+.7*kfco*kenvmod)*kaccent*kaccurve*2000 The accent code
#ifdef LB_24_IGNORE_ENVELOPE
@@ -270,7 +268,7 @@ void lb302Filter3Pole::envRecalc()
#else
kres = (((fs->reso))) * (((-2.7079*kp1 + 10.963)*kp1 - 14.934)*kp1 + 8.4974);
#endif
value = 1.0+( (fs->dist) *(1.5 + 2.0*kres*(1.0-kfcn))); // ENVMOD was DIST*/
value = 1.0+( (fs->dist) *(1.5 + 2.0*kres*(1.0-kfcn))); // ENVMOD was DIST
}
float lb302Filter3Pole::process(const float& samp)
@@ -317,7 +315,22 @@ void lb302Filter3Pole::setState(const lb302FilterState* fs)
//
lb302Synth::lb302Synth( instrumentTrack * _channel_track ) :
instrument( _channel_track, &lb302_plugin_descriptor )
instrument( _channel_track, &lb302_plugin_descriptor ),
vco_shape( SAWTOOTH ),
vco_inc( 0.0 ),
vco_c( 0 ),
vco_k( 0 ),
vco_detune( 0 ),
vco_slide( 0 ),
vco_slideinc( 0 ),
vco_slidebase( 0 ),
vcf_envpos( ENVINC ),
vca_a0( 0.5 ),
vca_mode( 3 ),
vca_a( 0 ),
//vca_attack( 1.0 - 0.94406088 ),
vca_attack( 1.0 - 0.96406088 ),
vca_decay( 0.99897516 )
{
// GUI
@@ -353,7 +366,7 @@ lb302Synth::lb302Synth( instrumentTrack * _channel_track ) :
vcf_dec_knob->setHintText( tr( "Decay:" ) + " ", "" );
vcf_dec_knob->setLabel( tr("DEC") );
slideToggle = new ledCheckBox( "Slide", this,
slideToggle = new ledCheckBox( "Slide", this,
tr( "Slide" ),
_channel_track );
slideToggle->move( 10, 180 );
@@ -447,36 +460,11 @@ lb302Synth::lb302Synth( instrumentTrack * _channel_track ) :
setErasePixmap( PLUGIN_NAME::getIconPixmap( "artwork" ) );
#endif
// SYNTH
vco_inc = 0.0;
vco_c = 0;
vco_k = 0;
vco_slide = 0; vco_slideinc = 0;
vco_slidebase = 0;
fs.cutoff = 0; fs.envmod = 0;
fs.reso = 0; fs.envdecay = 0;
fs.dist = 0;
vcf_envpos = ENVINC;
vco_detune = 0;
vca_mode = 3; vca_a = 0; // Start VCA on an attack.
//vca_attack = 1.0 - 0.94406088;
vca_attack = 1.0 - 0.96406088;
vca_decay = 0.99897516;
vco_shape = SAWTOOTH;
vca_a0 = 0.5; // Experimenting between original (0.5) and 1.0
vca_a = 9;
vca_mode = 3;
vcf = new lb302FilterIIR2(&fs);
use_hold_note = false;
sample_cnt = 0;
@@ -484,16 +472,28 @@ lb302Synth::lb302Synth( instrumentTrack * _channel_track ) :
catch_frame = 0;
catch_decay = 0;
recalcFilter();
framesPerPeriod = engine::getMixer()->framesPerPeriod();
lastFramesPlayed = 1; // because we subtract 1 later
last_offset = 0;
period_states = NULL;
period_states_cnt = 0;
vcf = new lb302FilterIIR2(&fs);
recalcFilter();
filterChanged(0.0);
detuneChanged(0.0);
period_states_cnt = 0;
period_states = new lb302State[framesPerPeriod];
for (int i=0; i < framesPerPeriod; i++) {
period_states[i].vco_c = vco_c;
period_states[i].vca_a = vca_a;
period_states[i].vca_mode = vca_mode;
period_states[i].sample_cnt = sample_cnt;
vcf->getState(&period_states[i].fs);
}
}
@@ -559,7 +559,7 @@ void lb302Synth::filterChanged( float )
float d = 0.2 + (2.3*vcf_dec_knob->value());
d*=LB_HZ; // d *= smpl rate
fs.envdecay = pow(0.1, 1.0/d * ENVINC); // decay is 0.1 to the 1/d * ENVINC
fs.envdecay = pow(0.1, 1.0/d * ENVINC); // decay is 0.1 to the 1/d * ENVINC
// vcf_envdecay is now adjusted for both
// sampling rate and ENVINC
recalcFilter();
@@ -618,7 +618,7 @@ QString lb302Synth::nodeName( void ) const
return( lb302_plugin_descriptor.name );
}
// OBSOLETE. Break apart once we get Q_OBJECT to work. >:[
// OBSOLETE
void lb302Synth::recalcFilter()
{
vcf->recalc();
@@ -636,29 +636,14 @@ void lb302Synth::recalcFilter()
vcf_envpos = ENVINC; // Trigger filter update in process()
}
inline int MIN(int a, int b) {
return (a<b)?a:b;
}
int lb302Synth::process(sampleFrame *outbuf, const Uint32 size)
{
unsigned int i;
float w;
float samp;
for(i=0;i<size;i++) {
/* TODO: ONLY DO THIS IF WE ARE EDGE-TO-EDGE NON-DEAD */
/*if (sample_cnt < desiredTransitionFrames()) {
for(int c=0; c<DEFAULT_CHANNELS; c++)
outbuf[i][c]=0;
sample_cnt++;
continue;
}
*/
// update vcf
if(vcf_envpos >= ENVINC) {
vcf->envRecalc();
@@ -669,15 +654,12 @@ int lb302Synth::process(sampleFrame *outbuf, const Uint32 size)
vco_inc=vco_slidebase-vco_slide;
// Calculate coeff from dec_knob on knob change.
vco_slide*= 0.9+(slide_dec_knob->value()*0.0999); // TODO: Adjust for Hz and ENVINC
}
}
sample_cnt++;
vcf_envpos++;
float old_vco_k = vco_k;
bool looking;
int decay_frames = 128;
// update vco
@@ -695,7 +677,6 @@ int lb302Synth::process(sampleFrame *outbuf, const Uint32 size)
}
}
switch(int(rint(wave_knob->value()))) {
case 0: vco_shape = SAWTOOTH; break;
case 1: vco_shape = INVERTED_SAWTOOTH; break;
@@ -709,51 +690,46 @@ int lb302Synth::process(sampleFrame *outbuf, const Uint32 size)
// add vco_shape_param the changes the shape of each curve.
// merge sawtooths with triangle and square with round square?
switch (vco_shape) {
case SAWTOOTH: // p0: curviness of line
vco_k = vco_c; // Is this sawtooth backwards?
case SAWTOOTH: // p0: curviness of line
vco_k = vco_c; // Is this sawtooth backwards?
break;
case INVERTED_SAWTOOTH: // p0: curviness of line
vco_k = -vco_c; // Is this sawtooth backwards?
vco_k = -vco_c; // Is this sawtooth backwards?
break;
// TODO: I think TRIANGLE is broken.
case TRIANGLE: // p0: duty rev.saw<->triangle<->saw p1: curviness
case TRIANGLE: // p0: duty rev.saw<->triangle<->saw p1: curviness
vco_k = (vco_c*2.0)+0.5;
if (vco_k>0.5) vco_k = 1.0-vco_k;
break;
case SQUARE: // p0: slope of top
case SQUARE: // p0: slope of top
vco_k = (vco_c<0)?0.5:-0.5;
break;
case ROUND_SQUARE: // p0: width of round
case ROUND_SQUARE: // p0: width of round
vco_k = (vco_c<0)?(sqrtf(1-(vco_c*vco_c*4))-0.5):-0.5;
break;
case MOOG: // Maybe the fall should be exponential/sinsoidal instead of quadric.
// [-0.5, 0]: Rise, [0,0.25]: Slope down, [0.25,0.5]: Low
case MOOG: // Maybe the fall should be exponential/sinsoidal instead of quadric.
// [-0.5, 0]: Rise, [0,0.25]: Slope down, [0.25,0.5]: Low
vco_k = (vco_c*2.0)+0.5;
if (vco_k>1.0) vco_k = -0.5 ;
else if (vco_k>0.5) {
w = 2.0*(vco_k-0.5)-1.0;
vco_k = 0.5 - sqrtf(1.0-(w*w));
}
vco_k *= 2.0; // MOOG wave gets filtered away
vco_k *= 2.0; // MOOG wave gets filtered away
break;
}
//vca_a = 0.5;
// 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;
*/
samp *= (float)(decay_frames - catch_decay)/(float)decay_frames;
for(int c=0; c<DEFAULT_CHANNELS; c++) {
@@ -778,7 +754,8 @@ int lb302Synth::process(sampleFrame *outbuf, const Uint32 size)
// the following line actually speeds up processing
if(vca_a < (1/65536.0)) { vca_a = 0; vca_mode = 3; }
}
// Store state
// Store state
period_states[i].vco_c = vco_c;
period_states[i].vca_a = vca_a; // Doesn't change anything (currently)
period_states[i].vca_mode = vca_mode; // Doesn't change anything (currently)
@@ -843,7 +820,6 @@ void lb302Synth::initNote( lb302Note *n)
void lb302Synth::playNote( notePlayHandle * _n, bool )
{
fpp_t framesPerPeriod = engine::getMixer()->framesPerPeriod();
///=== WEIRD CODE FOR MONOPHONIC BEHAVIOUR - BEGIN === ///
@@ -929,17 +905,6 @@ void lb302Synth::playNote( notePlayHandle * _n, bool )
///=== WEIRD CODE FOR MONOPHONIC BEHAVIOUR - END === ///
/// Malloc our period history buffer
if (period_states == NULL) {
period_states = new lb302State[framesPerPeriod];
for (int i=0; i < framesPerPeriod; i++) {
period_states[i].vco_c = vco_c;
period_states[i].vca_a = vca_a; // Doesn't change anything (currently)
period_states[i].vca_mode = vca_mode; // Doesn't change anything (currently)
period_states[i].sample_cnt = sample_cnt; // Doesn't change anything (currently)
vcf->getState(&period_states[i].fs);
}
}
// now resume at the proper position and process as usual
@@ -968,20 +933,12 @@ void lb302Synth::playNote( notePlayHandle * _n, bool )
if(deadToggle->value()==0 && decay_note) {
#ifdef LB_DECAY
if (catch_decay < 1) {
// 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;
catch_decay = 1;
}
else {
}
#else
lb302Note note;
note.vco_inc = _n->frequency()*vco_detune/LB_HZ; // TODO: Use actual sampling rate.

2
plugins/lb302/lb302.h
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@@ -283,7 +283,9 @@ private:
int sample_cnt;
int last_offset;
int framesPerPeriod;
lb302State *period_states;
int period_states_cnt;

27
src/lib/sample_buffer.cpp
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@@ -88,6 +88,8 @@
#include "base64.h"
#include "debug.h"
#include "drumsynth.h"
#ifndef QT4
@@ -258,6 +260,12 @@ void sampleBuffer::update( bool _keep_settings )
samplerate );
}
#endif
if( m_frames == 0 )
{
m_frames = decodeSampleDS( f, buf, channels,
samplerate );
}
if( m_frames > 0 && buf != NULL )
{
// following code transforms int-samples into
@@ -489,7 +497,18 @@ long qfileTellCallback( void * _udata )
}
f_cnt_t sampleBuffer::decodeSampleDS( const char * _f,
int_sample_t * & _buf,
ch_cnt_t & _channels,
sample_rate_t & _samplerate )
{
f_cnt_t frames = 0;
DrumSynth ds;
frames = ds.GetDSFileSamples( _f, _buf, _channels);
return frames;
}
f_cnt_t sampleBuffer::decodeSampleOGGVorbis( const char * _f,
int_sample_t * & _buf,
@@ -896,10 +915,11 @@ QString sampleBuffer::openAudioFile( void ) const
// set filters
#ifdef QT4
QStringList types;
types << tr( "All Audio-Files (*.wav *.ogg *.flac *.spx *.voc *.aif "
types << tr( "All Audio-Files (*.wav *.ogg *.flac *.ds *.spx *.voc *.aif "
"*.aiff *.au *.raw)" )
<< tr( "Wave-Files (*.wav)" )
<< tr( "OGG-Files (*.ogg)" )
<< tr( "DrumSynth-Files (*.ds)" )
<< tr( "FLAC-Files (*.flac)" )
<< tr( "SPEEX-Files (*.spx)" )
//<< tr( "MP3-Files (*.mp3)" )
@@ -912,10 +932,11 @@ QString sampleBuffer::openAudioFile( void ) const
;
ofd.setFilters( types );
#else
ofd.addFilter( tr( "All Audio-Files (*.wav *.ogg *.flac *.spx *.voc "
ofd.addFilter( tr( "All Audio-Files (*.wav *.ogg *.flac *.ds *.spx *.voc "
"*.aif *.aiff *.au *.raw)" ) );
ofd.addFilter( tr( "Wave-Files (*.wav)" ) );
ofd.addFilter( tr( "OGG-Files (*.ogg)" ) );
ofd.addFilter( tr( "DrumSynth-Files (*.ds)" ) );
ofd.addFilter( tr( "FLAC-Files (*.flac)" ) );
ofd.addFilter( tr( "SPEEX-Files (*.spx)" ) );
//ofd.addFilter (tr("MP3-Files (*.mp3)"));
@@ -925,7 +946,7 @@ QString sampleBuffer::openAudioFile( void ) const
ofd.addFilter( tr( "AU-Files (*.au)" ) );
ofd.addFilter( tr( "RAW-Files (*.raw)" ) );
//ofd.addFilter (tr("MOD-Files (*.mod)"));
ofd.setSelectedFilter( tr( "All Audio-Files (*.wav *.ogg *.flac *.spx "
ofd.setSelectedFilter( tr( "All Audio-Files (*.wav *.ogg *.ds *.flac *.spx "
"*.voc *.aif *.aiff *.au *.raw)" ) );
#endif
if( m_audioFile != "" )