simplified filtering, VST fixes
git-svn-id: https://lmms.svn.sf.net/svnroot/lmms/trunk/lmms@478 0778d3d1-df1d-0410-868b-ea421aaaa00d
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@@ -700,6 +700,8 @@ bool flpImport::tryImport( trackContainer * _tc )
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case FLP_FilterParams:
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{
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// TODO: Dirty hack!
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// SIMPLE_FLT_CNT equals to old DOUBLE_LOWPASS
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const basicFilters<>::filterTypes
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mappedFilter[] =
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{
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@@ -708,9 +710,9 @@ bool flpImport::tryImport( trackContainer * _tc )
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basicFilters<>::BANDPASS_CSG,
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basicFilters<>::HIPASS,
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basicFilters<>::NOTCH,
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basicFilters<>::DOUBLE_LOWPASS,
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basicFilters<>::SIMPLE_FLT_CNT,
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basicFilters<>::LOWPASS,
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basicFilters<>::DOUBLE_LOWPASS
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basicFilters<>::SIMPLE_FLT_CNT
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} ;
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Uint32 * p = (Uint32 *) text;
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envelopeTabWidget * etw = it->m_envWidget;
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@@ -262,8 +262,10 @@ void vestigeInstrument::setParameter( const QString & _param,
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m_plugin->showEditor();
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connect( engine::getSongEditor(),
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SIGNAL( tempoChanged( bpm_t ) ),
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m_plugin, SLOT( setTempo( bpm_t ) ) );
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m_plugin, SLOT( setTempo( bpm_t ) ) );
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m_plugin->setTempo( engine::getSongEditor()->getTempo() );
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connect( engine::getMixer(), SIGNAL( sampleRateChanged() ),
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m_plugin, SLOT( updateSampleRate() ) );
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if( set_ch_name == TRUE )
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{
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getInstrumentTrack()->setName( m_plugin->name() );
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@@ -453,6 +453,17 @@ void remoteVSTPlugin::setTempo( bpm_t _bpm )
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void remoteVSTPlugin::updateSampleRate( void )
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{
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lock();
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writeValueS<Sint16>( VST_SAMPLE_RATE );
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writeValueS<sample_rate_t>( engine::getMixer()->sampleRate() );
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unlock();
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}
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const QMap<QString, QString> & remoteVSTPlugin::parameterDump( void )
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{
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writeValueS<Sint16>( VST_GET_PARAMETER_DUMP );
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@@ -127,6 +127,7 @@ public:
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public slots:
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void setTempo( bpm_t _bpm );
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void updateSampleRate( void );
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private:
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@@ -1,7 +1,7 @@
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/*
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* lvsl_server.cpp - LMMS VST Support Layer Server
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*
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* Copyright (c) 2005-2006 Tobias Doerffel <tobydox/at/users.sourceforge.net>
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* Copyright (c) 2005-2007 Tobias Doerffel <tobydox/at/users.sourceforge.net>
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*
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* This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
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*
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@@ -466,6 +466,7 @@ void VSTPlugin::process( void )
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for( ch_cnt_t i = 0; i < outputCount(); ++i )
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{
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m_outputs[i] = &m_shm[( i + inputCount() ) * m_blockSize];
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memset( m_outputs[i], 0, m_blockSize * sizeof( float ) );
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}
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#ifdef OLD_VST_SDK
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