split effect into effect and ladspaEffect
git-svn-id: https://lmms.svn.sf.net/svnroot/lmms/trunk/lmms@325 0778d3d1-df1d-0410-868b-ea421aaaa00d
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Danny McRae committed
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@@ -39,18 +39,10 @@
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#include "effect.h"
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#include "mixer.h"
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#include "config_mgr.h"
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#include "buffer_allocator.h"
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#include "audio_device.h"
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#include "ladspa_control.h"
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effect::effect( const ladspa_key_t & _key, engine * _engine ) :
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effect::effect( engine * _engine ) :
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engineObject( _engine ),
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m_key( _key ),
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m_ladspa( _engine->getLADSPAManager() ),
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m_output( NULL ),
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m_okay( TRUE ),
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m_noRun( FALSE ),
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m_running( FALSE ),
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@@ -60,183 +52,6 @@ effect::effect( const ladspa_key_t & _key, engine * _engine ) :
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m_wetDry( 1.0f ),
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m_gate( 0.0f )
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{
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if( m_ladspa->getDescription( _key ) == NULL )
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{
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QMessageBox::warning( 0, "Effect", "Uknown LADSPA plugin requested: " + _key.first, QMessageBox::Ok, QMessageBox::NoButton );
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m_okay = FALSE;
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return;
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}
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m_name = m_ladspa->getShortName( _key );
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// Calculate how many processing units are needed.
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ch_cnt_t lmms_chnls = eng()->getMixer()->audioDev()->channels();
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m_effectChannels = m_ladspa->getDescription( _key )->inputChannels;
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m_processors = lmms_chnls / m_effectChannels;
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// Categorize the ports, and create the buffers.
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m_bufferSize = eng()->getMixer()->framesPerAudioBuffer();
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m_portCount = m_ladspa->getPortCount( _key );
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for( ch_cnt_t proc = 0; proc < m_processors; proc++ )
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{
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multi_proc_t ports;
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for( Uint16 port = 0; port < m_portCount; port++ )
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{
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port_desc_t * p = new portDescription;
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p->name = m_ladspa->getPortName( _key, port );
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p->proc = proc;
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p->port_id = port;
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// Determine the port's category.
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if( m_ladspa->isPortAudio( _key, port ) )
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{
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// Nasty manual memory management--was having difficulty
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// with some prepackaged plugins that were segfaulting
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// during cleanup. It was easier to troubleshoot with the
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// memory management all taking place in one file.
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p->buffer = new LADSPA_Data[m_bufferSize];
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if( p->name.upper().contains( "IN" ) && m_ladspa->isPortInput( _key, port ) )
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{
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p->rate = CHANNEL_IN;
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}
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else if( p->name.upper().contains( "OUT" ) && m_ladspa->isPortOutput( _key, port ) )
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{
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p->rate = CHANNEL_OUT;
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}
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else if( m_ladspa->isPortInput( _key, port ) )
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{
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p->rate = AUDIO_RATE_INPUT;
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}
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else
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{
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p->rate = AUDIO_RATE_OUTPUT;
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}
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}
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else
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{
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p->buffer = new LADSPA_Data;
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if( m_ladspa->isPortInput( _key, port ) )
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{
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p->rate = CONTROL_RATE_INPUT;
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}
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else
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{
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p->rate = CONTROL_RATE_OUTPUT;
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}
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}
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if( m_ladspa->isPortToggled( _key, port ) )
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{
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p->data_type = TOGGLED;
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}
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else if( m_ladspa->isInteger( _key, port ) )
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{
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p->data_type = INTEGER;
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}
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else
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{
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p->data_type = FLOAT;
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}
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// Get the range and default values.
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p->is_scaled = m_ladspa->areHintsSampleRateDependent( _key, port );
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p->max = m_ladspa->getUpperBound( _key, port );
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if( p->max == NOHINT )
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{
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p->max = 999999.0f;
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}
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else if( p->is_scaled )
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{
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p->max *= eng()->getMixer()->sampleRate();
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}
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p->min = m_ladspa->getLowerBound( _key, port );
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if( p->min == NOHINT )
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{
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p->min = -999999.0f;
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}
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else if( p->is_scaled )
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{
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p->min *= eng()->getMixer()->sampleRate();
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}
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p->def = m_ladspa->getDefaultSetting( _key, port );
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if( p->def == NOHINT )
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{
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if( p->data_type != TOGGLED )
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{
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p->def = ( p->min + p->max ) / 2.0f;
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}
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else
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{
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p->def = 1.0f;
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}
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}
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p->value = p->def;
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ports.append( p );
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// For convenience, keep a separate list of the ports that are used
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// to control the processors.
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if( p->rate == AUDIO_RATE_INPUT || p->rate == CONTROL_RATE_INPUT )
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{
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p->control_id = m_controls.count();
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m_controls.append( p );
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}
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}
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m_ports.append( ports );
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}
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// Instantiate the processing units.
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m_descriptor = m_ladspa->getDescriptor( _key );
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if( m_descriptor == NULL )
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{
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QMessageBox::warning( 0, "Effect", "Can't get LADSPA descriptor function: " + _key.first, QMessageBox::Ok, QMessageBox::NoButton );
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m_okay = FALSE;
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return;
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}
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if( m_descriptor->run == NULL )
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{
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QMessageBox::warning( 0, "Effect", "Plugin has no processor: " + _key.first, QMessageBox::Ok, QMessageBox::NoButton );
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m_noRun = TRUE;
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}
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for( ch_cnt_t proc = 0; proc < m_processors; proc++ )
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{
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LADSPA_Handle effect = m_ladspa->instantiate( _key, eng()->getMixer()->sampleRate() );
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if( effect == NULL )
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{
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QMessageBox::warning( 0, "Effect", "Can't get LADSPA instance: " + _key.first, QMessageBox::Ok, QMessageBox::NoButton );
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m_okay = FALSE;
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return;
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}
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m_handles.append( effect );
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}
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// Connect the ports.
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for( ch_cnt_t proc = 0; proc < m_processors; proc++ )
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{
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for( Uint16 port = 0; port < m_portCount; port++ )
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{
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if( !m_ladspa->connectPort( _key, m_handles[proc], port, m_ports[proc][port]->buffer ) )
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{
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QMessageBox::warning( 0, "Effect", "Failed to connect port: " + _key.first, QMessageBox::Ok, QMessageBox::NoButton );
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m_noRun = TRUE;
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}
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}
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}
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// Activate the processing units.
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for( ch_cnt_t proc = 0; proc < m_processors; proc++ )
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{
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m_ladspa->activate( _key, m_handles[proc] );
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}
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}
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@@ -244,28 +59,6 @@ effect::effect( const ladspa_key_t & _key, engine * _engine ) :
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effect::~effect()
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{
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if( !m_okay )
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{
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return;
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}
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m_processLock.lock();
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for( ch_cnt_t proc = 0; proc < m_processors; proc++ )
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{
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m_ladspa->deactivate( m_key, m_handles[proc] );
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m_ladspa->cleanup( m_key, m_handles[proc] );
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for( Uint16 port = 0; port < m_portCount; port++ )
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{
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free( m_ports[proc][port]->buffer );
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free( m_ports[proc][port] );
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}
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m_ports[proc].clear();
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}
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m_ports.clear();
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m_handles.clear();
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m_processLock.unlock();
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}
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@@ -273,120 +66,7 @@ effect::~effect()
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bool FASTCALL effect::processAudioBuffer( surroundSampleFrame * _buf, const fpab_t _frames )
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{
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if( !m_okay || m_noRun || !m_running || m_bypass || !m_processLock.tryLock() )
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{
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return( FALSE );
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}
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// Copy the LMMS audio buffer to the LADSPA input buffer and initialize
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// the control ports. Need to change this to handle non-in-place-broken
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// plugins--would speed things up to use the same buffer for both
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// LMMS and LADSPA.
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ch_cnt_t channel = 0;
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for( ch_cnt_t proc = 0; proc < m_processors; proc++)
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{
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for( Uint16 port = 0; port < m_portCount; port++ )
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{
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switch( m_ports[proc][port]->rate )
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{
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case CHANNEL_IN:
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for( fpab_t frame = 0; frame < _frames; frame++ )
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{
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m_ports[proc][port]->buffer[frame] = _buf[frame][channel];
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}
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channel++;
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break;
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case AUDIO_RATE_INPUT:
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m_ports[proc][port]->value = static_cast<LADSPA_Data>( m_ports[proc][port]->control->getValue() );
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// This only supports control rate ports, so the audio rates are
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// treated as though they were control rate by setting the
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// port buffer to all the same value.
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for( fpab_t frame = 0; frame < _frames; frame++ )
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{
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m_ports[proc][port]->buffer[frame] = m_ports[proc][port]->value;
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}
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break;
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case CONTROL_RATE_INPUT:
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m_ports[proc][port]->value = static_cast<LADSPA_Data>( m_ports[proc][port]->control->getValue() );
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m_ports[proc][port]->buffer[0] = m_ports[proc][port]->value;
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break;
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case CHANNEL_OUT:
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case AUDIO_RATE_OUTPUT:
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case CONTROL_RATE_OUTPUT:
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break;
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default:
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break;
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}
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}
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}
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// Process the buffers.
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for( ch_cnt_t proc = 0; proc < m_processors; proc++ )
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{
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(m_descriptor->run)(m_handles[proc], _frames);
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}
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// Copy the LADSPA output buffers to the LMMS buffer.
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double out_sum = 0.0;
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channel = 0;
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for( ch_cnt_t proc = 0; proc < m_processors; proc++)
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{
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for( Uint16 port = 0; port < m_portCount; port++ )
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{
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switch( m_ports[proc][port]->rate )
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{
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case CHANNEL_IN:
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case AUDIO_RATE_INPUT:
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case CONTROL_RATE_INPUT:
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break;
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case CHANNEL_OUT:
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for( fpab_t frame = 0; frame < _frames; frame++ )
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{
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_buf[frame][channel] = ( 1.0f - m_wetDry ) * _buf[frame][channel] + m_wetDry * m_ports[proc][port]->buffer[frame];
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out_sum += _buf[frame][channel] * _buf[frame][channel];
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}
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channel++;
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break;
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case AUDIO_RATE_OUTPUT:
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case CONTROL_RATE_OUTPUT:
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break;
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default:
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break;
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}
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}
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}
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// Check whether we need to continue processing input. Restart the
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// counter if the threshold has been exceeded.
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if( out_sum <= ( m_gate ) )
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{
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m_bufferCount++;
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if( m_bufferCount > m_silenceTimeout )
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{
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m_running = FALSE;
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m_bufferCount = 0;
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}
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}
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else
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{
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m_bufferCount = 0;
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}
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m_processLock.unlock();
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return( m_running );
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}
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void FASTCALL effect::setControl( Uint16 _control, LADSPA_Data _value )
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{
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if( !m_okay )
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{
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return;
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}
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m_processLock.lock();
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m_controls[_control]->value = _value;
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m_processLock.unlock();
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return( FALSE );
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}
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@@ -395,12 +75,13 @@ void FASTCALL effect::setControl( Uint16 _control, LADSPA_Data _value )
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void FASTCALL effect::setGate( float _level )
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{
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m_processLock.lock();
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m_gate = _level * _level * m_processors * m_bufferSize;
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m_gate = _level * _level * m_processors * eng()->getMixer()->framesPerAudioBuffer();
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m_processLock.unlock();
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}
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#endif
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#endif
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