Mixer: rewrote processing chain of rendered audio buffers
Until now, Mixer not only was responsible for rendering audio buffers but also managed writing them to audio backend (through a FIFO) and handled various quality related parameters. All this functionality has been moved into the new AudioOutputContext class. It glues together AudioBackend (formerly called AudioDevice), global quality settings and the Mixer. The AudioOutputContext class creates a FifoWriter which calls Mixer::renderNextBuffer() and writes the output into the BufferFifo of the AudioOutputContext it belongs to. The BufferFifo is read by the according AudioBackend which belongs to the AudioOutputContext as well. The AudioOutputContext also handles resampling in case the AudioBackend wants the buffers in a different samplerate. During this rewrite the Mixer class and the according source files have been renamed from "mixer" to "Mixer". This results in small changes all over LMMS' code base.
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@@ -2,6 +2,7 @@
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* ladspa_description.cpp - LADSPA plugin description
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*
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* Copyright (c) 2007 Javier Serrano Polo <jasp00/at/users.sourceforge.net>
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* Copyright (c) 2009 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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@@ -30,10 +31,11 @@
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#include <QtGui/QScrollArea>
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#include <QtGui/QVBoxLayout>
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#include "AudioDevice.h"
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#include "AudioBackend.h"
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#include "AudioOutputContext.h"
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#include "engine.h"
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#include "ladspa_2_lmms.h"
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#include "mixer.h"
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#include "Mixer.h"
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@@ -73,8 +75,8 @@ ladspaDescription::ladspaDescription( QWidget * _parent,
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it != plugins.end(); it++ )
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{
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if( _type != VALID ||
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manager->getDescription( ( *it ).second )->inputChannels
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<= engine::getMixer()->audioDev()->channels() )
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manager->getDescription( ( *it ).second )->inputChannels <=
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engine::mixer()->audioOutputContext()->audioBackend()->channels() )
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{
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pluginNames.push_back( ( *it ).first );
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m_pluginKeys.push_back( ( *it ).second );
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@@ -2,7 +2,8 @@
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* ladspa_port_dialog.cpp - dialog to test a LADSPA plugin
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*
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* Copyright (c) 2006-2008 Danny McRae <khjklujn/at/users.sourceforge.net>
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*
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* Copyright (c) 2009 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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* This program is free software; you can redistribute it and/or
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@@ -22,7 +23,6 @@
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*
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*/
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#include "ladspa_port_dialog.h"
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#include <QtGui/QLayout>
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@@ -31,7 +31,7 @@
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#include "embed.h"
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#include "engine.h"
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#include "ladspa_2_lmms.h"
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#include "mixer.h"
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#include "Mixer.h"
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ladspaPortDialog::ladspaPortDialog( const ladspa_key_t & _key )
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@@ -95,11 +95,11 @@ ladspaPortDialog::ladspaPortDialog( const ladspa_key_t & _key )
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{
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if( min != NOHINT )
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{
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min *= engine::getMixer()->processingSampleRate();
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min *= engine::mixer()->processingSampleRate();
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}
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if( max != NOHINT )
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{
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max *= engine::getMixer()->processingSampleRate();
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max *= engine::mixer()->processingSampleRate();
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}
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}
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@@ -25,14 +25,15 @@
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#include <QtGui/QMessageBox>
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#include "AudioBackend.h"
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#include "AudioOutputContext.h"
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#include "LadspaEffect.h"
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#include "mmp.h"
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#include "AudioDevice.h"
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#include "config_mgr.h"
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#include "ladspa_2_lmms.h"
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#include "LadspaControl.h"
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#include "LadspaSubPluginFeatures.h"
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#include "mixer.h"
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#include "Mixer.h"
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#include "EffectChain.h"
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#include "Cpu.h"
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#include "automation_pattern.h"
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@@ -87,7 +88,7 @@ LadspaEffect::LadspaEffect( Model * _parent,
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pluginInstantiation();
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connect( engine::getMixer(), SIGNAL( sampleRateChanged() ),
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connect( engine::mixer(), SIGNAL( sampleRateChanged() ),
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this, SLOT( changeSampleRate() ) );
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}
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@@ -144,13 +145,13 @@ bool LadspaEffect::processAudioBuffer( sampleFrame * _buf,
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int frames = _frames;
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sampleFrame * o_buf = NULL;
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if( m_maxSampleRate < engine::getMixer()->processingSampleRate() )
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if( m_maxSampleRate < engine::mixer()->processingSampleRate() )
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{
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o_buf = _buf;
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_buf = CPU::allocFrames( _frames );
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sampleDown( o_buf, _buf, m_maxSampleRate );
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frames = _frames * m_maxSampleRate /
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engine::getMixer()->processingSampleRate();
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engine::mixer()->processingSampleRate();
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}
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// Copy the LMMS audio buffer to the LADSPA input buffer and initialize
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@@ -298,7 +299,8 @@ void LadspaEffect::pluginInstantiation()
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ladspa2LMMS * manager = engine::getLADSPAManager();
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// Calculate how many processing units are needed.
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const ch_cnt_t lmms_chnls = engine::getMixer()->audioDev()->channels();
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const ch_cnt_t lmms_chnls = engine::mixer()->audioOutputContext()->
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audioBackend()->channels();
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int effect_channels = manager->getDescription( m_key )->inputChannels;
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setProcessorCount( lmms_chnls / effect_channels );
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@@ -325,7 +327,7 @@ void LadspaEffect::pluginInstantiation()
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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 =
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new LADSPA_Data[engine::getMixer()->framesPerPeriod()];
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new LADSPA_Data[engine::mixer()->framesPerPeriod()];
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if( p->name.toUpper().contains( "IN" ) &&
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manager->isPortInput( m_key, port ) )
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@@ -566,7 +568,7 @@ sample_rate_t LadspaEffect::maxSamplerate( const QString & _name )
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{
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return __buggy_plugins[_name];
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}
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return engine::getMixer()->processingSampleRate();
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return engine::mixer()->processingSampleRate();
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}
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@@ -28,12 +28,13 @@
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#include <QtGui/QHBoxLayout>
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#include <QtGui/QLabel>
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#include "AudioOutputContext.h"
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#include "AudioBackend.h"
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#include "LadspaSubPluginFeatures.h"
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#include "AudioDevice.h"
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#include "engine.h"
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#include "ladspa_2_lmms.h"
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#include "LadspaBase.h"
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#include "mixer.h"
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#include "Mixer.h"
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LadspaSubPluginFeatures::LadspaSubPluginFeatures( Plugin::PluginTypes _type ) :
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@@ -142,7 +143,7 @@ void LadspaSubPluginFeatures::listSubPluginKeys(
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it != plugins.end(); ++it )
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{
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if( lm->getDescription( ( *it ).second )->inputChannels <=
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engine::getMixer()->audioDev()->channels() )
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engine::mixer()->audioOutputContext()->audioBackend()->channels() )
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{
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_kl.push_back( ladspaKeyToSubPluginKey( _desc, ( *it ).first, ( *it ).second ) );
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}
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@@ -37,7 +37,6 @@
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#include "InstrumentView.h"
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#include "led_checkbox.h"
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#include "knob.h"
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#include "mixer.h"
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class lb302SynthView;
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class notePlayHandle;
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@@ -30,6 +30,8 @@
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#include <QtGui/QFileDialog>
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#include <QtXml/QDomDocument>
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#include "AudioBackend.h"
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#include "AudioOutputContext.h"
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#include "ResourceFileMapper.h"
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#include "sf2_player.h"
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#include "engine.h"
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@@ -122,14 +124,14 @@ sf2Instrument::sf2Instrument( InstrumentTrack * _instrument_track ) :
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m_settings = new_fluid_settings();
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fluid_settings_setint( m_settings, (char *) "audio.period-size",
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engine::getMixer()->framesPerPeriod() );
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engine::mixer()->framesPerPeriod() );
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// This is just our starting instance of synth. It is recreated
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// everytime we load a new soundfont.
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m_synth = new_fluid_synth( m_settings );
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InstrumentPlayHandle * iph = new InstrumentPlayHandle( this );
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engine::getMixer()->addPlayHandle( iph );
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engine::mixer()->addPlayHandle( iph );
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//loadFile( configManager::inst()->defaultSoundfont() );
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@@ -147,7 +149,7 @@ sf2Instrument::sf2Instrument( InstrumentTrack * _instrument_track ) :
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connect( &m_patchNum, SIGNAL( dataChanged() ),
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this, SLOT( updatePatch() ) );
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connect( engine::getMixer(), SIGNAL( sampleRateChanged() ),
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connect( engine::mixer(), SIGNAL( sampleRateChanged() ),
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this, SLOT( updateSampleRate() ) );
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// Gain
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@@ -192,7 +194,7 @@ sf2Instrument::sf2Instrument( InstrumentTrack * _instrument_track ) :
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sf2Instrument::~sf2Instrument()
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{
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engine::getMixer()->removePlayHandles( instrumentTrack() );
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engine::mixer()->removePlayHandles( instrumentTrack() );
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freeFont();
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delete_fluid_synth( m_synth );
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delete_fluid_settings( m_settings );
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@@ -498,7 +500,7 @@ void sf2Instrument::updateSampleRate()
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// Set & get, returns the true sample rate
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fluid_settings_setnum( m_settings, (char *) "synth.sample-rate",
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engine::getMixer()->processingSampleRate() );
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engine::mixer()->processingSampleRate() );
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fluid_settings_getnum( m_settings, (char *) "synth.sample-rate",
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&tempRate );
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m_internalSampleRate = static_cast<int>( tempRate );
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@@ -529,8 +531,9 @@ void sf2Instrument::updateSampleRate()
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}
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m_synthMutex.lock();
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if( engine::getMixer()->currentQualitySettings().interpolation >=
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mixer::qualitySettings::Interpolation_SincFastest )
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if( engine::mixer()->audioOutputContext()->qualitySettings().
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interpolation() >=
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AudioOutputContext::QualitySettings::Interpolation_SincFastest )
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{
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fluid_synth_set_interp_method( m_synth, -1,
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FLUID_INTERP_7THORDER );
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@@ -541,7 +544,7 @@ void sf2Instrument::updateSampleRate()
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FLUID_INTERP_DEFAULT );
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}
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m_synthMutex.unlock();
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if( m_internalSampleRate < engine::getMixer()->processingSampleRate() )
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if( m_internalSampleRate < engine::mixer()->processingSampleRate() )
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{
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m_synthMutex.lock();
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if( m_srcState != NULL )
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@@ -549,9 +552,9 @@ void sf2Instrument::updateSampleRate()
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src_delete( m_srcState );
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}
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int error;
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m_srcState = src_new( engine::getMixer()->
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currentQualitySettings().libsrcInterpolation(),
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DEFAULT_CHANNELS, &error );
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m_srcState = src_new( engine::mixer()->audioOutputContext()->
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qualitySettings().libsrcInterpolation(),
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DEFAULT_CHANNELS, &error );
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if( m_srcState == NULL || error )
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{
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printf( "error while creating SRC-data-"
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@@ -26,7 +26,7 @@
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#include "vibrating_string.h"
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#include "templates.h"
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#include "interpolation.h"
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#include "mixer.h"
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#include "Mixer.h"
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#include "engine.h"
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@@ -42,7 +42,7 @@ vibratingString::vibratingString( float _pitch,
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float _detune,
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bool _state ) :
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m_oversample( 2 * _oversample / (int)( _sample_rate /
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engine::getMixer()->baseSampleRate() ) ),
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engine::mixer()->baseSampleRate() ) ),
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m_randomize( _randomize ),
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m_stringLoss( 1.0f - _string_loss ),
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m_state( 0.1f )
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@@ -29,7 +29,6 @@
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#include <QtCore/QMutex>
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#include <QtGui/QWidget>
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#include "mixer.h"
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#include "JournallingObject.h"
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#include "communication.h"
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