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Merge pull request #911 from diizy/master-nph

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Revision of handling of frameoffset for NPH, SPH
This commit is contained in:
Vesa V
2014-06-30 12:52:44 +03:00
parent 9c8c80683b 23433a70b5
commit 6ffcc0e799
28 changed files with 235 additions and 204 deletions
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27
include/AutomatableModel.h
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@@ -26,6 +26,7 @@
#define AUTOMATABLE_MODEL_H
#include <math.h>
#include <QtCore/QMutex>
#include "JournallingObject.h"
#include "Model.h"
@@ -103,9 +104,6 @@ public:
{
return isAutomated() || m_controllerConnection != NULL;
}
bool hasSampleExactData() const;
ControllerConnection* controllerConnection() const
{
@@ -142,10 +140,8 @@ public:
float controllerValue( int frameOffset ) const;
// returns sample-exact data as a ValueBuffer
// should only be called when sample-exact data exists
// in other cases (eg. for automation), the receiving end should interpolate
// the values themselves
//! @brief Function that returns sample-exact data as a ValueBuffer
//! @return pointer to model's valueBuffer when s.ex.data exists, NULL otherwise
ValueBuffer * valueBuffer();
template<class T>
@@ -264,6 +260,16 @@ public:
{
m_hasStrictStepSize = b;
}
static void incrementPeriodCounter()
{
++s_periodCounter;
}
static void resetPeriodCounter()
{
s_periodCounter = 0;
}
public slots:
virtual void reset();
@@ -332,6 +338,13 @@ private:
static float s_copiedValue;
ValueBuffer m_valueBuffer;
long m_lastUpdatedPeriod;
static long s_periodCounter;
bool m_hasSampleExactData;
// prevent several threads from attempting to write the same vb at the same time
QMutex m_valueBufferMutex;
signals:
void initValueChanged( float val );

1
include/Mixer.h
View File

@@ -312,7 +312,6 @@ public:
// audio-buffer-mgm
void bufferToPort( const sampleFrame * _buf,
const fpp_t _frames,
const f_cnt_t _offset,
stereoVolumeVector _volume_vector,
AudioPort * _port );

13
include/NotePlayHandle.h
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@@ -74,6 +74,16 @@ public:
{
return m_midiChannel;
}
/*! convenience function that returns offset for the first period and zero otherwise,
used by instruments to handle the offset: instruments have to check this property and
add the correct number of empty frames in the beginning of the period */
f_cnt_t noteOffset() const
{
return m_totalFramesPlayed == 0
? offset()
: 0;
}
const float& frequency() const
{
@@ -94,7 +104,7 @@ public:
/*! Returns whether playback of note is finished and thus handle can be deleted */
virtual bool isFinished() const
{
return m_released && framesLeft() <= 0 && m_scheduledNoteOff < 0;
return m_released && framesLeft() <= 0;
}
/*! Returns number of frames left for playback */
@@ -264,7 +274,6 @@ private:
// played after release
f_cnt_t m_releaseFramesDone; // number of frames done after
// release of note
f_cnt_t m_scheduledNoteOff; // variable for scheduling noteoff at next period
NotePlayHandleList m_subNotes; // used for chords and arpeggios
volatile bool m_released; // indicates whether note is released
bool m_hasParent;

4
include/PlayHandle.h
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@@ -88,8 +88,8 @@ public:
virtual void play( sampleFrame* buffer ) = 0;
virtual bool isFinished( void ) const = 0;
// returns how many frames this play-handle is aligned ahead, i.e.
// at which position it is inserted in the according buffer
// returns the frameoffset at the start of the playhandle,
// ie. how many empty frames should be inserted at the start of the first period
f_cnt_t offset() const
{
return m_offset;

6
include/SamplePlayHandle.h
View File

@@ -22,8 +22,8 @@
*
*/
#ifndef _SAMPLE_PLAY_HANDLE_H
#define _SAMPLE_PLAY_HANDLE_H
#ifndef SAMPLE_PLAY_HANDLE_H
#define SAMPLE_PLAY_HANDLE_H
#include "Mixer.h"
#include "SampleBuffer.h"
@@ -49,7 +49,7 @@ public:
}
virtual void play( sampleFrame * _working_buffer );
virtual void play( sampleFrame * buffer );
virtual bool isFinished() const;
virtual bool isFromTrack( const track * _track ) const;

19
plugins/Amplifier/Amplifier.cpp
View File

@@ -75,21 +75,10 @@ bool AmplifierEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames )
const float d = dryLevel();
const float w = wetLevel();
ValueBuffer * volBuf = m_ampControls.m_volumeModel.hasSampleExactData()
? m_ampControls.m_volumeModel.valueBuffer()
: NULL;
ValueBuffer * panBuf = m_ampControls.m_panModel.hasSampleExactData()
? m_ampControls.m_panModel.valueBuffer()
: NULL;
ValueBuffer * leftBuf = m_ampControls.m_leftModel.hasSampleExactData()
? m_ampControls.m_leftModel.valueBuffer()
: NULL;
ValueBuffer * rightBuf = m_ampControls.m_rightModel.hasSampleExactData()
? m_ampControls.m_rightModel.valueBuffer()
: NULL;
ValueBuffer * volBuf = m_ampControls.m_volumeModel.valueBuffer();
ValueBuffer * panBuf = m_ampControls.m_panModel.valueBuffer();
ValueBuffer * leftBuf = m_ampControls.m_leftModel.valueBuffer();
ValueBuffer * rightBuf = m_ampControls.m_rightModel.valueBuffer();
for( fpp_t f = 0; f < frames; ++f )
{

5
plugins/audio_file_processor/audio_file_processor.cpp
View File

@@ -119,6 +119,7 @@ void audioFileProcessor::playNote( NotePlayHandle * _n,
sampleFrame * _working_buffer )
{
const fpp_t frames = _n->framesLeftForCurrentPeriod();
const f_cnt_t offset = _n->noteOffset();
// Magic key - a frequency < 20 (say, the bottom piano note if using
// a A4 base tuning) restarts the start point. The note is not actually
@@ -165,14 +166,14 @@ void audioFileProcessor::playNote( NotePlayHandle * _n,
if( ! _n->isFinished() )
{
if( m_sampleBuffer.play( _working_buffer,
if( m_sampleBuffer.play( _working_buffer + offset,
(handleState *)_n->m_pluginData,
frames, _n->frequency(),
static_cast<SampleBuffer::LoopMode>( m_loopModel.value() ) ) )
{
applyRelease( _working_buffer, _n );
instrumentTrack()->processAudioBuffer( _working_buffer,
frames,_n );
frames + offset, _n );
emit isPlaying( ((handleState *)_n->m_pluginData)->frameIndex() );
}

5
plugins/bit_invader/bit_invader.cpp
View File

@@ -282,9 +282,10 @@ void bitInvader::playNote( NotePlayHandle * _n,
}
const fpp_t frames = _n->framesLeftForCurrentPeriod();
const f_cnt_t offset = _n->noteOffset();
bSynth * ps = static_cast<bSynth *>( _n->m_pluginData );
for( fpp_t frame = 0; frame < frames; ++frame )
for( fpp_t frame = offset; frame < frames + offset; ++frame )
{
const sample_t cur = ps->nextStringSample();
for( ch_cnt_t chnl = 0; chnl < DEFAULT_CHANNELS; ++chnl )
@@ -295,7 +296,7 @@ void bitInvader::playNote( NotePlayHandle * _n,
applyRelease( _working_buffer, _n );
instrumentTrack()->processAudioBuffer( _working_buffer, frames, _n );
instrumentTrack()->processAudioBuffer( _working_buffer, frames + offset, _n );
}

12
plugins/kicker/kicker.cpp
View File

@@ -163,6 +163,8 @@ typedef KickerOsc<DspEffectLibrary::MonoToStereoAdaptor<DistFX> > SweepOsc;
void kickerInstrument::playNote( NotePlayHandle * _n,
sampleFrame * _working_buffer )
{
const fpp_t frames = _n->framesLeftForCurrentPeriod();
const f_cnt_t offset = _n->noteOffset();
const float decfr = m_decayModel.value() *
engine::mixer()->processingSampleRate() / 1000.0f;
const f_cnt_t tfp = _n->totalFramesPlayed();
@@ -187,10 +189,8 @@ void kickerInstrument::playNote( NotePlayHandle * _n,
_n->noteOff();
}
const fpp_t frames = _n->framesLeftForCurrentPeriod();
SweepOsc * so = static_cast<SweepOsc *>( _n->m_pluginData );
so->update( _working_buffer, frames, engine::mixer()->processingSampleRate() );
so->update( _working_buffer + offset, frames, engine::mixer()->processingSampleRate() );
if( _n->isReleased() )
{
@@ -199,12 +199,12 @@ void kickerInstrument::playNote( NotePlayHandle * _n,
for( fpp_t f = 0; f < frames; ++f )
{
const float fac = ( done+f < desired ) ? ( 1.0f - ( ( done+f ) / desired ) ) : 0;
_working_buffer[f][0] *= fac;
_working_buffer[f][1] *= fac;
_working_buffer[f+offset][0] *= fac;
_working_buffer[f+offset][1] *= fac;
}
}
instrumentTrack()->processAudioBuffer( _working_buffer, frames, _n );
instrumentTrack()->processAudioBuffer( _working_buffer, frames + offset, _n );
}

9
plugins/monstro/Monstro.cpp
View File

@@ -1245,21 +1245,22 @@ MonstroInstrument::~MonstroInstrument()
void MonstroInstrument::playNote( NotePlayHandle * _n,
sampleFrame * _working_buffer )
{
const fpp_t frames = _n->framesLeftForCurrentPeriod();
const f_cnt_t offset = _n->offset();
if ( _n->totalFramesPlayed() == 0 || _n->m_pluginData == NULL )
{
const sample_rate_t samplerate = m_samplerate;
_n->m_pluginData = new MonstroSynth( this, _n, samplerate, m_fpp );
}
const fpp_t frames = _n->framesLeftForCurrentPeriod();
MonstroSynth * ms = static_cast<MonstroSynth *>( _n->m_pluginData );
ms->renderOutput( frames, _working_buffer );
ms->renderOutput( frames, _working_buffer + offset );
//applyRelease( _working_buffer, _n ); // we have our own release
instrumentTrack()->processAudioBuffer( _working_buffer, frames, _n );
instrumentTrack()->processAudioBuffer( _working_buffer, frames + offset, _n );
}
void MonstroInstrument::deleteNotePluginData( NotePlayHandle * _n )

9
plugins/nes/Nes.cpp
View File

@@ -556,21 +556,22 @@ NesInstrument::~NesInstrument()
void NesInstrument::playNote( NotePlayHandle * n, sampleFrame * workingBuffer )
{
const fpp_t frames = n->framesLeftForCurrentPeriod();
const f_cnt_t offset = n->noteOffset();
if ( n->totalFramesPlayed() == 0 || n->m_pluginData == NULL )
{
NesObject * nes = new NesObject( this, engine::mixer()->processingSampleRate(), n, engine::mixer()->framesPerPeriod() );
n->m_pluginData = nes;
}
const fpp_t frames = n->framesLeftForCurrentPeriod();
NesObject * nes = static_cast<NesObject *>( n->m_pluginData );
nes->renderOutput( workingBuffer, frames );
nes->renderOutput( workingBuffer + offset, frames );
applyRelease( workingBuffer, n );
instrumentTrack()->processAudioBuffer( workingBuffer, frames, n );
instrumentTrack()->processAudioBuffer( workingBuffer, frames + offset, n );
}

11
plugins/organic/organic.cpp
View File

@@ -227,6 +227,9 @@ QString organicInstrument::nodeName() const
void organicInstrument::playNote( NotePlayHandle * _n,
sampleFrame * _working_buffer )
{
const fpp_t frames = _n->framesLeftForCurrentPeriod();
const f_cnt_t offset = _n->noteOffset();
if( _n->totalFramesPlayed() == 0 || _n->m_pluginData == NULL )
{
Oscillator * oscs_l[m_numOscillators];
@@ -296,10 +299,8 @@ void organicInstrument::playNote( NotePlayHandle * _n,
Oscillator * osc_l = static_cast<oscPtr *>( _n->m_pluginData )->oscLeft;
Oscillator * osc_r = static_cast<oscPtr *>( _n->m_pluginData)->oscRight;
const fpp_t frames = _n->framesLeftForCurrentPeriod();
osc_l->update( _working_buffer, frames, 0 );
osc_r->update( _working_buffer, frames, 1 );
osc_l->update( _working_buffer + offset, frames, 0 );
osc_r->update( _working_buffer + offset, frames, 1 );
// -- fx section --
@@ -317,7 +318,7 @@ void organicInstrument::playNote( NotePlayHandle * _n,
// -- --
instrumentTrack()->processAudioBuffer( _working_buffer, frames, _n );
instrumentTrack()->processAudioBuffer( _working_buffer, frames + offset, _n );
}

5
plugins/papu/papu_instrument.cpp
View File

@@ -238,6 +238,7 @@ void papuInstrument::playNote( NotePlayHandle * _n,
const f_cnt_t tfp = _n->totalFramesPlayed();
const int samplerate = engine::mixer()->processingSampleRate();
const fpp_t frames = _n->framesLeftForCurrentPeriod();
const f_cnt_t offset = _n->noteOffset();
int data = 0;
int freq = _n->frequency();
@@ -400,12 +401,12 @@ void papuInstrument::playNote( NotePlayHandle * _n,
for( ch_cnt_t ch = 0; ch < DEFAULT_CHANNELS; ++ch )
{
sample_t s = float(buf[frame*2+ch])/32768.0;
_working_buffer[frames-framesleft+frame][ch] = s;
_working_buffer[frames-framesleft+frame+offset][ch] = s;
}
}
framesleft -= count;
}
instrumentTrack()->processAudioBuffer( _working_buffer, frames, _n );
instrumentTrack()->processAudioBuffer( _working_buffer, frames + offset, _n );
}

5
plugins/patman/patman.cpp
View File

@@ -138,6 +138,7 @@ void patmanInstrument::playNote( NotePlayHandle * _n,
}
const fpp_t frames = _n->framesLeftForCurrentPeriod();
const f_cnt_t offset = _n->noteOffset();
if( !_n->m_pluginData )
{
@@ -148,12 +149,12 @@ void patmanInstrument::playNote( NotePlayHandle * _n,
float play_freq = hdata->tuned ? _n->frequency() :
hdata->sample->frequency();
if( hdata->sample->play( _working_buffer, hdata->state, frames,
if( hdata->sample->play( _working_buffer + offset, hdata->state, frames,
play_freq, m_loopedModel.value() ? SampleBuffer::LoopOn : SampleBuffer::LoopOff ) )
{
applyRelease( _working_buffer, _n );
instrumentTrack()->processAudioBuffer( _working_buffer,
frames, _n );
frames + offset, _n );
}
}

5
plugins/sfxr/sfxr.cpp
View File

@@ -454,6 +454,7 @@ void sfxrInstrument::playNote( NotePlayHandle * _n, sampleFrame * _working_buffe
float currentSampleRate = engine::mixer()->processingSampleRate();
fpp_t frameNum = _n->framesLeftForCurrentPeriod();
const f_cnt_t offset = _n->noteOffset();
if ( _n->totalFramesPlayed() == 0 || _n->m_pluginData == NULL )
{
_n->m_pluginData = new SfxrSynth( this );
@@ -477,7 +478,7 @@ void sfxrInstrument::playNote( NotePlayHandle * _n, sampleFrame * _working_buffe
{
for( ch_cnt_t j=0; j<DEFAULT_CHANNELS; j++ )
{
_working_buffer[i][j] = pitchedBuffer[i*pitchedFrameNum/frameNum][j];
_working_buffer[i+offset][j] = pitchedBuffer[i*pitchedFrameNum/frameNum][j];
}
}
@@ -485,7 +486,7 @@ void sfxrInstrument::playNote( NotePlayHandle * _n, sampleFrame * _working_buffe
applyRelease( _working_buffer, _n );
instrumentTrack()->processAudioBuffer( _working_buffer, frameNum, _n );
instrumentTrack()->processAudioBuffer( _working_buffer, frameNum + offset, _n );
}

5
plugins/sid/sid_instrument.cpp
View File

@@ -317,6 +317,7 @@ void sidInstrument::playNote( NotePlayHandle * _n,
_n->m_pluginData = sid;
}
const fpp_t frames = _n->framesLeftForCurrentPeriod();
const f_cnt_t offset = _n->noteOffset();
cSID *sid = static_cast<cSID *>( _n->m_pluginData );
int delta_t = clockrate * frames / samplerate + 4;
@@ -430,11 +431,11 @@ void sidInstrument::playNote( NotePlayHandle * _n,
sample_t s = float(buf[frame])/32768.0;
for( ch_cnt_t ch = 0; ch < DEFAULT_CHANNELS; ++ch )
{
_working_buffer[frame][ch] = s;
_working_buffer[frame+offset][ch] = s;
}
}
instrumentTrack()->processAudioBuffer( _working_buffer, frames, _n );
instrumentTrack()->processAudioBuffer( _working_buffer, frames + offset, _n );
}

5
plugins/stk/mallets/mallets.cpp
View File

@@ -265,6 +265,7 @@ void malletsInstrument::playNote( NotePlayHandle * _n,
}
const fpp_t frames = _n->framesLeftForCurrentPeriod();
const f_cnt_t offset = _n->noteOffset();
malletsSynth * ps = static_cast<malletsSynth *>( _n->m_pluginData );
ps->setFrequency( freq );
@@ -274,7 +275,7 @@ void malletsInstrument::playNote( NotePlayHandle * _n,
{
add_scale = static_cast<sample_t>( m_strikeModel.value() ) * freq * 2.5f;
}
for( fpp_t frame = 0; frame < frames; ++frame )
for( fpp_t frame = offset; frame < frames + offset; ++frame )
{
_working_buffer[frame][0] = ps->nextSampleLeft() *
( m_scalers[m_presetsModel.value()] + add_scale );
@@ -282,7 +283,7 @@ void malletsInstrument::playNote( NotePlayHandle * _n,
( m_scalers[m_presetsModel.value()] + add_scale );
}
instrumentTrack()->processAudioBuffer( _working_buffer, frames, _n );
instrumentTrack()->processAudioBuffer( _working_buffer, frames + offset, _n );
}

7
plugins/triple_oscillator/TripleOscillator.cpp
View File

@@ -358,13 +358,14 @@ void TripleOscillator::playNote( NotePlayHandle * _n,
Oscillator * osc_r = static_cast<oscPtr *>( _n->m_pluginData )->oscRight;
const fpp_t frames = _n->framesLeftForCurrentPeriod();
const f_cnt_t offset = _n->noteOffset();
osc_l->update( _working_buffer, frames, 0 );
osc_r->update( _working_buffer, frames, 1 );
osc_l->update( _working_buffer + offset, frames, 0 );
osc_r->update( _working_buffer + offset, frames, 1 );
applyRelease( _working_buffer, _n );
instrumentTrack()->processAudioBuffer( _working_buffer, frames, _n );
instrumentTrack()->processAudioBuffer( _working_buffer, frames + offset, _n );
}

5
plugins/vibed/vibed.cpp
View File

@@ -302,10 +302,11 @@ void vibed::playNote( NotePlayHandle * _n, sampleFrame * _working_buffer )
}
const fpp_t frames = _n->framesLeftForCurrentPeriod();
const f_cnt_t offset = _n->noteOffset();
stringContainer * ps = static_cast<stringContainer *>(
_n->m_pluginData );
for( fpp_t i = 0; i < frames; ++i )
for( fpp_t i = offset; i < frames + offset; ++i )
{
_working_buffer[i][0] = 0.0f;
_working_buffer[i][1] = 0.0f;
@@ -324,7 +325,7 @@ void vibed::playNote( NotePlayHandle * _n, sampleFrame * _working_buffer )
}
}
instrumentTrack()->processAudioBuffer( _working_buffer, frames, _n );
instrumentTrack()->processAudioBuffer( _working_buffer, frames + offset, _n );
}

12
plugins/watsyn/Watsyn.cpp
View File

@@ -343,6 +343,8 @@ void WatsynInstrument::playNote( NotePlayHandle * _n,
}
const fpp_t frames = _n->framesLeftForCurrentPeriod();
const f_cnt_t offset = _n->noteOffset();
sampleFrame * buffer = _working_buffer + offset;
WatsynObject * w = static_cast<WatsynObject *>( _n->m_pluginData );
@@ -424,9 +426,9 @@ void WatsynInstrument::playNote( NotePlayHandle * _n,
const float amix = 1.0 - bmix;
// mix a/b streams according to mixing knob
_working_buffer[f][0] = ( abuf[f][0] * amix ) +
buffer[f][0] = ( abuf[f][0] * amix ) +
( bbuf[f][0] * bmix );
_working_buffer[f][1] = ( abuf[f][1] * amix ) +
buffer[f][1] = ( abuf[f][1] * amix ) +
( bbuf[f][1] * bmix );
}
}
@@ -440,16 +442,16 @@ void WatsynInstrument::playNote( NotePlayHandle * _n,
for( fpp_t f=0; f < frames; f++ )
{
// mix a/b streams according to mixing knob
_working_buffer[f][0] = ( abuf[f][0] * amix ) +
buffer[f][0] = ( abuf[f][0] * amix ) +
( bbuf[f][0] * bmix );
_working_buffer[f][1] = ( abuf[f][1] * amix ) +
buffer[f][1] = ( abuf[f][1] * amix ) +
( bbuf[f][1] * bmix );
}
}
applyRelease( _working_buffer, _n );
instrumentTrack()->processAudioBuffer( _working_buffer, frames, _n );
instrumentTrack()->processAudioBuffer( _working_buffer, frames + offset, _n );
}

77
src/core/AutomatableModel.cpp
View File

@@ -30,7 +30,7 @@
#include "lmms_math.h"
float AutomatableModel::s_copiedValue = 0;
long AutomatableModel::s_periodCounter = 0;
@@ -51,7 +51,9 @@ AutomatableModel::AutomatableModel( DataType type,
m_hasStrictStepSize( false ),
m_hasLinkedModels( false ),
m_controllerConnection( NULL ),
m_valueBuffer( static_cast<int>( engine::mixer()->framesPerPeriod() ) )
m_valueBuffer( static_cast<int>( engine::mixer()->framesPerPeriod() ) ),
m_lastUpdatedPeriod( -1 ),
m_hasSampleExactData( false )
{
setInitValue( val );
@@ -86,39 +88,6 @@ bool AutomatableModel::isAutomated() const
return AutomationPattern::isAutomated( this );
}
bool AutomatableModel::hasSampleExactData() const
{
// if a controller is connected...
if( m_controllerConnection != NULL )
{
// ...and is sample-exact, then return true
if( m_controllerConnection->getController()->isSampleExact() )
{
return true;
}
}
// check also the same for the first linked model
if( hasLinkedModels() )
{
AutomatableModel* lm = m_linkedModels.first();
if( lm->m_controllerConnection != NULL )
{
if( lm->m_controllerConnection->getController()->isSampleExact() )
{
return true;
}
}
}
// if we have values we can interpolate return true
if( m_oldValue != m_value )
{
return true;
}
// otherwise, return false
return false;
}
void AutomatableModel::saveSettings( QDomDocument& doc, QDomElement& element, const QString& name )
{
@@ -554,6 +523,23 @@ float AutomatableModel::controllerValue( int frameOffset ) const
ValueBuffer * AutomatableModel::valueBuffer()
{
// if we've already calculated the valuebuffer this period, return the cached buffer
if( m_lastUpdatedPeriod == s_periodCounter )
{
return m_hasSampleExactData
? &m_valueBuffer
: NULL;
}
QMutexLocker m( &m_valueBufferMutex );
if( m_lastUpdatedPeriod == s_periodCounter )
{
return m_hasSampleExactData
? &m_valueBuffer
: NULL;
}
float val = m_value; // make sure our m_value doesn't change midway
ValueBuffer * vb;
if( m_controllerConnection && m_controllerConnection->getController()->isSampleExact() )
{
@@ -581,6 +567,8 @@ ValueBuffer * AutomatableModel::valueBuffer()
"lacks implementation for a scale type");
break;
}
m_lastUpdatedPeriod = s_periodCounter;
m_hasSampleExactData = true;
return &m_valueBuffer;
}
}
@@ -598,20 +586,25 @@ ValueBuffer * AutomatableModel::valueBuffer()
{
nvalues[i] = fittedValue( values[i], false );
}
m_lastUpdatedPeriod = s_periodCounter;
m_hasSampleExactData = true;
return &m_valueBuffer;
}
if( m_oldValue != m_value )
if( m_oldValue != val )
{
m_valueBuffer.interpolate( m_oldValue, m_value );
m_oldValue = m_value;
m_valueBuffer.interpolate( m_oldValue, val );
m_oldValue = val;
m_lastUpdatedPeriod = s_periodCounter;
m_hasSampleExactData = true;
return &m_valueBuffer;
}
// if we have no sample-exact source for a ValueBuffer, create one and fill it with current value
// ideally, recipients should check first if we hasSampleExactData before fetching ValueBuffers
m_valueBuffer.fill( m_value );
return &m_valueBuffer;
// if we have no sample-exact source for a ValueBuffer, return NULL to signify that no data is available at the moment
// in which case the recipient knows to use the static value() instead
m_lastUpdatedPeriod = s_periodCounter;
m_hasSampleExactData = false;
return NULL;
}

12
src/core/FxMixer.cpp
View File

@@ -126,12 +126,8 @@ void FxChannel::doProcessing( sampleFrame * _buf )
if( sender->m_hasInput || sender->m_stillRunning )
{
// figure out if we're getting sample-exact input
ValueBuffer * sendBuf = sendModel->hasSampleExactData()
? sendModel->valueBuffer()
: NULL;
ValueBuffer * volBuf = sender->m_volumeModel.hasSampleExactData()
? sender->m_volumeModel.valueBuffer()
: NULL;
ValueBuffer * sendBuf = sendModel->valueBuffer();
ValueBuffer * volBuf = sender->m_volumeModel.valueBuffer();
// mix it's output with this one's output
sampleFrame * ch_buf = sender->m_buffer;
@@ -526,9 +522,7 @@ void FxMixer::masterMix( sampleFrame * _buf )
//m_sendsMutex.unlock();
// handle sample-exact data in master volume fader
ValueBuffer * volBuf = m_fxChannels[0]->m_volumeModel.hasSampleExactData()
? m_fxChannels[0]->m_volumeModel.valueBuffer()
: NULL;
ValueBuffer * volBuf = m_fxChannels[0]->m_volumeModel.valueBuffer();
if( volBuf )
{

2
src/core/InstrumentSoundShaping.cpp
View File

@@ -133,7 +133,7 @@ void InstrumentSoundShaping::processAudioBuffer( sampleFrame* buffer,
if( n->isReleased() == false )
{
envReleaseBegin += engine::mixer()->framesPerPeriod();
envReleaseBegin += frames;
}
// because of optimizations, there's special code for several cases:

2
src/core/LfoController.cpp
View File

@@ -109,7 +109,7 @@ void LfoController::updateValueBuffer()
? m_sampleFunction( phase )
: m_userDefSampleBuffer->userWaveSample( phase );
values[i] = m_baseModel.value() + ( m_amountModel.value() * currentSample / 2.0f );
values[i] = qBound( 0.0f, m_baseModel.value() + ( m_amountModel.value() * currentSample / 2.0f ), 1.0f );
phase += 1.0 / m_duration;
}

51
src/core/Mixer.cpp
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@@ -418,6 +418,7 @@ const surroundSampleFrame * Mixer::renderNextBuffer()
// and trigger LFOs
EnvelopeAndLfoParameters::instances()->trigger();
Controller::triggerFrameCounter();
AutomatableModel::incrementPeriodCounter();
const float new_cpu_load = timer.elapsed() / 10000.0f *
processingSampleRate() / m_framesPerPeriod;
@@ -451,44 +452,40 @@ void Mixer::clear()
void Mixer::bufferToPort( const sampleFrame * _buf,
const fpp_t _frames,
const f_cnt_t _offset,
stereoVolumeVector _vv,
AudioPort * _port )
void Mixer::bufferToPort( const sampleFrame * buf,
const fpp_t frames,
stereoVolumeVector vv,
AudioPort * port )
{
const int start_frame = _offset % m_framesPerPeriod;
int end_frame = start_frame + _frames;
const int loop1_frame = qMin<int>( end_frame, m_framesPerPeriod );
const int loop1_frame = qMin<int>( frames, m_framesPerPeriod );
_port->lockFirstBuffer();
MixHelpers::addMultipliedStereo( _port->firstBuffer()+start_frame, // dst
_buf, // src
_vv.vol[0], _vv.vol[1], // coeff left/right
loop1_frame - start_frame ); // frame count
_port->unlockFirstBuffer();
port->lockFirstBuffer();
MixHelpers::addMultipliedStereo( port->firstBuffer(), // dst
buf, // src
vv.vol[0], vv.vol[1], // coeff left/right
loop1_frame ); // frame count
port->unlockFirstBuffer();
_port->lockSecondBuffer();
if( end_frame > m_framesPerPeriod )
if( frames > m_framesPerPeriod )
{
const int frames_done = m_framesPerPeriod - start_frame;
end_frame -= m_framesPerPeriod;
end_frame = qMin<int>( end_frame, m_framesPerPeriod );
port->lockSecondBuffer();
const fpp_t framesLeft = qMin<int>( frames - m_framesPerPeriod, m_framesPerPeriod );
MixHelpers::addMultipliedStereo( _port->secondBuffer(), // dst
_buf+frames_done, // src
_vv.vol[0], _vv.vol[1], // coeff left/right
end_frame ); // frame count
MixHelpers::addMultipliedStereo( port->secondBuffer(), // dst
buf + m_framesPerPeriod, // src
vv.vol[0], vv.vol[1], // coeff left/right
framesLeft ); // frame count
// we used both buffers so set flags
_port->m_bufferUsage = AudioPort::BothBuffers;
port->m_bufferUsage = AudioPort::BothBuffers;
port->unlockSecondBuffer();
}
else if( _port->m_bufferUsage == AudioPort::NoUsage )
else if( port->m_bufferUsage == AudioPort::NoUsage )
{
// only first buffer touched
_port->m_bufferUsage = AudioPort::FirstBuffer;
port->m_bufferUsage = AudioPort::FirstBuffer;
}
_port->unlockSecondBuffer();
}

58
src/core/NotePlayHandle.cpp
View File

@@ -61,7 +61,6 @@ NotePlayHandle::NotePlayHandle( InstrumentTrack* instrumentTrack,
m_framesBeforeRelease( 0 ),
m_releaseFramesToDo( 0 ),
m_releaseFramesDone( 0 ),
m_scheduledNoteOff( -1 ),
m_released( false ),
m_hasParent( parent != NULL ),
m_hadChildren( false ),
@@ -119,13 +118,6 @@ NotePlayHandle::NotePlayHandle( InstrumentTrack* instrumentTrack,
NotePlayHandle::~NotePlayHandle()
{
noteOff( 0 );
if( m_scheduledNoteOff >= 0 ) // ensure that scheduled noteoffs get triggered if somehow the nph got destructed prematurely
{
m_instrumentTrack->processOutEvent(
MidiEvent( MidiNoteOff, midiChannel(), midiKey(), 0 ),
MidiTime::fromFrames( m_scheduledNoteOff, engine::framesPerTick() ),
m_scheduledNoteOff );
}
if( hasParent() == false )
{
@@ -190,23 +182,20 @@ int NotePlayHandle::midiKey() const
void NotePlayHandle::play( sampleFrame * _working_buffer )
{
if( m_scheduledNoteOff >= 0 ) // always trigger scheduled noteoffs, because they're only scheduled if the note is released
{
m_instrumentTrack->processOutEvent(
MidiEvent( MidiNoteOff, midiChannel(), midiKey(), 0 ),
MidiTime::fromFrames( m_scheduledNoteOff, engine::framesPerTick() ),
m_scheduledNoteOff );
m_scheduledNoteOff = -1;
}
if( m_muted )
{
return;
}
// number of frames that can be played this period
f_cnt_t framesThisPeriod = m_totalFramesPlayed == 0
? engine::mixer()->framesPerPeriod() - offset()
: engine::mixer()->framesPerPeriod();
// check if we start release during this period
if( m_released == false &&
instrumentTrack()->isSustainPedalPressed() == false &&
m_totalFramesPlayed + engine::mixer()->framesPerPeriod() > m_frames )
m_totalFramesPlayed + framesThisPeriod > m_frames )
{
noteOff( m_frames - m_totalFramesPlayed );
}
@@ -216,13 +205,18 @@ void NotePlayHandle::play( sampleFrame * _working_buffer )
// decreasing release of an instrument-track while the note is active
if( framesLeft() > 0 )
{
// clear offset frames if we're at the first period
if( m_totalFramesPlayed == 0 )
{
memset( _working_buffer, 0, sizeof( sampleFrame ) * offset() );
}
// play note!
m_instrumentTrack->playNote( this, _working_buffer );
}
if( m_released )
{
f_cnt_t todo = engine::mixer()->framesPerPeriod();
f_cnt_t todo = framesThisPeriod;
// if this note is base-note for arpeggio, always set
// m_releaseFramesToDo to bigger value than m_releaseFramesDone
@@ -238,8 +232,7 @@ void NotePlayHandle::play( sampleFrame * _working_buffer )
{
// yes, then look whether these samples can be played
// within one audio-buffer
if( m_framesBeforeRelease <=
engine::mixer()->framesPerPeriod() )
if( m_framesBeforeRelease <= framesThisPeriod )
{
// yes, then we did less releaseFramesDone
todo -= m_framesBeforeRelease;
@@ -251,8 +244,7 @@ void NotePlayHandle::play( sampleFrame * _working_buffer )
// and wait for next loop... (we're not in
// release-phase yet)
todo = 0;
m_framesBeforeRelease -=
engine::mixer()->framesPerPeriod();
m_framesBeforeRelease -= framesThisPeriod;
}
}
// look whether we're in release-phase
@@ -290,7 +282,7 @@ void NotePlayHandle::play( sampleFrame * _working_buffer )
}
// update internal data
m_totalFramesPlayed += engine::mixer()->framesPerPeriod();
m_totalFramesPlayed += framesThisPeriod;
}
@@ -318,6 +310,10 @@ f_cnt_t NotePlayHandle::framesLeft() const
fpp_t NotePlayHandle::framesLeftForCurrentPeriod() const
{
if( m_totalFramesPlayed == 0 )
{
return (fpp_t) qMin<f_cnt_t>( framesLeft(), engine::mixer()->framesPerPeriod() - offset() );
}
return (fpp_t) qMin<f_cnt_t>( framesLeft(), engine::mixer()->framesPerPeriod() );
}
@@ -352,18 +348,10 @@ void NotePlayHandle::noteOff( const f_cnt_t _s )
if( hasParent() || ! m_instrumentTrack->isArpeggioEnabled() )
{
// send MidiNoteOff event
f_cnt_t realOffset = offset() + _s; // get actual frameoffset of release, in global time
if( realOffset < engine::mixer()->framesPerPeriod() ) // if release happens during this period, trigger midievent
{
m_instrumentTrack->processOutEvent(
m_instrumentTrack->processOutEvent(
MidiEvent( MidiNoteOff, midiChannel(), midiKey(), 0 ),
MidiTime::fromFrames( realOffset, engine::framesPerTick() ),
realOffset );
}
else // if release flows over to next period, use m_scheduledNoteOff to trigger it later
{
m_scheduledNoteOff = realOffset - engine::mixer()->framesPerPeriod();
}
MidiTime::fromFrames( _s, engine::framesPerTick() ),
_s );
}
// inform attached components about MIDI finished (used for recording in Piano Roll)

20
src/core/SamplePlayHandle.cpp
View File

@@ -96,7 +96,7 @@ SamplePlayHandle::~SamplePlayHandle()
void SamplePlayHandle::play( sampleFrame * _working_buffer )
void SamplePlayHandle::play( sampleFrame * buffer )
{
//play( 0, _try_parallelizing );
if( framesDone() >= totalFrames() )
@@ -104,17 +104,27 @@ void SamplePlayHandle::play( sampleFrame * _working_buffer )
return;
}
const fpp_t frames = engine::mixer()->framesPerPeriod();
sampleFrame * workingBuffer = buffer;
const fpp_t fpp = engine::mixer()->framesPerPeriod();
f_cnt_t frames = fpp;
// apply offset for the first period
if( framesDone() == 0 )
{
buffer += offset();
frames -= offset();
}
if( !( m_track && m_track->isMuted() )
&& !( m_bbTrack && m_bbTrack->isMuted() ) )
{
stereoVolumeVector v =
{ { m_volumeModel->value() / DefaultVolume,
m_volumeModel->value() / DefaultVolume } };
m_sampleBuffer->play( _working_buffer, &m_state, frames,
m_sampleBuffer->play( workingBuffer, &m_state, frames,
BaseFreq );
engine::mixer()->bufferToPort( _working_buffer, frames,
offset(), v, m_audioPort );
engine::mixer()->bufferToPort( buffer, fpp,
v, m_audioPort );
}
m_frame += frames;

47
src/tracks/InstrumentTrack.cpp
View File

@@ -76,7 +76,7 @@
#include "tab_widget.h"
#include "tooltip.h"
#include "track_label_button.h"
#include "ValueBuffer.h"
const char * volume_help = QT_TRANSLATE_NOOP( "InstrumentTrack",
@@ -186,33 +186,58 @@ void InstrumentTrack::processAudioBuffer( sampleFrame* buf, const fpp_t frames,
// now
m_audioPort.effects()->startRunning();
float v_scale = (float) getVolume() / DefaultVolume;
// get volume knob data
static const float DefaultVolumeRatio = 1.0f / DefaultVolume;
ValueBuffer * volBuf = m_volumeModel.valueBuffer();
float v_scale = volBuf
? 1.0f
: getVolume() * DefaultVolumeRatio;
// instruments using instrument-play-handles will call this method
// without any knowledge about notes, so they pass NULL for n, which
// is no problem for us since we just bypass the envelopes+LFOs
if( m_instrument->flags().testFlag( Instrument::IsSingleStreamed ) == false && n != NULL )
{
m_soundShaping.processAudioBuffer( buf, frames, n );
v_scale *= ( (float) n->getVolume() / DefaultVolume );
const f_cnt_t offset = n->noteOffset();
m_soundShaping.processAudioBuffer( buf + offset, frames - offset, n );
v_scale *= ( (float) n->getVolume() * DefaultVolumeRatio );
}
m_audioPort.setNextFxChannel( m_effectChannelModel.value() );
int framesToMix = frames;
int offset = 0;
int panning = m_panningModel.value();
// get panning knob data
ValueBuffer * panBuf = m_panningModel.valueBuffer();
int panning = panBuf
? 0
: m_panningModel.value();
if( n )
{
framesToMix = qMin<f_cnt_t>( n->framesLeftForCurrentPeriod(), framesToMix );
offset = n->offset();
panning += n->getPanning();
panning = tLimit<int>( panning, PanningLeft, PanningRight );
}
engine::mixer()->bufferToPort( buf, framesToMix, offset, panningToVolumeVector( panning, v_scale ), &m_audioPort );
// apply sample-exact volume/panning data
if( volBuf )
{
for( f_cnt_t f = 0; f < frames; ++f )
{
float v = volBuf->values()[ f ] * 0.01f;
buf[f][0] *= v;
buf[f][1] *= v;
}
}
if( panBuf )
{
for( f_cnt_t f = 0; f < frames; ++f )
{
float p = panBuf->values()[ f ] * 0.01f;
buf[f][0] *= ( p <= 0 ? 1.0f : 1.0f - p );
buf[f][1] *= ( p >= 0 ? 1.0f : 1.0f + p );
}
}
engine::mixer()->bufferToPort( buf, frames, panningToVolumeVector( panning, v_scale ), &m_audioPort );
}