bool BassBoosterEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames )
{
if( !isEnabled() || !isRunning () )
{
return( false );
}
// check out changed controls
if( m_frequencyChangeNeeded || m_bbControls.m_freqModel.isValueChanged() )
{
changeFrequency();
m_frequencyChangeNeeded = false;
}
if( m_bbControls.m_gainModel.isValueChanged() ) { changeGain(); }
if( m_bbControls.m_ratioModel.isValueChanged() ) { changeRatio(); }
float gain = m_bbControls.m_gainModel.value();
ValueBuffer *gainBuffer = m_bbControls.m_gainModel.valueBuffer();
int gainInc = gainBuffer ? 1 : 0;
float *gainPtr = gainBuffer ? &( gainBuffer->values()[ 0 ] ) : &gain;
double outSum = 0.0;
const float d = dryLevel();
const float w = wetLevel();
if( gainBuffer )
{
//process period using sample exact data
for( fpp_t f = 0; f < frames; ++f )
{
m_bbFX.leftFX().setGain( *gainPtr );
m_bbFX.rightFX().setGain( *gainPtr );
outSum += buf[f][0]*buf[f][0] + buf[f][1]*buf[f][1];
sample_t s[2] = { buf[f][0], buf[f][1] };
m_bbFX.nextSample( s[0], s[1] );
buf[f][0] = d * buf[f][0] + w * s[0];
buf[f][1] = d * buf[f][1] + w * s[1];
gainPtr += gainInc;
}
} else
{
//process period without sample exact data
m_bbFX.leftFX().setGain( *gainPtr );
m_bbFX.rightFX().setGain( *gainPtr );
for( fpp_t f = 0; f < frames; ++f )
{
outSum += buf[f][0]*buf[f][0] + buf[f][1]*buf[f][1];
sample_t s[2] = { buf[f][0], buf[f][1] };
m_bbFX.nextSample( s[0], s[1] );
buf[f][0] = d * buf[f][0] + w * s[0];
buf[f][1] = d * buf[f][1] + w * s[1];
}
}
checkGate( outSum / frames );
return isRunning();
}
void AudioPort::doProcessing()
{
if( m_mutedModel && m_mutedModel->value() )
{
return;
}
const fpp_t fpp = Engine::mixer()->framesPerPeriod();
m_portBuffer = BufferManager::acquire(); // get buffer for processing
Engine::mixer()->clearAudioBuffer( m_portBuffer, fpp ); // clear the audioport buffer so we can use it
//qDebug( "Playhandles: %d", m_playHandles.size() );
foreach( PlayHandle * ph, m_playHandles ) // now we mix all playhandle buffers into the audioport buffer
{
if( ph->buffer() )
{
if( ph->usesBuffer() )
{
m_bufferUsage = true;
MixHelpers::add( m_portBuffer, ph->buffer(), fpp );
}
ph->releaseBuffer(); // gets rid of playhandle's buffer and sets
// pointer to null, so if it doesn't get re-acquired we know to skip it next time
}
}
if( m_bufferUsage )
{
// handle volume and panning
// has both vol and pan models
if( m_volumeModel && m_panningModel )
{
ValueBuffer * volBuf = m_volumeModel->valueBuffer();
ValueBuffer * panBuf = m_panningModel->valueBuffer();
// both vol and pan have s.ex.data:
if( volBuf && panBuf )
{
for( f_cnt_t f = 0; f < fpp; ++f )
{
float v = volBuf->values()[ f ] * 0.01f;
float p = panBuf->values()[ f ] * 0.01f;
m_portBuffer[f][0] *= ( p <= 0 ? 1.0f : 1.0f - p ) * v;
m_portBuffer[f][1] *= ( p >= 0 ? 1.0f : 1.0f + p ) * v;
}
}
// only vol has s.ex.data:
else if( volBuf )
{
float p = m_panningModel->value() * 0.01f;
float l = ( p <= 0 ? 1.0f : 1.0f - p );
float r = ( p >= 0 ? 1.0f : 1.0f + p );
for( f_cnt_t f = 0; f < fpp; ++f )
{
float v = volBuf->values()[ f ] * 0.01f;
m_portBuffer[f][0] *= v * l;
m_portBuffer[f][1] *= v * r;
}
}
// only pan has s.ex.data:
else if( panBuf )
{
float v = m_volumeModel->value() * 0.01f;
for( f_cnt_t f = 0; f < fpp; ++f )
{
float p = panBuf->values()[ f ] * 0.01f;
m_portBuffer[f][0] *= ( p <= 0 ? 1.0f : 1.0f - p ) * v;
m_portBuffer[f][1] *= ( p >= 0 ? 1.0f : 1.0f + p ) * v;
}
}
// neither has s.ex.data:
else
{
float p = m_panningModel->value() * 0.01f;
float v = m_volumeModel->value() * 0.01f;
for( f_cnt_t f = 0; f < fpp; ++f )
{
m_portBuffer[f][0] *= ( p <= 0 ? 1.0f : 1.0f - p ) * v;
m_portBuffer[f][1] *= ( p >= 0 ? 1.0f : 1.0f + p ) * v;
}
}
}
// has vol model only
else if( m_volumeModel )
{
ValueBuffer * volBuf = m_volumeModel->valueBuffer();
if( volBuf )
{
for( f_cnt_t f = 0; f < fpp; ++f )
{
float v = volBuf->values()[ f ] * 0.01f;
m_portBuffer[f][0] *= v;
m_portBuffer[f][1] *= v;
}
}
else
{
float v = m_volumeModel->value() * 0.01f;
for( f_cnt_t f = 0; f < fpp; ++f )
{
m_portBuffer[f][0] *= v;
m_portBuffer[f][1] *= v;
}
}
}
}
// as of now there's no situation where we only have panning model but no volume model
// if we have neither, we don't have to do anything here - just pass the audio as is
// handle effects
const bool me = processEffects();
if( me || m_bufferUsage )
{
Engine::fxMixer()->mixToChannel( m_portBuffer, m_nextFxChannel ); // send output to fx mixer
// TODO: improve the flow here - convert to pull model
m_bufferUsage = false;
}
BufferManager::release( m_portBuffer ); // release buffer, we don't need it anymore
}
bool DelayEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames )
{
if( !isEnabled() || !isRunning () )
{
return( false );
}
double outSum = 0.0;
const float sr = Engine::mixer()->processingSampleRate();
const float d = dryLevel();
const float w = wetLevel();
sample_t dryS[2];
float lPeak = 0.0;
float rPeak = 0.0;
float length = m_delayControls.m_delayTimeModel.value();
float amplitude = m_delayControls.m_lfoAmountModel.value() * sr;
float lfoTime = 1.0 / m_delayControls.m_lfoTimeModel.value();
float feedback = m_delayControls.m_feedbackModel.value();
ValueBuffer *lengthBuffer = m_delayControls.m_delayTimeModel.valueBuffer();
ValueBuffer *feedbackBuffer = m_delayControls.m_feedbackModel.valueBuffer();
ValueBuffer *lfoTimeBuffer = m_delayControls.m_lfoTimeModel.valueBuffer();
ValueBuffer *lfoAmountBuffer = m_delayControls.m_lfoAmountModel.valueBuffer();
int lengthInc = lengthBuffer ? 1 : 0;
int amplitudeInc = lfoAmountBuffer ? 1 : 0;
int lfoTimeInc = lfoTimeBuffer ? 1 : 0;
int feedbackInc = feedbackBuffer ? 1 : 0;
float *lengthPtr = lengthBuffer ? &( lengthBuffer->values()[ 0 ] ) : &length;
float *amplitudePtr = lfoAmountBuffer ? &( lfoAmountBuffer->values()[ 0 ] ) : &litude;
float *lfoTimePtr = lfoTimeBuffer ? &( lfoTimeBuffer->values()[ 0 ] ) : &lfoTime;
float *feedbackPtr = feedbackBuffer ? &( feedbackBuffer->values()[ 0 ] ) : &feedback;
if( m_delayControls.m_outGainModel.isValueChanged() )
{
m_outGain = dbfsToAmp( m_delayControls.m_outGainModel.value() );
}
int sampleLength;
for( fpp_t f = 0; f < frames; ++f )
{
dryS[0] = buf[f][0];
dryS[1] = buf[f][1];
m_delay->setFeedback( *feedbackPtr );
m_lfo->setFrequency( *lfoTimePtr );
sampleLength = *lengthPtr * Engine::mixer()->processingSampleRate();
m_currentLength = sampleLength;
m_delay->setLength( m_currentLength + ( *amplitudePtr * ( float )m_lfo->tick() ) );
m_delay->tick( buf[f] );
buf[f][0] *= m_outGain;
buf[f][1] *= m_outGain;
lPeak = buf[f][0] > lPeak ? buf[f][0] : lPeak;
rPeak = buf[f][1] > rPeak ? buf[f][1] : rPeak;
buf[f][0] = ( d * dryS[0] ) + ( w * buf[f][0] );
buf[f][1] = ( d * dryS[1] ) + ( w * buf[f][1] );
outSum += buf[f][0]*buf[f][0] + buf[f][1]*buf[f][1];
lengthPtr += lengthInc;
amplitudePtr += amplitudeInc;
lfoTimePtr += lfoTimeInc;
feedbackPtr += feedbackInc;
}
checkGate( outSum / frames );
m_delayControls.m_outPeakL = lPeak;
m_delayControls.m_outPeakR = rPeak;
return isRunning();
}
void FxMixer::masterMix( sampleFrame * _buf )
{
const int fpp = Engine::mixer()->framesPerPeriod();
if( m_sendsMutex.tryLock() )
{
// add the channels that have no dependencies (no incoming senders, ie. no receives)
// to the jobqueue. The channels that have receives get added when their senders get processed, which
// is detected by dependency counting.
// also instantly add all muted channels as they don't need to care about their senders, and can just increment the deps of
// their recipients right away.
MixerWorkerThread::resetJobQueue( MixerWorkerThread::JobQueue::Dynamic );
for( FxChannel * ch : m_fxChannels )
{
ch->m_muted = ch->m_muteModel.value();
if( ch->m_muted ) // instantly "process" muted channels
{
ch->processed();
ch->done();
}
else if( ch->m_receives.size() == 0 )
{
ch->m_queued = true;
MixerWorkerThread::addJob( ch );
}
}
while( m_fxChannels[0]->state() != ThreadableJob::Done )
{
MixerWorkerThread::startAndWaitForJobs();
}
m_sendsMutex.unlock();
}
// handle sample-exact data in master volume fader
ValueBuffer * volBuf = m_fxChannels[0]->m_volumeModel.valueBuffer();
if( volBuf )
{
for( int f = 0; f < fpp; f++ )
{
m_fxChannels[0]->m_buffer[f][0] *= volBuf->values()[f];
m_fxChannels[0]->m_buffer[f][1] *= volBuf->values()[f];
}
}
const float v = volBuf
? 1.0f
: m_fxChannels[0]->m_volumeModel.value();
MixHelpers::addSanitizedMultiplied( _buf, m_fxChannels[0]->m_buffer, v, fpp );
// clear all channel buffers and
// reset channel process state
for( int i = 0; i < numChannels(); ++i)
{
BufferManager::clear( m_fxChannels[i]->m_buffer,
Engine::mixer()->framesPerPeriod() );
m_fxChannels[i]->reset();
m_fxChannels[i]->m_queued = false;
// also reset hasInput
m_fxChannels[i]->m_hasInput = false;
m_fxChannels[i]->m_dependenciesMet = 0;
}
}
bool AmplifierEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames )
{
if( !isEnabled() || !isRunning () )
{
return( false );
}
double outSum = 0.0;
const float d = dryLevel();
const float w = wetLevel();
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 )
{
// qDebug( "offset %d, value %f", f, m_ampControls.m_volumeModel.value( f ) );
sample_t s[2] = { buf[f][0], buf[f][1] };
// vol knob
if( volBuf )
{
s[0] *= volBuf->values()[ f ] * 0.01f;
s[1] *= volBuf->values()[ f ] * 0.01f;
}
else
{
s[0] *= m_ampControls.m_volumeModel.value() * 0.01f;
s[1] *= m_ampControls.m_volumeModel.value() * 0.01f;
}
// convert pan values to left/right values
const float pan = panBuf
? panBuf->values()[ f ]
: m_ampControls.m_panModel.value();
const float left1 = pan <= 0
? 1.0
: 1.0 - pan * 0.01f;
const float right1 = pan >= 0
? 1.0
: 1.0 + pan * 0.01f;
// second stage amplification
const float left2 = leftBuf
? leftBuf->values()[ f ]
: m_ampControls.m_leftModel.value();
const float right2 = rightBuf
? rightBuf->values()[ f ]
: m_ampControls.m_rightModel.value();
s[0] *= left1 * left2 * 0.01;
s[1] *= right1 * right2 * 0.01;
buf[f][0] = d * buf[f][0] + w * s[0];
buf[f][1] = d * buf[f][1] + w * s[1];
outSum += buf[f][0]*buf[f][0] + buf[f][1]*buf[f][1];
}
checkGate( outSum / frames );
return isRunning();
}
bool EqEffect::processAudioBuffer( sampleFrame *buf, const fpp_t frames )
{
// setup sample exact controls
float hpRes = m_eqControls.m_hpResModel.value();
float lowShelfRes = m_eqControls.m_lowShelfResModel.value();
float para1Bw = m_eqControls.m_para1BwModel.value();
float para2Bw = m_eqControls.m_para2BwModel.value();
float para3Bw = m_eqControls.m_para3BwModel.value();
float para4Bw = m_eqControls.m_para4BwModel.value();
float highShelfRes = m_eqControls.m_highShelfResModel.value();
float lpRes = m_eqControls.m_lpResModel.value();
float hpFreq = m_eqControls.m_hpFeqModel.value();
float lowShelfFreq = m_eqControls.m_lowShelfFreqModel.value();
float para1Freq = m_eqControls.m_para1FreqModel.value();
float para2Freq = m_eqControls.m_para2FreqModel.value();
float para3Freq = m_eqControls.m_para3FreqModel.value();
float para4Freq = m_eqControls.m_para4FreqModel.value();
float highShelfFreq = m_eqControls.m_highShelfFreqModel.value();
float lpFreq = m_eqControls.m_lpFreqModel.value();
ValueBuffer *hpResBuffer = m_eqControls.m_hpResModel.valueBuffer();
ValueBuffer *lowShelfResBuffer = m_eqControls.m_lowShelfResModel.valueBuffer();
ValueBuffer *para1BwBuffer = m_eqControls.m_para1BwModel.valueBuffer();
ValueBuffer *para2BwBuffer = m_eqControls.m_para2BwModel.valueBuffer();
ValueBuffer *para3BwBuffer = m_eqControls.m_para3BwModel.valueBuffer();
ValueBuffer *para4BwBuffer = m_eqControls.m_para4BwModel.valueBuffer();
ValueBuffer *highShelfResBuffer = m_eqControls.m_highShelfResModel.valueBuffer();
ValueBuffer *lpResBuffer = m_eqControls.m_lpResModel.valueBuffer();
ValueBuffer *hpFreqBuffer = m_eqControls.m_hpFeqModel.valueBuffer();
ValueBuffer *lowShelfFreqBuffer = m_eqControls.m_lowShelfFreqModel.valueBuffer();
ValueBuffer *para1FreqBuffer = m_eqControls.m_para1FreqModel.valueBuffer();
ValueBuffer *para2FreqBuffer = m_eqControls.m_para2FreqModel.valueBuffer();
ValueBuffer *para3FreqBuffer = m_eqControls.m_para3FreqModel.valueBuffer();
ValueBuffer *para4FreqBuffer = m_eqControls.m_para4FreqModel.valueBuffer();
ValueBuffer *highShelfFreqBuffer = m_eqControls.m_highShelfFreqModel.valueBuffer();
ValueBuffer *lpFreqBuffer = m_eqControls.m_lpFreqModel.valueBuffer();
int hpResInc = hpResBuffer ? 1 : 0;
int lowShelfResInc = lowShelfResBuffer ? 1 : 0;
int para1BwInc = para1BwBuffer ? 1 : 0;
int para2BwInc = para2BwBuffer ? 1 : 0;
int para3BwInc = para3BwBuffer ? 1 : 0;
int para4BwInc = para4BwBuffer ? 1 : 0;
int highShelfResInc = highShelfResBuffer ? 1 : 0;
int lpResInc = lpResBuffer ? 1 : 0;
int hpFreqInc = hpFreqBuffer ? 1 : 0;
int lowShelfFreqInc = lowShelfFreqBuffer ? 1 : 0;
int para1FreqInc = para1FreqBuffer ? 1 : 0;
int para2FreqInc = para2FreqBuffer ? 1 : 0;
int para3FreqInc = para3FreqBuffer ? 1 : 0;
int para4FreqInc = para4FreqBuffer ? 1 : 0;
int highShelfFreqInc = highShelfFreqBuffer ? 1 : 0;
int lpFreqInc = lpFreqBuffer ? 1 : 0;
float *hpResPtr = hpResBuffer ? &( hpResBuffer->values()[ 0 ] ) : &hpRes;
float *lowShelfResPtr = lowShelfResBuffer ? &( lowShelfResBuffer->values()[ 0 ] ) : &lowShelfRes;
float *para1BwPtr = para1BwBuffer ? &( para1BwBuffer->values()[ 0 ] ) : ¶1Bw;
float *para2BwPtr = para2BwBuffer ? &( para2BwBuffer->values()[ 0 ] ) : ¶2Bw;
float *para3BwPtr = para3BwBuffer ? &( para3BwBuffer->values()[ 0 ] ) : ¶3Bw;
float *para4BwPtr = para4BwBuffer ? &( para4BwBuffer->values()[ 0 ] ) : ¶4Bw;
float *highShelfResPtr = highShelfResBuffer ? &( highShelfResBuffer->values()[ 0 ] ) : &highShelfRes;
float *lpResPtr = lpResBuffer ? &( lpResBuffer->values()[ 0 ] ) : &lpRes;
float *hpFreqPtr = hpFreqBuffer ? &( hpFreqBuffer->values()[ 0 ] ) : &hpFreq;
float *lowShelfFreqPtr = lowShelfFreqBuffer ? &( lowShelfFreqBuffer->values()[ 0 ] ) : &lowShelfFreq;
float *para1FreqPtr = para1FreqBuffer ? &(para1FreqBuffer->values()[ 0 ] ) : ¶1Freq;
float *para2FreqPtr = para2FreqBuffer ? &(para2FreqBuffer->values()[ 0 ] ) : ¶2Freq;
float *para3FreqPtr = para3FreqBuffer ? &(para3FreqBuffer->values()[ 0 ] ) : ¶3Freq;
float *para4FreqPtr = para4FreqBuffer ? &(para4FreqBuffer->values()[ 0 ] ) : ¶4Freq;
float *hightShelfFreqPtr = highShelfFreqBuffer ? &(highShelfFreqBuffer->values()[ 0 ] ) : &highShelfFreq;
float *lpFreqPtr = lpFreqBuffer ? &(lpFreqBuffer ->values()[ 0 ] ) : &lpFreq;
bool hpActive = m_eqControls.m_hpActiveModel.value();
bool hp24Active = m_eqControls.m_hp24Model.value();
bool hp48Active = m_eqControls.m_hp48Model.value();
bool lowShelfActive = m_eqControls.m_lowShelfActiveModel.value();
bool para1Active = m_eqControls.m_para1ActiveModel.value();
bool para2Active = m_eqControls.m_para2ActiveModel.value();
bool para3Active = m_eqControls.m_para3ActiveModel.value();
bool para4Active = m_eqControls.m_para4ActiveModel.value();
bool highShelfActive = m_eqControls.m_highShelfActiveModel.value();
bool lpActive = m_eqControls.m_lpActiveModel.value();
bool lp24Active = m_eqControls.m_lp24Model.value();
bool lp48Active = m_eqControls.m_lp48Model.value();
float lowShelfGain = m_eqControls.m_lowShelfGainModel.value();
float para1Gain = m_eqControls.m_para1GainModel.value();
float para2Gain = m_eqControls.m_para2GainModel.value();
float para3Gain = m_eqControls.m_para3GainModel.value();
float para4Gain = m_eqControls.m_para4GainModel.value();
float highShelfGain = m_eqControls.m_highShelfGainModel.value();
if( !isEnabled() || !isRunning () )
{
return( false );
}
if( m_eqControls.m_outGainModel.isValueChanged() )
{
m_outGain = dbfsToAmp(m_eqControls.m_outGainModel.value());
}
if( m_eqControls.m_inGainModel.isValueChanged() )
{
m_inGain = dbfsToAmp(m_eqControls.m_inGainModel.value());
}
m_eqControls.m_inProgress = true;
double outSum = 0.0;
for( fpp_t f = 0; f < frames; ++f )
{
outSum += buf[f][0]*buf[f][0] + buf[f][1]*buf[f][1];
}
const float outGain = m_outGain;
const int sampleRate = Engine::mixer()->processingSampleRate();
sampleFrame m_inPeak = { 0, 0 };
if(m_eqControls.m_analyseInModel.value( true ) && outSum > 0 )
{
m_eqControls.m_inFftBands.analyze( buf, frames );
}
else
{
m_eqControls.m_inFftBands.clear();
}
gain( buf, frames, m_inGain, &m_inPeak );
m_eqControls.m_inPeakL = m_eqControls.m_inPeakL < m_inPeak[0] ? m_inPeak[0] : m_eqControls.m_inPeakL;
m_eqControls.m_inPeakR = m_eqControls.m_inPeakR < m_inPeak[1] ? m_inPeak[1] : m_eqControls.m_inPeakR;
for( fpp_t f = 0; f < frames; f++)
{
if( hpActive )
{
m_hp12.setParameters( sampleRate, *hpFreqPtr, *hpResPtr, 1 );
buf[f][0] = m_hp12.update( buf[f][0], 0 );
buf[f][1] = m_hp12.update( buf[f][1], 1 );
if( hp24Active || hp48Active )
{
m_hp24.setParameters( sampleRate, *hpFreqPtr, *hpResPtr, 1 );
buf[f][0] = m_hp24.update( buf[f][0], 0 );
buf[f][1] = m_hp24.update( buf[f][1], 1 );
}
if( hp48Active )
{
m_hp480.setParameters( sampleRate, *hpFreqPtr, *hpResPtr, 1 );
buf[f][0] = m_hp480.update( buf[f][0], 0 );
buf[f][1] = m_hp480.update( buf[f][1], 1 );
m_hp481.setParameters( sampleRate, *hpFreqPtr, *hpResPtr, 1 );
buf[f][0] = m_hp481.update( buf[f][0], 0 );
buf[f][1] = m_hp481.update( buf[f][1], 1 );
}
}
if( lowShelfActive )
{
m_lowShelf.setParameters( sampleRate, *lowShelfFreqPtr, *lowShelfResPtr, lowShelfGain );
buf[f][0] = m_lowShelf.update( buf[f][0], 0 );
buf[f][1] = m_lowShelf.update( buf[f][1], 1 );
}
if( para1Active )
{
m_para1.setParameters( sampleRate, *para1FreqPtr, *para1BwPtr, para1Gain );
buf[f][0] = m_para1.update( buf[f][0], 0 );
buf[f][1] = m_para1.update( buf[f][1], 1 );
}
if( para2Active )
{
m_para2.setParameters( sampleRate, *para2FreqPtr, *para2BwPtr, para2Gain );
buf[f][0] = m_para2.update( buf[f][0], 0 );
buf[f][1] = m_para2.update( buf[f][1], 1 );
}
if( para3Active )
{
m_para3.setParameters( sampleRate, *para3FreqPtr, *para3BwPtr, para3Gain );
buf[f][0] = m_para3.update( buf[f][0], 0 );
buf[f][1] = m_para3.update( buf[f][1], 1 );
}
if( para4Active )
{
m_para4.setParameters( sampleRate, *para4FreqPtr, *para4BwPtr, para4Gain );
buf[f][0] = m_para4.update( buf[f][0], 0 );
buf[f][1] = m_para4.update( buf[f][1], 1 );
}
if( highShelfActive )
{
m_highShelf.setParameters( sampleRate, *hightShelfFreqPtr, *highShelfResPtr, highShelfGain );
buf[f][0] = m_highShelf.update( buf[f][0], 0 );
buf[f][1] = m_highShelf.update( buf[f][1], 1 );
}
if( lpActive ){
m_lp12.setParameters( sampleRate, *lpFreqPtr, *lpResPtr, 1 );
buf[f][0] = m_lp12.update( buf[f][0], 0 );
buf[f][1] = m_lp12.update( buf[f][1], 1 );
if( lp24Active || lp48Active )
{
m_lp24.setParameters( sampleRate, *lpFreqPtr, *lpResPtr, 1 );
buf[f][0] = m_lp24.update( buf[f][0], 0 );
buf[f][1] = m_lp24.update( buf[f][1], 1 );
}
if( lp48Active )
{
m_lp480.setParameters( sampleRate, *lpFreqPtr, *lpResPtr, 1 );
buf[f][0] = m_lp480.update( buf[f][0], 0 );
buf[f][1] = m_lp480.update( buf[f][1], 1 );
m_lp481.setParameters( sampleRate, *lpFreqPtr, *lpResPtr, 1 );
buf[f][0] = m_lp481.update( buf[f][0], 0 );
buf[f][1] = m_lp481.update( buf[f][1], 1 );
}
}
//increment pointers if needed
hpResPtr += hpResInc;
lowShelfResPtr += lowShelfResInc;
para1BwPtr += para1BwInc;
para2BwPtr += para2BwInc;
para3BwPtr += para3BwInc;
para4BwPtr += para4BwInc;
highShelfResPtr += highShelfResInc;
lpResPtr += lpResInc;
hpFreqPtr += hpFreqInc;
lowShelfFreqPtr += lowShelfFreqInc;
para1FreqPtr += para1FreqInc;
para2FreqPtr += para2FreqInc;
para3FreqPtr += para3FreqInc;
para4FreqPtr += para4FreqInc;
hightShelfFreqPtr += highShelfFreqInc;
lpFreqPtr += lpFreqInc;
}
sampleFrame outPeak = { 0, 0 };
gain( buf, frames, outGain, &outPeak );
m_eqControls.m_outPeakL = m_eqControls.m_outPeakL < outPeak[0] ? outPeak[0] : m_eqControls.m_outPeakL;
m_eqControls.m_outPeakR = m_eqControls.m_outPeakR < outPeak[1] ? outPeak[1] : m_eqControls.m_outPeakR;
checkGate( outSum / frames );
if(m_eqControls.m_analyseOutModel.value( true ) && outSum > 0 )
{
m_eqControls.m_outFftBands.analyze( buf, frames );
setBandPeaks( &m_eqControls.m_outFftBands , ( int )( sampleRate ) );
}
else
{
m_eqControls.m_outFftBands.clear();
}
m_eqControls.m_inProgress = false;
return isRunning();
}
void InstrumentTrack::processAudioBuffer( sampleFrame* buf, const fpp_t frames, NotePlayHandle* n )
{
// we must not play the sound if this InstrumentTrack is muted...
if( isMuted() || ( n && n->isBbTrackMuted() ) || ! m_instrument )
{
return;
}
// Test for silent input data if instrument provides a single stream only (i.e. driven by InstrumentPlayHandle)
// We could do that in all other cases as well but the overhead for silence test is bigger than
// what we potentially save. While playing a note, a NotePlayHandle-driven instrument will produce sound in
// 99 of 100 cases so that test would be a waste of time.
if( m_instrument->flags().testFlag( Instrument::IsSingleStreamed ) &&
MixHelpers::isSilent( buf, frames ) )
{
// at least pass one silent buffer to allow
if( m_silentBuffersProcessed )
{
// skip further processing
return;
}
m_silentBuffersProcessed = true;
}
else
{
m_silentBuffersProcessed = false;
}
// if effects "went to sleep" because there was no input, wake them up
// now
m_audioPort.effects()->startRunning();
// 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 )
{
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() );
// get panning knob data
ValueBuffer * panBuf = m_panningModel.valueBuffer();
int panning = panBuf
? 0
: m_panningModel.value();
if( n )
{
panning += n->getPanning();
panning = tLimit<int>( panning, PanningLeft, PanningRight );
}
// 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 );
}
bool DualFilterEffect::processAudioBuffer( sampleFrame* buf, const fpp_t frames )
{
if( !isEnabled() || !isRunning () )
{
return( false );
}
double outSum = 0.0;
const float d = dryLevel();
const float w = wetLevel();
if( m_dfControls.m_filter1Model.isValueChanged() || m_filter1changed )
{
m_filter1->setFilterType( m_dfControls.m_filter1Model.value() );
m_filter1changed = true;
}
if( m_dfControls.m_filter2Model.isValueChanged() || m_filter2changed )
{
m_filter2->setFilterType( m_dfControls.m_filter2Model.value() );
m_filter2changed = true;
}
float cut1 = m_dfControls.m_cut1Model.value();
float res1 = m_dfControls.m_res1Model.value();
float gain1 = m_dfControls.m_gain1Model.value();
float cut2 = m_dfControls.m_cut2Model.value();
float res2 = m_dfControls.m_res2Model.value();
float gain2 = m_dfControls.m_gain2Model.value();
float mix = m_dfControls.m_mixModel.value();
ValueBuffer *cut1Buffer = m_dfControls.m_cut1Model.valueBuffer();
ValueBuffer *res1Buffer = m_dfControls.m_res1Model.valueBuffer();
ValueBuffer *gain1Buffer = m_dfControls.m_gain1Model.valueBuffer();
ValueBuffer *cut2Buffer = m_dfControls.m_cut2Model.valueBuffer();
ValueBuffer *res2Buffer = m_dfControls.m_res2Model.valueBuffer();
ValueBuffer *gain2Buffer = m_dfControls.m_gain2Model.valueBuffer();
ValueBuffer *mixBuffer = m_dfControls.m_mixModel.valueBuffer();
int cut1Inc = cut1Buffer ? 1 : 0;
int res1Inc = res1Buffer ? 1 : 0;
int gain1Inc = gain1Buffer ? 1 : 0;
int cut2Inc = cut2Buffer ? 1 : 0;
int res2Inc = res2Buffer ? 1 : 0;
int gain2Inc = gain2Buffer ? 1 : 0;
int mixInc = mixBuffer ? 1 : 0;
float *cut1Ptr = cut1Buffer ? &( cut1Buffer->values()[ 0 ] ) : &cut1;
float *res1Ptr = res1Buffer ? &( res1Buffer->values()[ 0 ] ) : &res1;
float *gain1Ptr = gain1Buffer ? &( gain1Buffer->values()[ 0 ] ) : &gain1;
float *cut2Ptr = cut2Buffer ? &( cut2Buffer->values()[ 0 ] ) : &cut2;
float *res2Ptr = res2Buffer ? &( res2Buffer->values()[ 0 ] ) : &res2;
float *gain2Ptr = gain2Buffer ? &( gain2Buffer->values()[ 0 ] ) : &gain2;
float *mixPtr = mixBuffer ? &( mixBuffer->values()[ 0 ] ) : &mix;
const bool enabled1 = m_dfControls.m_enabled1Model.value();
const bool enabled2 = m_dfControls.m_enabled2Model.value();
// buffer processing loop
for( fpp_t f = 0; f < frames; ++f )
{
// get mix amounts for wet signals of both filters
const float mix2 = ( ( *mixPtr + 1.0f ) * 0.5f );
const float mix1 = 1.0f - mix2;
const float gain1 = *gain1Ptr * 0.01f;
const float gain2 = *gain2Ptr * 0.01f;
sample_t s[2] = { 0.0f, 0.0f }; // mix
sample_t s1[2] = { buf[f][0], buf[f][1] }; // filter 1
sample_t s2[2] = { buf[f][0], buf[f][1] }; // filter 2
// update filter 1
if( enabled1 )
{
//update filter 1 params here
// recalculate only when necessary: either cut/res is changed, or the changed-flag is set (filter type or samplerate changed)
if( ( ( *cut1Ptr != m_currentCut1 ||
*res1Ptr != m_currentRes1 ) ) || m_filter1changed )
{
m_filter1->calcFilterCoeffs( *cut1Ptr, *res1Ptr );
m_filter1changed = false;
m_currentCut1 = *cut1Ptr;
m_currentRes1 = *res1Ptr;
}
s1[0] = m_filter1->update( s1[0], 0 );
s1[1] = m_filter1->update( s1[1], 1 );
// apply gain
s1[0] *= gain1;
s1[1] *= gain1;
// apply mix
s[0] += ( s1[0] * mix1 );
s[1] += ( s1[1] * mix1 );
}
// update filter 2
if( enabled2 )
{
//update filter 2 params here
if( ( ( *cut2Ptr != m_currentCut2 ||
*res2Ptr != m_currentRes2 ) ) || m_filter2changed )
{
m_filter2->calcFilterCoeffs( *cut2Ptr, *res2Ptr );
m_filter2changed = false;
m_currentCut2 = *cut2Ptr;
m_currentRes2 = *res2Ptr;
}
s2[0] = m_filter2->update( s2[0], 0 );
s2[1] = m_filter2->update( s2[1], 1 );
//apply gain
s2[0] *= gain2;
s2[1] *= gain2;
// apply mix
s[0] += ( s2[0] * mix2 );
s[1] += ( s2[1] * mix2 );
}
outSum += buf[f][0]*buf[f][0] + buf[f][1]*buf[f][1];
// do another mix with dry signal
buf[f][0] = d * buf[f][0] + w * s[0];
buf[f][1] = d * buf[f][1] + w * s[1];
//increment pointers
cut1Ptr += cut1Inc;
res1Ptr += res1Inc;
gain1Ptr += gain1Inc;
cut2Ptr += cut2Inc;
res2Ptr += res2Inc;
gain2Ptr += gain2Inc;
mixPtr += mixInc;
}
checkGate( outSum / frames );
return isRunning();
}
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() )
{
vb = m_controllerConnection->valueBuffer();
if( vb )
{
float * values = vb->values();
float * nvalues = m_valueBuffer.values();
switch( m_scaleType )
{
case Linear:
for( int i = 0; i < m_valueBuffer.length(); i++ )
{
nvalues[i] = minValue<float>() + ( range() * values[i] );
}
break;
case Logarithmic:
for( int i = 0; i < m_valueBuffer.length(); i++ )
{
nvalues[i] = logToLinearScale( values[i] );
}
break;
default:
qFatal("AutomatableModel::valueBuffer() "
"lacks implementation for a scale type");
break;
}
m_lastUpdatedPeriod = s_periodCounter;
m_hasSampleExactData = true;
return &m_valueBuffer;
}
}
AutomatableModel* lm = NULL;
if( m_hasLinkedModels )
{
lm = m_linkedModels.first();
}
if( lm && lm->controllerConnection() && lm->controllerConnection()->getController()->isSampleExact() )
{
vb = lm->valueBuffer();
float * values = vb->values();
float * nvalues = m_valueBuffer.values();
for( int i = 0; i < vb->length(); i++ )
{
nvalues[i] = fittedValue( values[i] );
}
m_lastUpdatedPeriod = s_periodCounter;
m_hasSampleExactData = true;
return &m_valueBuffer;
}
if( m_oldValue != val )
{
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, 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;
}
bool LadspaEffect::processAudioBuffer( sampleFrame * _buf,
const fpp_t _frames )
{
m_pluginMutex.lock();
if( !isOkay() || dontRun() || !isRunning() || !isEnabled() )
{
m_pluginMutex.unlock();
return( false );
}
int frames = _frames;
sampleFrame * o_buf = NULL;
sampleFrame sBuf [_frames];
if( m_maxSampleRate < Engine::mixer()->processingSampleRate() )
{
o_buf = _buf;
_buf = &sBuf[0];
sampleDown( o_buf, _buf, m_maxSampleRate );
frames = _frames * m_maxSampleRate /
Engine::mixer()->processingSampleRate();
}
// Copy the LMMS audio buffer to the LADSPA input buffer and initialize
// the control ports.
ch_cnt_t channel = 0;
for( ch_cnt_t proc = 0; proc < processorCount(); ++proc )
{
for( int port = 0; port < m_portCount; ++port )
{
port_desc_t * pp = m_ports.at( proc ).at( port );
switch( pp->rate )
{
case CHANNEL_IN:
for( fpp_t frame = 0;
frame < frames; ++frame )
{
pp->buffer[frame] =
_buf[frame][channel];
}
++channel;
break;
case AUDIO_RATE_INPUT:
{
ValueBuffer * vb = pp->control->valueBuffer();
if( vb )
{
memcpy( pp->buffer, vb->values(), frames * sizeof(float) );
}
else
{
pp->value = static_cast<LADSPA_Data>(
pp->control->value() / pp->scale );
// This only supports control rate ports, so the audio rates are
// treated as though they were control rate by setting the
// port buffer to all the same value.
for( fpp_t frame = 0;
frame < frames; ++frame )
{
pp->buffer[frame] =
pp->value;
}
}
break;
}
case CONTROL_RATE_INPUT:
if( pp->control == NULL )
{
break;
}
pp->value = static_cast<LADSPA_Data>(
pp->control->value() / pp->scale );
pp->buffer[0] =
pp->value;
break;
case CHANNEL_OUT:
case AUDIO_RATE_OUTPUT:
case CONTROL_RATE_OUTPUT:
break;
default:
break;
}
}
}
// Process the buffers.
for( ch_cnt_t proc = 0; proc < processorCount(); ++proc )
{
(m_descriptor->run)( m_handles[proc], frames );
}
// Copy the LADSPA output buffers to the LMMS buffer.
double out_sum = 0.0;
channel = 0;
const float d = dryLevel();
const float w = wetLevel();
for( ch_cnt_t proc = 0; proc < processorCount(); ++proc )
{
for( int port = 0; port < m_portCount; ++port )
{
port_desc_t * pp = m_ports.at( proc ).at( port );
switch( pp->rate )
{
case CHANNEL_IN:
case AUDIO_RATE_INPUT:
case CONTROL_RATE_INPUT:
break;
case CHANNEL_OUT:
for( fpp_t frame = 0;
frame < frames; ++frame )
{
_buf[frame][channel] = d * _buf[frame][channel] + w * pp->buffer[frame];
out_sum += _buf[frame][channel] * _buf[frame][channel];
}
++channel;
break;
case AUDIO_RATE_OUTPUT:
case CONTROL_RATE_OUTPUT:
break;
default:
break;
}
}
}
if( o_buf != NULL )
{
sampleBack( _buf, o_buf, m_maxSampleRate );
}
checkGate( out_sum / frames );
bool is_running = isRunning();
m_pluginMutex.unlock();
return( is_running );
}