Made 0.4.5 release

Bumped version number to 0.4.5 in CMakeLists.txt and README.

CMakeLists.txt

@@ -15,7 +15,7 @@ INCLUDE(FindPkgConfig)
 
 SET(VERSION_MAJOR "0")
 SET(VERSION_MINOR "4")
-SET(VERSION_PATCH "4")
+SET(VERSION_PATCH "5")
 #SET(VERSION_SUFFIX "")
 SET(VERSION "${VERSION_MAJOR}.${VERSION_MINOR}.${VERSION_PATCH}")
 IF(VERSION_SUFFIX)
@@ -33,6 +33,7 @@ ELSE(LMMS_HOST_X86_64)
 ENDIF(LMMS_HOST_X86_64)
 
 OPTION(WANT_ALSA	"Include ALSA (Advanced Linux Sound Architecture) support" ON)
+OPTION(WANT_CALF	"Include CALF LADSPA plugins" ON)
 OPTION(WANT_CAPS	"Include C* Audio Plugin Suite (LADSPA plugins)" ON)
 OPTION(WANT_CMT		"Include Computer Music Toolkit LADSPA plugins" ON)
 OPTION(WANT_FFTW3F	"Include SpectrumAnalyzer and ZynAddSubFX plugin" ON)
@@ -118,6 +119,13 @@ IF(NOT SNDFILE_FOUND)
 	MESSAGE(FATAL_ERROR "LMMS requires libsndfile1 and libsndfile1-dev >= 1.0.11 - please install, remove CMakeCache.txt and try again!")
 ENDIF(NOT SNDFILE_FOUND)
 
+IF(WANT_CALF)
+SET(LMMS_HAVE_CALF TRUE)
+SET(STATUS_CALF "OK")
+ELSE(WANT_CALF)
+SET(STATUS_CALF "not built as requested")
+ENDIF(WANT_CALF)
+
 IF(WANT_CAPS)
 SET(LMMS_HAVE_CAPS TRUE)
 SET(STATUS_CAPS "OK")
@@ -446,7 +454,9 @@ IF(LMMS_BUILD_WIN32)
 	ENDIF(LMMS_BUILD_WIN64)
 
 ELSE(LMMS_BUILD_WIN32)
-	SET_TARGET_PROPERTIES(lmms PROPERTIES LINK_FLAGS "${LINK_FLAGS} -Wl,-E")
+	IF(NOT LMMS_BUILD_APPLE)
+		SET_TARGET_PROPERTIES(lmms PROPERTIES LINK_FLAGS "${LINK_FLAGS} -Wl,-E")
+	ENDIF(NOT LMMS_BUILD_APPLE)
 
 	INSTALL(TARGETS lmms RUNTIME DESTINATION bin)
 	INSTALL(FILES ${CMAKE_BINARY_DIR}/lmms.1.gz DESTINATION ${CMAKE_INSTALL_PREFIX}/share/man/man1/ PERMISSIONS OWNER_READ GROUP_READ WORLD_READ)
@@ -602,6 +612,7 @@ MESSAGE(
 "* VST-instrument hoster       : ${STATUS_VST}\n"
 "* VST-effect hoster           : ${STATUS_VST}\n"
 "* SpectrumAnalyzer            : ${STATUS_FFTW3F}\n"
+"* CALF LADSPA plugins         : ${STATUS_CALF}\n"
 "* CAPS LADSPA plugins         : ${STATUS_CAPS}\n"
 "* CMT LADSPA plugins          : ${STATUS_CMT}\n"
 "* TAP LADSPA plugins          : ${STATUS_TAP}\n"

INSTALL

@@ -5,11 +5,10 @@ then run
 
 mkdir build
 cd build
-cmake ../ -DCMAKE_INSTALL_PREFIX=/usr
+cmake ../
 make
 sudo make install
 
-
 This way an out-of-tree build is performed. You can also run "cmake ." directly
 in the root of source-tree although this is not recommended. When performing an
 out-of-tree build after there's already an in-tree build, make sure to run
@@ -23,3 +22,9 @@ that are going to be built into LMMS or built as plugins. Install the
 according libraries and development files if a certain feature is not enabled.
 Then remove CMakeCache.txt and run cmake again.
 
+If you want to supply an install prefix to cmake, add the flag:
+
+-DCMAKE_INSTALL_PREFIX=<prefix>
+
+Where <prefix> can be /usr, /usr/local, /opt, etc. The default is /usr/local.
+

README

@@ -1,4 +1,4 @@
-Linux MultiMedia Studio 0.4.4
+Linux MultiMedia Studio 0.4.5
 ==============================
 
 Copyright (c) 2004-2009 by LMMS-developers

TODO

@@ -1,52 +1,32 @@
-- do not process effects when playing frozen patterns
-- select number of channels in export-project-dialog
-- try to make vestige-plugin-dlls relative
-- do song-editor-tempo-connection to vst-plugin inside remoteVSTPlugin
-- add/remove-steps button in bb-editor for adding/removing according number of steps to/from all patterns of visible beat/baseline
-- replace rest of wizard by simple directory-selection-dialog for working-dir when running the first time
-- correctly load steps/dots from FLP-files
-- convert FL-Plucked!-parameters to Vibed-parameters
-- in flp-import-filter: merge play-list-items if possible
-- integrated sample-browser in context-menu of sample-track/-tco
-- make note able of journalling
-- before calling undoStep/redoStep from journallingObject, save journalling-state-context and disabled journalling, restore afterwards
-- intelligent journal-entry-merging
-- undo/redo-support in note/track etc.
-- save tco-settings in trackContentWidget::saveSettings() etc. instead of track::...
-- restore stacking-order of windows when loading project
-- bristol-bindings?
-- resample sample-track-tcos when using hq-mode
-- add support for panes-interface (like blender) (instead of MDI etc.)
+Version 0.4.x
+=============
+
+- save tco-settings in trackContentWidget::saveSettings() etc. instead of 
+  track::...
+- resample sample-track-tcos when exporting at different samplerate
 - message to user when importing unsupported MIDI-file (track-count = 0)
-- AMS/OMS-bindings
-- remove binary-embed-system (Qt4-resource-system?)
-- recording-functionality
-- do not hang when saving while loading VST-plugin (because then we call dispatcher while the load-process is still going on)
-- tempo-recogn. and sync of beat-samples
-- make color-scheme switchable: LMMS / user
+- piano roll: mouse cursor isn't updated correctly in selection mode
+  (from resizing note edit area)
+- when you add vestige, have it automatically pop the find VST plugin dialog
+- try to make vestige-plugin-dlls relative
+- select all MIDI devices by default when you bring up the "connect to controller"
+  window and wait for first event - then uncheck all other MIDI devices that no
+  events were detected from
+- load asdlol.mmpz. if you render it without playing it, or if you play it
+  the first time, you hear unwanted artifacts. (solution: apply automation
+  before playing)
 - autosave every 30s (configurable!) and offer recovery at startup after crash
-- make piano-roll use rubberband instead of implementing a simple one on it's own
-- level-meters in output-graph and instrument-track
-- MIDI-program/MIDI-mapping/process program-/channel-change-events from MIDI-files
-- DSSI-support
-- use drawLineF() for drawing notes in pattern::paintEvent() in qt4-version
-- only redraw region given by paint-event in pattern, bbTCO, sampleTCO etc.
-- pre-listen when opening sample with QFileDialog
-- panning-editing in piano-roll
 - speed up painting of sampleTCO
-- panning env+lfo
-- rewrite export-project-dialog using layout-mechanism
-- make piano-roll use the global clipboard??
-- add more localizations:
-	- Swedish
-	- Norwegian
-	- Greece
-	- ...
+- do not process effects when playing frozen patterns
+- copy-pasted automation patterns have to be manually linked back to
+  their knob for some reason
+- improve TrackLabelButton: split 80%-20% (80%=name, 20%=button showing a popup
+  menu with track operations, make the midi input a top-level menu item)
+- when you click and drag a mixer bar, it doesn't click and drag, it sets
+  absolutely. this is annoying
 
 
-- effect-board -> live-fx from input
-- chord-editor?
-- WAVE/OGG/MP3-Import -> FFT-analysis -> write notes
-- classical note-edit-window -> also ability of printing and maybe later scanning & recognition of notes
-- add FLAC as export-format?
+See TODO file in master branch and/or the TODO list in the Wiki for details
+regarding the development series.
+
 

build_mingw32

@@ -1,7 +1,4 @@
-MINGW=/opt/mingw
+MINGW=/opt/mingw32
 export PATH=$PATH:$MINGW/bin
-#./configure --prefix=$MINGW --host=i586-mingw32 CC=$MINGW/bin/i586-mingw32-gcc CXX=$MINGW/bin/i586-mingw32-g++ DLLTOOL=$MINGW/bin/i586-mingw32-dlltool OBJDUMP=$MINGW/bin/i586-mingw32-objdump RANLIB=$MINGW/bin/i586-mingw32-ranlib AR=$MINGW/bin/i586-mingw32-ar AS=$MINGW/bin/i586-mingw32-as STRIP=$MINGW/bin/i586-mingw32-strip --with-qtdir=$MINGW --with-win32 CFLAGS="-mtune=generic -mmmx -fno-strict-aliasing" CXXFLAGS="-mtune=generic -mmmx -fno-strict-aliasing"
-#./configure --prefix=$MINGW --host=i586-mingw32 --build=i586-mingw32 CC=$MINGW/bin/i586-mingw32-gcc CXX=$MINGW/bin/i586-mingw32-g++ --with-qtdir=$MINGW --with-win32
-
-cmake . -DCMAKE_TOOLCHAIN_FILE=cmake/modules/Win32Toolchain.cmake
+cmake .. -DCMAKE_TOOLCHAIN_FILE=cmake/modules/Win32Toolchain.cmake
 

cmake/modules/Win32Toolchain.cmake

@@ -1,26 +1,28 @@
+SET(MINGW_PREFIX /opt/mingw32)
+
 # this one is important
 SET(CMAKE_SYSTEM_NAME Windows)
 #this one not so much
 SET(CMAKE_SYSTEM_VERSION 1)
 
 SET(CMAKE_SYSTEM_PROCESSOR i686)
-SET(CMAKE_INSTALL_PREFIX /opt/mingw32)
+SET(CMAKE_INSTALL_PREFIX ${MINGW_PREFIX})
 
-SET(CC_PREFIX /opt/mingw32)
 
 # specify the cross compiler
-SET(CMAKE_C_COMPILER   ${CC_PREFIX}/bin/i586-pc-mingw32-gcc)
-SET(CMAKE_CXX_COMPILER ${CC_PREFIX}/bin/i586-pc-mingw32-g++)
+SET(CMAKE_C_COMPILER   ${MINGW_PREFIX}/bin/i586-pc-mingw32-gcc)
+SET(CMAKE_CXX_COMPILER ${MINGW_PREFIX}/bin/i586-pc-mingw32-g++)
 
 # where is the target environment 
 SET(CMAKE_FIND_ROOT_PATH  /opt/mingw32)
 
-SET(QT_BINARY_DIR   ${CC_PREFIX}/bin)
-SET(QT_LIBRARY_DIR  ${CC_PREFIX}/lib)
-SET(QT_QTCORE_LIBRARY   ${CC_PREFIX}/lib/libQtCore4.a)
-SET(QT_INCLUDE_DIR ${CC_PREFIX}/include/qt4)
-SET(QT_QTCORE_INCLUDE_DIR ${CC_PREFIX}/include/qt4/QtCore)
-SET(QT_MKSPECS_DIR  ${CC_PREFIX}/share/qt4/mkspecs)
+SET(QT_BINARY_DIR   ${MINGW_PREFIX}/bin)
+SET(QT_LIBRARY_DIR  ${MINGW_PREFIX}/lib)
+SET(QT_QTCORE_LIBRARY   ${MINGW_PREFIX}/lib/libQtCore4.a)
+SET(QT_INCLUDE_DIR ${MINGW_PREFIX}/include/qt4)
+SET(QT_HEADERS_DIR ${MINGW_PREFIX}/include/qt4)
+SET(QT_QTCORE_INCLUDE_DIR ${MINGW_PREFIX}/include/qt4/QtCore)
+SET(QT_MKSPECS_DIR  ${MINGW_PREFIX}/share/qt4/mkspecs)
 SET(QT_MOC_EXECUTABLE  ${QT_BINARY_DIR}/moc.exe)
 SET(QT_RCC_EXECUTABLE  ${QT_BINARY_DIR}/rcc.exe)
 SET(QT_QMAKE_EXECUTABLE  /usr/bin/qmake)
@@ -32,8 +34,8 @@ SET(CMAKE_FIND_ROOT_PATH_MODE_PROGRAM NEVER)
 SET(CMAKE_FIND_ROOT_PATH_MODE_LIBRARY ONLY)
 SET(CMAKE_FIND_ROOT_PATH_MODE_INCLUDE ONLY)
 
-SET(PKG_CONFIG_EXECUTABLE ${CC_PREFIX}/bin/pkg-config)
+SET(PKG_CONFIG_EXECUTABLE ${MINGW_PREFIX}/bin/pkg-config)
 
-INCLUDE_DIRECTORIES(${CC_PREFIX}/include)
-LINK_DIRECTORIES(${CC_PREFIX}/lib ${CC_PREFIX}/bin)
+INCLUDE_DIRECTORIES(${MINGW_PREFIX}/include)
+LINK_DIRECTORIES(${MINGW_PREFIX}/lib ${MINGW_PREFIX}/bin)
 

include/Controller.h

@@ -1,8 +1,8 @@
 /*
- * controller.h - declaration of class controller, which provides a
+ * Controller.h - declaration of class controller, which provides a
  *                standard for all controllers and controller plugins
  *
- * Copyright (c) 2008 Paul Giblock <pgllama/at/gmail.com>
+ * Copyright (c) 2008-2009 Paul Giblock <pgllama/at/gmail.com>
  *
  * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
  *
@@ -32,13 +32,13 @@
 #include "mv_base.h"
 #include "journalling_object.h"
 
-class controllerDialog;
-class controller; 
+class ControllerDialog;
+class Controller; 
 
-typedef QVector<controller *> controllerVector;
+typedef QVector<Controller *> ControllerVector;
 
 
-class controller : public model, public journallingObject
+class Controller : public model, public journallingObject
 {
 	Q_OBJECT
 public:
@@ -55,10 +55,10 @@ public:
 		NumControllerTypes
 	} ;
 
-	controller( ControllerTypes _type, model * _parent,
+	Controller( ControllerTypes _type, model * _parent,
 						const QString & _display_name );
 
-	virtual ~controller();
+	virtual ~Controller();
 
 	virtual float currentValue( int _offset );
 
@@ -103,8 +103,8 @@ public:
 	virtual void loadSettings( const QDomElement & _this );
 	virtual QString nodeName( void ) const;
 
-	static controller * create( ControllerTypes _tt, model * _parent );
-	static controller * create( const QDomElement & _this,
+	static Controller * create( ControllerTypes _tt, model * _parent );
+	static Controller * create( const QDomElement & _this,
 							model * _parent );
 
 	inline static float fittedValue( float _val )
@@ -120,7 +120,7 @@ public:
 
 
 public slots:
-	virtual controllerDialog * createDialog( QWidget * _parent );
+	virtual ControllerDialog * createDialog( QWidget * _parent );
 
 	virtual void setName( const QString & _new_name )
 	{
@@ -140,7 +140,7 @@ protected:
 	QString m_name;
 	ControllerTypes m_type;
 
-	static controllerVector s_controllers;
+	static ControllerVector s_controllers;
 
 	static unsigned int s_frames;
 
@@ -149,7 +149,7 @@ signals:
 	// The value changed while the mixer isn't running (i.e: MIDI CC)
 	void valueChanged( void );
 
-	friend class controllerDialog;
+	friend class ControllerDialog;
 
 } ;
 

include/ControllerConnection.h

@@ -1,5 +1,5 @@
 /*
- * controller_connection.h - declaration of a controller connect, which
+ * ControllerConnection.h - declaration of a controller connect, which
  *              provides a definition of the link between a controller and
  *              model, also handles deferred creation of links while
  *              loading project
@@ -32,30 +32,30 @@
 #include <QtCore/QObject>
 #include <QtCore/QVector>
 
-#include "controller.h"
+#include "Controller.h"
 #include "journalling_object.h"
 
-class controllerConnection;
+class ControllerConnection;
 
-typedef QVector<controllerConnection *> controllerConnectionVector;
+typedef QVector<ControllerConnection *> ControllerConnectionVector;
 
 
-class EXPORT controllerConnection : public QObject, public journallingObject
+class EXPORT ControllerConnection : public QObject, public journallingObject
 {
 	Q_OBJECT
 public:
 
-	controllerConnection( controller * _controller );
-	controllerConnection( int _controllerId );
+	ControllerConnection( Controller * _controller );
+	ControllerConnection( int _controllerId );
 
-	virtual ~controllerConnection();
+	virtual ~ControllerConnection();
 
-	inline controller * getController( void )
+	inline Controller * getController( void )
 	{
 		return m_controller;
 	}
 
-	void setController( controller * _controller );
+	void setController( Controller * _controller );
 
 	inline void setController( int _controllerId );
 
@@ -87,19 +87,19 @@ public slots:
 
 protected:
 	//virtual controllerDialog * createDialog( QWidget * _parent );
-	controller * m_controller;
+	Controller * m_controller;
 	QString m_targetName;
 	int m_controllerId;	
 	
 	bool m_ownsController;
 
-	static controllerConnectionVector s_connections;
+	static ControllerConnectionVector s_connections;
 
 signals:
 	// The value changed while the mixer isn't running (i.e: MIDI CC)
 	void valueChanged( void );
 
-	friend class controllerConnectionDialog;
+	friend class ControllerConnectionDialog;
 };
 
 #endif

include/ControllerConnectionDialog.h

@@ -1,5 +1,5 @@
 /*
- * controller_connection_dialog.h - dialog allowing the user to create and
+ * ControllerConnectionDialog.h - dialog allowing the user to create and
  *	modify links between controllers and models
  *
  * Copyright (c) 2008  Paul Giblock <drfaygo/at/gmail.com>
@@ -31,14 +31,14 @@
 #include <QtGui/QSortFilterProxyModel>
 #include <QtGui/QStandardItemModel>
 
-#include "controller.h"
+#include "Controller.h"
 #include "automatable_model.h"
 
 
 class QLineEdit;
 class QListView;
 class QScrollArea;
-class autoDetectMidiController;
+class AutoDetectMidiController;
 class comboBox;
 class groupBox;
 class tabWidget;
@@ -48,15 +48,15 @@ class midiPortMenu;
 
 
 
-class controllerConnectionDialog : public QDialog
+class ControllerConnectionDialog : public QDialog
 {
 	Q_OBJECT
 public:
-	controllerConnectionDialog( QWidget * _parent,
+	ControllerConnectionDialog( QWidget * _parent,
 			const automatableModel * _target_model );
-	virtual ~controllerConnectionDialog();
+	virtual ~ControllerConnectionDialog();
 
-	controller * chosenController( void )
+	Controller * chosenController( void )
 	{
 		return m_controller;
 	}
@@ -91,11 +91,11 @@ private:
 	tabWidget * m_mappingBox;
 	QLineEdit * m_mappingFunction;
 
-	controller * m_controller;
+	Controller * m_controller;
 	const automatableModel * m_targetModel;
 
 	// Temporary midiController 
-	autoDetectMidiController * m_midiController;
+	AutoDetectMidiController * m_midiController;
 } ;
 
 #endif

include/ControllerDialog.h

@@ -1,8 +1,8 @@
 /*
- * controller_dialog.h - per-controller-specific view for changing a
+ * ControllerDialog.h - per-controller-specific view for changing a
  * controller's settings
  *
- * Copyright (c) 2008 Paul Giblock <drfaygo/at/gmail.com>
+ * Copyright (c) 2008-2009 Paul Giblock <drfaygo/at/gmail.com>
  * 
  * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
  *
@@ -30,25 +30,22 @@
 
 #include "mv_base.h"
 
-class controller;
+class Controller;
 
 
-class controllerDialog : public QWidget, public modelView
+class ControllerDialog : public QWidget, public modelView
 {
     Q_OBJECT
 public:
-	controllerDialog( controller * _controller, QWidget * _parent );
+	ControllerDialog( Controller * _controller, QWidget * _parent );
 
-	virtual ~controllerDialog();
+	virtual ~ControllerDialog();
 
 signals:
 	void closed();
 
 
 protected:
-/*	virtual void contextMenuEvent( QContextMenuEvent * _me ) {};
-	virtual void paintEvent( QPaintEvent * _pe ) {};
-	virtual void modelChanged( void ) {};*/
 	virtual void closeEvent( QCloseEvent * _ce );
 
 } ;

include/ControllerRackView.h

@@ -1,7 +1,7 @@
 /*
- * controller_rack_view.h - view for song's controllers
+ * ControllerRackView.h - view for song's controllers
  *
- * Copyright (c) 2008 Paul Giblock <drfaygo/at/gmail.com>
+ * Copyright (c) 2008-2009 Paul Giblock <drfaygo/at/gmail.com>
  *
  * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
  *
@@ -33,29 +33,28 @@
 
 class QPushButton;
 class QScrollArea;
-//class QVBoxLayout;
 
-class controllerView;
+class ControllerView;
 
 
-class controllerRackView : public QWidget, public serializingObject
+class ControllerRackView : public QWidget, public serializingObject
 {
 	Q_OBJECT
 public:
-	controllerRackView();
-	virtual ~controllerRackView();
+	ControllerRackView();
+	virtual ~ControllerRackView();
 
 	virtual void saveSettings( QDomDocument & _doc, QDomElement & _parent );
 	virtual void loadSettings( const QDomElement & _this );
 
 	inline virtual QString nodeName( void ) const
 	{
-		return( "controllerrackview" );
+		return( "ControllerRackView" );
 	}
 
 
 public slots:
-	void deleteController( controllerView * _view );
+	void deleteController( ControllerView * _view );
 
 
 private slots:
@@ -64,7 +63,7 @@ private slots:
 
 
 private:
-	QVector<controllerView *> m_controllerViews;
+	QVector<ControllerView *> m_controllerViews;
 
 	QVBoxLayout * m_mainLayout;
 	QScrollArea * m_scrollArea;

include/ControllerView.h

@@ -1,5 +1,5 @@
 /*
- * controller_view.h - view-component for an control
+ * ControllerView.h - view-component for an control
  *
  * Copyright (c) 2008 Paul Giblock <drfaygo/at/gmail.com>
  * 
@@ -28,7 +28,7 @@
 #include <QtGui/QWidget>
 
 #include "automatable_model.h"
-#include "controller.h"
+#include "Controller.h"
 #include "mv_base.h"
 
 class QGroupBox;
@@ -39,21 +39,21 @@ class QMdiSubWindow;
 class ledCheckBox;
 
 
-class controllerView : public QWidget, public modelView
+class ControllerView : public QWidget, public modelView
 {
 	Q_OBJECT
 public:
-	controllerView( controller * _controller, QWidget * _parent );
-	virtual ~controllerView();
+	ControllerView( Controller * _controller, QWidget * _parent );
+	virtual ~ControllerView();
 	
-	inline controller * getController( void )
+	inline Controller * getController( void )
 	{
-		return( castModel<controller>() );
+		return( castModel<Controller>() );
 	}
 
-	inline const controller * getController( void ) const
+	inline const Controller * getController( void ) const
 	{
-		return( castModel<controller>() );
+		return( castModel<Controller>() );
 	}
 
 
@@ -65,7 +65,7 @@ public slots:
 
 	
 signals:
-	void deleteController( controllerView * _view );
+	void deleteController( ControllerView * _view );
 
 
 protected:
@@ -79,7 +79,7 @@ private:
 	QPixmap m_bg;
 	ledCheckBox * m_bypass;
 	QMdiSubWindow * m_subWindow;
-	controllerDialog * m_controllerDlg;
+	ControllerDialog * m_controllerDlg;
 	bool m_show;
 
 } ;

include/LfoController.h

@@ -1,5 +1,5 @@
 /*
- * lfo_controller.h - A LFO-based controller and dialog
+ * LfoController.h - A LFO-based controller and dialog
  *
  * Copyright (c) 2008 Paul Giblock <drfaygo/at/gmail.com>
  * 
@@ -29,8 +29,8 @@
 
 #include "mv_base.h"
 #include "automatable_model.h"
-#include "controller.h"
-#include "controller_dialog.h"
+#include "Controller.h"
+#include "ControllerDialog.h"
 #include "tempo_sync_knob.h"
 #include "oscillator.h"
 
@@ -42,13 +42,13 @@ class tempoSyncKnob;
 class pixmapButton;
 class oscillator;
 
-class lfoController : public controller 
+class LfoController : public Controller 
 {
 	Q_OBJECT
 public:
-	lfoController( model * _parent );
+	LfoController( model * _parent );
 
-	virtual ~lfoController();
+	virtual ~LfoController();
 
 	virtual void saveSettings( QDomDocument & _doc, QDomElement & _this );
 	virtual void loadSettings( const QDomElement & _this );
@@ -56,7 +56,7 @@ public:
 
 
 public slots:
-	virtual controllerDialog * createDialog( QWidget * _parent );
+	virtual ControllerDialog * createDialog( QWidget * _parent );
 
 
 protected:
@@ -80,24 +80,24 @@ protected:
 protected slots:
 	void updateSampleFunction( void );
 
-	friend class lfoControllerDialog;
+	friend class LfoControllerDialog;
 } ;
 
 
 
-class lfoControllerDialog : public controllerDialog
+class LfoControllerDialog : public ControllerDialog
 {
 	Q_OBJECT
 public:
-	lfoControllerDialog( controller * _controller, QWidget * _parent );
-	virtual ~lfoControllerDialog();
+	LfoControllerDialog( Controller * _controller, QWidget * _parent );
+	virtual ~LfoControllerDialog();
 
 
 protected:
 	virtual void contextMenuEvent( QContextMenuEvent * _me );
 	virtual void modelChanged( void );
 
-	lfoController * m_lfo;
+	LfoController * m_lfo;
 
 	knob * m_baseKnob;
 	tempoSyncKnob * m_speedKnob;

include/MidiController.h

@@ -1,5 +1,5 @@
 /*
- * midi_controller.h - A controller to receive MIDI control-changes
+ * MidiController.h - A controller to receive MIDI control-changes
  *
  * Copyright (c) 2008 Paul Giblock <drfaygo/at/gmail.com>
  * 
@@ -28,7 +28,7 @@
 #include <QtGui/QWidget>
 
 #include "automatable_model.h"
-#include "controller.h"
+#include "Controller.h"
 #include "midi_event_processor.h"
 #include "midi_port.h"
 
@@ -36,12 +36,12 @@
 class midiPort;
 
 
-class midiController : public controller, public midiEventProcessor
+class MidiController : public Controller, public MidiEventProcessor
 {
 	Q_OBJECT
 public:
-	midiController( model * _parent );
-	virtual ~midiController();
+	MidiController( model * _parent );
+	virtual ~MidiController();
 
 	virtual void processInEvent( const midiEvent & _me,
 					const midiTime & _time );
@@ -61,7 +61,7 @@ public:
 
 
 public slots:
-	virtual controllerDialog * createDialog( QWidget * _parent );
+	virtual ControllerDialog * createDialog( QWidget * _parent );
 	void updateName( void );
 
 
@@ -75,8 +75,8 @@ protected:
 
 	float m_lastValue;
 
-	friend class controllerConnectionDialog;
-	friend class autoDetectMidiController;
+	friend class ControllerConnectionDialog;
+	friend class AutoDetectMidiController;
 
 } ;
 

include/PeakController.h

@@ -1,7 +1,7 @@
 /*
- * peak_controller.h - peak-controller class
+ * PeakController.h - peak-controller class
  *
- * Copyright (c) 2008 Paul Giblock <drfaygo/at/gmail.com>
+ * Copyright (c) 2008-2009 Paul Giblock <drfaygo/at/gmail.com>
  * 
  * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
  *
@@ -29,8 +29,8 @@
 
 #include "mv_base.h"
 #include "automatable_model.h"
-#include "controller.h"
-#include "controller_dialog.h"
+#include "Controller.h"
+#include "ControllerDialog.h"
 
 class automatableButtonGroup;
 class knob;
@@ -39,15 +39,15 @@ class peakControllerEffect;
 typedef QVector<peakControllerEffect *> peakControllerEffectVector;
 
 
-class EXPORT peakController : public controller
+class EXPORT PeakController : public Controller
 {
 	Q_OBJECT
 public:
-	peakController( model * _parent,
+	PeakController( model * _parent,
 		peakControllerEffect *_peak_effect = NULL );
 
 
-	virtual ~peakController();
+	virtual ~PeakController();
 
 	virtual void saveSettings( QDomDocument & _doc, QDomElement & _this );
 	virtual void loadSettings( const QDomElement & _this );
@@ -58,7 +58,7 @@ public:
 
 
 public slots:
-	virtual controllerDialog * createDialog( QWidget * _parent );
+	virtual ControllerDialog * createDialog( QWidget * _parent );
 	void handleDestroyedEffect( ); 
 
 protected:
@@ -67,24 +67,24 @@ protected:
 
 	peakControllerEffect * m_peakEffect;
 
-	friend class peakControllerDialog;
+	friend class PeakControllerDialog;
 } ;
 
 
 
-class peakControllerDialog : public controllerDialog
+class PeakControllerDialog : public ControllerDialog
 {
 	Q_OBJECT
 public:
-	peakControllerDialog( controller * _controller, QWidget * _parent );
-	virtual ~peakControllerDialog();
+	PeakControllerDialog( Controller * _controller, QWidget * _parent );
+	virtual ~PeakControllerDialog();
 
 protected:
 	virtual void contextMenuEvent( QContextMenuEvent * _me );
 	virtual void paintEvent( QPaintEvent * _pe );
 	virtual void modelChanged( void );
 
-	peakController * m_peakController;
+	PeakController * m_peakController;
 
 } ;
 

include/aeffectx.h

@@ -118,7 +118,7 @@ const int kVstTempoValid = 1 << 10;
 const int kVstTransportPlaying = 1 << 1;
 
 
-class remoteVstPlugin;
+class RemoteVstPlugin;
 
 
 class VstMidiEvent
@@ -221,7 +221,7 @@ public:
 	// flags 24-27
 	int flags;
 	// Fill somewhere 28-2b
-	remoteVstPlugin * user;
+	RemoteVstPlugin * user;
 	// Zeroes 2c-2f 30-33 34-37 38-3b
 	char empty3[4 + 4 + 4 + 4];
 	// 1.0f 3c-3f

include/automatable_model.h

@@ -59,7 +59,7 @@
 
 
 
-class controllerConnection;
+class ControllerConnection;
 
 
 class EXPORT automatableModel : public model, public journallingObject
@@ -94,13 +94,13 @@ public:
 
 	bool isAutomated( void ) const;
 
-	inline controllerConnection * getControllerConnection( void ) const
+	inline ControllerConnection * getControllerConnection( void ) const
 	{
 		return m_controllerConnection;
 	}
 
 
-	void setControllerConnection( controllerConnection * _c );
+	void setControllerConnection( ControllerConnection * _c );
 
 
 	template<class T>
@@ -250,7 +250,7 @@ private:
 	bool m_hasLinkedModels;
 
 
-	controllerConnection * m_controllerConnection;
+	ControllerConnection * m_controllerConnection;
 
 
 	static float __copiedValue;

include/engine.h

@@ -46,7 +46,7 @@ class projectNotes;
 class song;
 class songEditor;
 class ladspa2LMMS;
-class controllerRackView;
+class ControllerRackView;
 
 
 class EXPORT engine
@@ -142,7 +142,7 @@ public:
 		return( s_dummyTC );
 	}
 
-	static controllerRackView * getControllerRackView( void )
+	static ControllerRackView * getControllerRackView( void )
 	{
 		return( s_controllerRackView );
 	}
@@ -171,7 +171,7 @@ private:
 	static bbTrackContainer * s_bbTrackContainer;
 	static projectJournal * s_projectJournal;
 	static dummyTrackContainer * s_dummyTC;
-	static controllerRackView * s_controllerRackView;
+	static ControllerRackView * s_controllerRackView;
 
 	// GUI
 	static mainWindow * s_mainWindow;

include/instrument.h

@@ -2,8 +2,8 @@
  * instrument.h - declaration of class instrument, which provides a
  *                standard interface for all instrument plugins
  *
- * Copyright (c) 2005-2008 Tobias Doerffel <tobydox/at/users.sourceforge.net>
- * 
+ * Copyright (c) 2005-2009 Tobias Doerffel <tobydox/at/users.sourceforge.net>
+ *
  * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
  *
  * This program is free software; you can redistribute it and/or
@@ -23,7 +23,6 @@
  *
  */
 
-
 #ifndef _INSTRUMENT_H
 #define _INSTRUMENT_H
 
@@ -80,20 +79,20 @@ public:
 	// if no envelope is active - such instruments can re-implement this
 	// method for returning how many frames they at least like to have for
 	// release
-	virtual f_cnt_t desiredReleaseFrames( void ) const
+	virtual f_cnt_t desiredReleaseFrames() const
 	{
 		return 0;
 	}
 
 	// return false if instrument is not bendable
-	inline virtual bool isBendable( void ) const
+	inline virtual bool isBendable() const
 	{
 		return true;
 	}
 
 	// return true if instruments reacts to MIDI events passed to
 	// handleMidiEvent() rather than playNote() & Co
-	inline virtual bool isMidiBased( void ) const
+	inline virtual bool isMidiBased() const
 	{
 		return false;
 	}
@@ -106,7 +105,7 @@ public:
 		return false;
 	}
 
-	virtual QString fullDisplayName( void ) const;
+	virtual QString fullDisplayName() const;
 
 	// --------------------------------------------------------------------
 	// provided functions:
@@ -121,7 +120,7 @@ public:
 
 
 protected:
-	inline instrumentTrack * getInstrumentTrack( void ) const
+	inline instrumentTrack * getInstrumentTrack() const
 	{
 		return m_instrumentTrack;
 	}
@@ -137,5 +136,4 @@ private:
 
 } ;
 
-
 #endif

include/instrument_track.h

@@ -34,6 +34,7 @@
 #include "instrument_sound_shaping.h"
 #include "midi_event_processor.h"
 #include "midi_port.h"
+#include "note_play_handle.h"
 #include "piano.h"
 #include "track.h"
 
@@ -53,12 +54,11 @@ class midiPortMenu;
 class multimediaProject;
 class notePlayHandle;
 class pluginView;
-class presetPreviewPlayHandle;
 class tabWidget;
 class trackLabelButton;
 
 
-class EXPORT instrumentTrack : public track, public midiEventProcessor
+class EXPORT instrumentTrack : public track, public MidiEventProcessor
 {
 	Q_OBJECT
 	mapPropertyFromModel(int,getVolume,setVolume,m_volumeModel);
@@ -76,7 +76,8 @@ public:
 					const midiTime & _time );
 	virtual void processOutEvent( const midiEvent & _me,
 						const midiTime & _time );
-
+	// silence all running notes played by this track
+	void silenceAllNotes();
 
 	f_cnt_t beatLen( notePlayHandle * _n ) const;
 
@@ -195,8 +196,6 @@ protected:
 	{
 		return "instrumenttrack";
 	}
-	// invalidates all note-play-handles linked to this instrument
-	void invalidateAllMyNPH( void );
 
 
 protected slots:
@@ -213,8 +212,7 @@ private:
 
 	intModel m_baseNoteModel;
 
-	QList<notePlayHandle *> m_processHandles;
-
+	NotePlayHandleList m_processHandles;
 
 	floatModel m_volumeModel;
 	floatModel m_panningModel;
@@ -233,7 +231,6 @@ private:
 	friend class instrumentTrackView;
 	friend class instrumentTrackWindow;
 	friend class notePlayHandle;
-	friend class presetPreviewPlayHandle;
 	friend class flpImport;
 
 } ;

include/midi.h

@@ -2,7 +2,7 @@
  * midi.h - constants, structs etc. concerning MIDI
  *
  * Copyright (c) 2005-2009 Tobias Doerffel <tobydox/at/users.sourceforge.net>
- * 
+ *
  * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
  *
  * This program is free software; you can redistribute it and/or
@@ -22,7 +22,6 @@
  *
  */
 
-
 #ifndef _MIDI_H
 #define _MIDI_H
 

include/midi_event_processor.h

@@ -1,8 +1,8 @@
 /*
  * midi_event_processor.h - base-class for midi-processing classes
  *
- * Copyright (c) 2005-2008 Tobias Doerffel <tobydox/at/users.sourceforge.net>
- * 
+ * Copyright (c) 2005-2009 Tobias Doerffel <tobydox/at/users.sourceforge.net>
+ *
  * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
  *
  * This program is free software; you can redistribute it and/or
@@ -22,7 +22,6 @@
  *
  */
 
-
 #ifndef _MIDI_EVENT_PROCESSOR_H
 #define _MIDI_EVENT_PROCESSOR_H
 
@@ -31,14 +30,14 @@ class midiTime;
 
 
 // all classes being able to process MIDI-events should inherit from this
-class midiEventProcessor
+class MidiEventProcessor
 {
 public:
-	inline midiEventProcessor( void )
+	inline MidiEventProcessor()
 	{
 	}
 
-	virtual inline ~midiEventProcessor()
+	virtual inline ~MidiEventProcessor()
 	{
 	}
 

include/midi_port.h

@@ -36,7 +36,7 @@
 
 
 class midiClient;
-class midiEventProcessor;
+class MidiEventProcessor;
 class midiPortMenu;
 class midiTime;
 
@@ -74,30 +74,34 @@ public:
 
 	midiPort( const QString & _name,
 			midiClient * _mc,
-			midiEventProcessor * _mep,
+			MidiEventProcessor * _mep,
 			model * _parent = NULL,
 			Modes _mode = Disabled );
 	virtual ~midiPort();
 
 	void setName( const QString & _name );
 
-	inline Modes mode( void ) const
+	inline Modes mode() const
 	{
-		return( m_mode );
+		return m_mode;
 	}
 
 	void setMode( Modes _mode );
 
-	inline bool inputEnabled( void ) const
+	inline bool inputEnabled() const
 	{
-		return( mode() == Input || mode() == Duplex );
+		return mode() == Input || mode() == Duplex;
 	}
 
-	inline bool outputEnabled( void ) const
+	inline bool outputEnabled() const
 	{
-		return( mode() == Output || mode() == Duplex );
+		return mode() == Output || mode() == Duplex;
 	}
 
+	inline int realOutputChannel() const
+	{
+		return outputChannel() - 1;
+	}
 
 	void processInEvent( const midiEvent & _me, const midiTime & _time );
 	void processOutEvent( const midiEvent & _me, const midiTime & _time );
@@ -106,9 +110,9 @@ public:
 	virtual void saveSettings( QDomDocument & _doc, QDomElement & _parent );
 	virtual void loadSettings( const QDomElement & _this );
 
-	virtual QString nodeName( void ) const
+	virtual QString nodeName() const
 	{
-		return( "midiport" );
+		return "midiport";
 	}
 
 	void subscribeReadablePort( const QString & _port,
@@ -116,14 +120,14 @@ public:
 	void subscribeWritablePort( const QString & _port,
 						bool _subscribe = TRUE );
 
-	const map & readablePorts( void ) const
+	const map & readablePorts() const
 	{
-		return( m_readablePorts );
+		return m_readablePorts;
 	}
 
-	const map & writablePorts( void ) const
+	const map & writablePorts() const
 	{
-		return( m_writablePorts );
+		return m_writablePorts;
 	}
 
 	midiPortMenu * m_readablePortsMenu;
@@ -131,18 +135,18 @@ public:
 
 
 public slots:
-	void updateMidiPortMode( void );
+	void updateMidiPortMode();
 
 
 private slots:
-	void updateReadablePorts( void );
-	void updateWritablePorts( void );
-	void updateOutputProgram( void );
+	void updateReadablePorts();
+	void updateWritablePorts();
+	void updateOutputProgram();
 
 
 private:
 	midiClient * m_midiClient;
-	midiEventProcessor * m_midiEventProcessor;
+	MidiEventProcessor * m_midiEventProcessor;
 
 	Modes m_mode;
 
@@ -160,14 +164,14 @@ private:
 	map m_writablePorts;
 
 
-	friend class controllerConnectionDialog;
+	friend class ControllerConnectionDialog;
 	friend class instrumentMidiIOView;
 
 
 signals:
-	void readablePortsChanged( void );
-	void writablePortsChanged( void );
-	void modeChanged( void );
+	void readablePortsChanged();
+	void writablePortsChanged();
+	void modeChanged();
 
 
 } ;

include/midi_time.h

@@ -135,7 +135,7 @@ public:
 
 	static int stepsPerTact( void )
 	{
-		return ticksPerTact() / DefaultBeatsPerTact;
+		return qMax( 1, ticksPerTact() / DefaultBeatsPerTact );
 	}
 
 	static void setTicksPerTact( tick_t _tpt )

include/mixer.h

@@ -234,7 +234,7 @@ public:
 
 	void removePlayHandle( playHandle * _ph );
 
-	inline playHandleVector & playHandles( void )
+	inline PlayHandleList & playHandles( void )
 	{
 		return m_playHandles;
 	}
@@ -443,8 +443,8 @@ private:
 	QWaitCondition m_queueReadyWaitCond;
 
 
-	playHandleVector m_playHandles;
-	constPlayHandleVector m_playHandlesToRemove;
+	PlayHandleList m_playHandles;
+	ConstPlayHandleList m_playHandlesToRemove;
 
 	struct qualitySettings m_qualitySettings;
 	float m_masterGain;

include/mmp.h

@@ -1,7 +1,7 @@
 /*
  * mmp.h - class for reading and writing multimedia-project-files
  *
- * Copyright (c) 2004-2008 Tobias Doerffel <tobydox/at/users.sourceforge.net>
+ * Copyright (c) 2004-2009 Tobias Doerffel <tobydox/at/users.sourceforge.net>
  * 
  * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
  *
@@ -52,9 +52,8 @@ public:
 	} ;
 
 
-	multimediaProject( const QString & _in_file_name,
-						bool _is_filename = TRUE,
-						bool _upgrade = TRUE );
+	multimediaProject( const QString & _fileName );
+	multimediaProject( const QByteArray & _data );
 	multimediaProject( ProjectTypes _project_type );
 	virtual ~multimediaProject();
 
@@ -76,8 +75,6 @@ public:
 		return( m_type );
 	}
 
-	static ProjectTypes typeOfFile( const QString & _fn );
-
 
 private:
 	static ProjectTypes type( const QString & _type_name );
@@ -87,6 +84,8 @@ private:
 
 	void upgrade( void );
 
+	void loadData( const QByteArray & _data, const QString & _sourceFile );
+
 
 	struct EXPORT typeDescStruct
 	{

include/note.h

@@ -243,7 +243,7 @@ private:
 } ;
 
 
-typedef QVector<note *> noteVector;
+typedef QVector<note *> NoteVector;
 
 
 #endif

include/note_play_handle.h

@@ -2,7 +2,7 @@
  * note_play_handle.h - declaration of class notePlayHandle which is needed
  *                      by LMMS-Play-Engine
  *
- * Copyright (c) 2004-2008 Tobias Doerffel <tobydox/at/users.sourceforge.net>
+ * Copyright (c) 2004-2009 Tobias Doerffel <tobydox/at/users.sourceforge.net>
  * 
  * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
  *
@@ -30,16 +30,16 @@
 #include "lmmsconfig.h"
 #include "mixer.h"
 #include "note.h"
-#include "instrument.h"
-#include "instrument_track.h"
 #include "engine.h"
+#include "track.h"
 
 
+class instrumentTrack;
 class notePlayHandle;
 
 template<ch_cnt_t=DEFAULT_CHANNELS> class basicFilters;
-typedef QVector<notePlayHandle *> notePlayHandleVector;
-typedef QVector<const notePlayHandle *> constNotePlayHandleVector;
+typedef QList<notePlayHandle *> NotePlayHandleList;
+typedef QList<const notePlayHandle *> ConstNotePlayHandleList;
 
 
 class EXPORT notePlayHandle : public playHandle, public note
@@ -79,55 +79,6 @@ public:
 		return( m_released && framesLeft() <= 0 );
 	}
 
-	bool willFinishThisPeriod( void ) const
-	{
-		f_cnt_t rftd = m_releaseFramesToDo;
-		f_cnt_t fbr = m_framesBeforeRelease;
-		f_cnt_t rfd = m_releaseFramesDone;
-		if( m_released == TRUE )
-		{
-			f_cnt_t todo = engine::getMixer()->framesPerPeriod();
-			if( isArpeggioBaseNote() )
-			{
-				rftd = rfd + 2 *
-					engine::getMixer()->framesPerPeriod();
-			}
-			if( fbr )
-			{
-				if( fbr <=
-					engine::getMixer()->framesPerPeriod() )
-				{
-					todo -= fbr;
-					fbr = 0;
-				}
-				else
-				{
-					todo = 0;
-					fbr -=
-					engine::getMixer()->framesPerPeriod();
-				}
-			}
-			if( todo && rfd < rftd )
-			{
-				if( rftd - rfd >= todo )
-				{
-					rfd += todo;
-				}
-				else
-				{
-					rfd = rftd;
-				}
-			}
-		}
-
-		if( isArpeggioBaseNote() && m_subNotes.size() == 0 )
-		{
-			rfd = rftd;
-		}
-
-		return( ( m_released && fbr == 0 && rfd >= rftd ) );
-	}
-
 	f_cnt_t framesLeft( void ) const;
 
 	inline fpp_t framesLeftForCurrentPeriod( void ) const
@@ -202,11 +153,7 @@ public:
 	}
 
 	// returns whether note is base-note for arpeggio
-	inline bool isArpeggioBaseNote( void ) const
-	{
-		return( isBaseNote() && ( m_partOfArpeggio || 
-				m_instrumentTrack->arpeggiatorEnabled() ) );
-	}
+	bool isArpeggioBaseNote( void ) const;
 
 	inline bool isMuted( void ) const
 	{
@@ -219,15 +166,15 @@ public:
 	// belonging to this instrument-track - used by arpeggiator
 	int index( void ) const;
 
-	// note-play-handles belonging to given channel, if _all_ph = TRUE,
+	// note-play-handles belonging to given channel, if _all_ph = true,
 	// also released note-play-handles are returned
-	static constNotePlayHandleVector nphsOfInstrumentTrack(
-			const instrumentTrack * _ct, bool _all_ph = FALSE );
+	static ConstNotePlayHandleList nphsOfInstrumentTrack(
+			const instrumentTrack * _ct, bool _all_ph = false );
 
 	// return whether given note-play-handle is equal to *this
 	bool operator==( const notePlayHandle & _nph ) const;
 
-	bool bbTrackMuted( void )
+	inline bool bbTrackMuted( void )
 	{
 		return( m_bbTrack && m_bbTrack->isMuted() );
 	}
@@ -288,7 +235,7 @@ private:
 					// played after release
 	f_cnt_t m_releaseFramesDone;	// number of frames done after
 					// release of note
-	notePlayHandleVector m_subNotes;// used for chords and arpeggios
+	NotePlayHandleList m_subNotes;	// used for chords and arpeggios
 	volatile bool m_released;	// indicates whether note is released
 	bool m_baseNote;		// indicates whether note is a
 					// base-note (i.e. no sub-note)

include/panning.h

@@ -3,7 +3,7 @@
  *             panning of a note
  *
  * Copyright (c) 2004-2009 Tobias Doerffel <tobydox/at/users.sourceforge.net>
- * 
+ *
  * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
  *
  * This program is free software; you can redistribute it and/or
@@ -41,9 +41,10 @@ inline stereoVolumeVector panningToVolumeVector( panning_t _p,
 	return v;
 }
 
-inline Sint16 panningToMidi( panning_t _p )
+
+inline int panningToMidi( panning_t _p )
 {
-	return MidiMinPanning + (panning_t) (
+	return MidiMinPanning + (int) (
 			  ( (float)( _p - PanningLeft ) ) / 
 			  ( (float)( PanningRight - PanningLeft ) ) *
 			  ( (float)( MidiMaxPanning - MidiMinPanning ) ) );

include/pattern.h

@@ -78,7 +78,7 @@ public:
 	void rearrangeAllNotes( void );
 	void clearNotes( void );
 
-	inline const noteVector & notes( void )
+	inline const NoteVector & notes( void ) const
 	{
 		return m_notes;
 	}
@@ -154,7 +154,7 @@ private:
 	PatternTypes m_patternType;
 
 	// data-stuff
-	noteVector m_notes;
+	NoteVector m_notes;
 	int m_steps;
 
 	// pattern freezing

include/piano_roll.h

@@ -109,7 +109,7 @@ protected:
 					int  _width, note * _n );
 	void removeSelection( void );
 	void selectAll( void );
-	void getSelectedNotes( noteVector & _selected_notes );
+	void getSelectedNotes( NoteVector & _selected_notes );
 
 
 protected slots:
@@ -150,7 +150,7 @@ private:
 		ModeDraw,
 		ModeErase,
 		ModeSelect,
-		ModeOpen
+		ModeEditDetuning,
 	};
 
 	enum actions
@@ -306,13 +306,12 @@ private:
 	timeLine * m_timeLine;
 	bool m_scrollBack;
 
-	void copy_to_clipboard( const noteVector & _notes ) const;
+	void copy_to_clipboard( const NoteVector & _notes ) const;
 
 	void drawDetuningInfo( QPainter & _p, note * _n, int _x, int _y );
 	bool mouseOverNote( void );
 	note * noteUnderMouse( void );
-	noteVector::const_iterator noteIteratorUnderMouse( void );
-	
+
 	// turn a selection rectangle into selected notes
 	void computeSelectedNotes( bool shift ); 
 	void clearSelectedNotes( void );

include/play_handle.h

@@ -1,8 +1,8 @@
 /*
  * play_handle.h - base-class playHandle - core of rendering engine
  *
- * Copyright (c) 2004-2008 Tobias Doerffel <tobydox/at/users.sourceforge.net>
- * 
+ * Copyright (c) 2004-2009 Tobias Doerffel <tobydox/at/users.sourceforge.net>
+ *
  * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
  *
  * This program is free software; you can redistribute it and/or
@@ -22,7 +22,6 @@
  *
  */
 
-
 #ifndef _PLAY_HANDLE_H
 #define _PLAY_HANDLE_H
 
@@ -98,8 +97,8 @@ private:
 } ;
 
 
-typedef QVector<playHandle *> playHandleVector;
-typedef QVector<const playHandle *> constPlayHandleVector;
+typedef QList<playHandle *> PlayHandleList;
+typedef QList<const playHandle *> ConstPlayHandleList;
 
 
 #endif

include/preset_preview_play_handle.h

@@ -48,7 +48,7 @@ public:
 
 	static void init( void );
 	static void cleanup( void );
-	static constNotePlayHandleVector nphsOfInstrumentTrack(
+	static ConstNotePlayHandleList nphsOfInstrumentTrack(
 						const instrumentTrack * _ct );
 
 

include/remote_plugin.h

@@ -36,6 +36,16 @@
 #include <cassert>
 
 #ifdef LMMS_BUILD_WIN32
+#define USE_QT_SEMAPHORES
+#define USE_QT_SHMEM
+#endif
+
+#ifdef LMMS_BUILD_APPLE
+#define USE_QT_SEMAPHORES
+#endif
+
+
+#ifdef USE_QT_SEMAPHORES
 
 #ifdef LMMS_HAVE_PROCESS_H
 #include <process.h>
@@ -44,22 +54,38 @@
 #include <Qt/qglobal.h>
 
 #if QT_VERSION >= 0x040400
-#include <QtCore/QSharedMemory>
 #include <QtCore/QSystemSemaphore>
 #else
-#error win32-build requires at least Qt 4.4.0
+#error building LMMS on this platform requires at least Qt 4.4.0
 #endif
 
-typedef int32_t key_t;
-
-#else
-
-#define USE_NATIVE_SHMEM
+#else /* USE_QT_SEMAPHORES */
 
 #ifdef LMMS_HAVE_SYS_IPC_H
 #include <sys/ipc.h>
 #endif
 
+#ifdef LMMS_HAVE_SEMAPHORE_H
+#include <semaphore.h>
+#endif
+
+#endif
+
+
+#ifdef USE_QT_SHMEM
+
+#include <Qt/qglobal.h>
+
+#if QT_VERSION >= 0x040400
+#include <QtCore/QSharedMemory>
+#else
+#error building LMMS on this platform requires at least Qt 4.4.0
+#endif
+
+typedef int32_t key_t;
+
+#else /* USE_QT_SHMEM */
+
 #ifdef LMMS_HAVE_SYS_SHM_H
 #include <sys/shm.h>
 #endif
@@ -71,9 +97,6 @@ typedef int32_t key_t;
 #endif
 
 
-#ifdef LMMS_HAVE_SEMAPHORE_H
-#include <semaphore.h>
-#endif
 
 #ifdef LMMS_HAVE_LOCALE_H
 #include <locale.h>
@@ -102,7 +125,7 @@ class shmFifo
 	// and 64 bit platforms
 	union sem32_t
 	{
-#ifdef LMMS_HAVE_SEMAPHORE_H
+#ifndef USE_QT_SEMAPHORES
 		sem_t sem;
 #endif
 		int semKey;
@@ -124,13 +147,13 @@ public:
 		m_invalid( false ),
 		m_master( true ),
 		m_shmKey( 0 ),
-#ifdef USE_NATIVE_SHMEM
-		m_shmID( -1 ),
-#else
+#ifdef USE_QT_SHMEM
 		m_shmObj(),
+#else
+		m_shmID( -1 ),
 #endif
 		m_data( NULL ),
-#ifdef LMMS_BUILD_WIN32
+#ifdef USE_QT_SEMAPHORES
 		m_dataSem( QString::null ),
 		m_messageSem( QString::null ),
 #else
@@ -139,13 +162,7 @@ public:
 #endif
 		m_lockDepth( 0 )
 	{
-#ifdef USE_NATIVE_SHMEM
-		while( ( m_shmID = shmget( ++m_shmKey, sizeof( shmData ),
-					IPC_CREAT | IPC_EXCL | 0600 ) ) == -1 )
-		{
-		}
-		m_data = (shmData *) shmat( m_shmID, 0, 0 );
-#else
+#ifdef USE_QT_SHMEM
 		do
 		{
 			m_shmObj.setKey( QString( "%1" ).arg( ++m_shmKey ) );
@@ -153,10 +170,16 @@ public:
 		} while( m_shmObj.error() != QSharedMemory::NoError );
 
 		m_data = (shmData *) m_shmObj.data();
+#else
+		while( ( m_shmID = shmget( ++m_shmKey, sizeof( shmData ),
+					IPC_CREAT | IPC_EXCL | 0600 ) ) == -1 )
+		{
+		}
+		m_data = (shmData *) shmat( m_shmID, 0, 0 );
 #endif
 		assert( m_data != NULL );
 		m_data->startPtr = m_data->endPtr = 0;
-#ifdef LMMS_BUILD_WIN32
+#ifdef USE_QT_SEMAPHORES
 		static int k = 0;
 		m_data->dataSem.semKey = ( getpid()<<10 ) + ++k;
 		m_data->messageSem.semKey = ( getpid()<<10 ) + ++k;
@@ -187,13 +210,13 @@ public:
 		m_invalid( false ),
 		m_master( false ),
 		m_shmKey( 0 ),
-#ifdef USE_NATIVE_SHMEM
-		m_shmID( shmget( _shm_key, 0, 0 ) ),
-#else
+#ifdef USE_QT_SHMEM
 		m_shmObj( QString::number( _shm_key ) ),
+#else
+		m_shmID( shmget( _shm_key, 0, 0 ) ),
 #endif
 		m_data( NULL ),
-#ifdef LMMS_BUILD_WIN32
+#ifdef USE_QT_SEMAPHORES
 		m_dataSem( QString::null ),
 		m_messageSem( QString::null ),
 #else
@@ -202,19 +225,19 @@ public:
 #endif
 		m_lockDepth( 0 )
 	{
-#ifdef USE_NATIVE_SHMEM
-		if( m_shmID != -1 )
-		{
-			m_data = (shmData *) shmat( m_shmID, 0, 0 );
-		}
-#else
+#ifdef USE_QT_SHMEM
 		if( m_shmObj.attach() )
 		{
 			m_data = (shmData *) m_shmObj.data();
 		}
+#else
+		if( m_shmID != -1 )
+		{
+			m_data = (shmData *) shmat( m_shmID, 0, 0 );
+		}
 #endif
 		assert( m_data != NULL );
-#ifdef LMMS_BUILD_WIN32
+#ifdef USE_QT_SEMAPHORES
 		m_dataSem.setKey( QString::number( m_data->dataSem.semKey ) );
 		m_messageSem.setKey( QString::number(
 						m_data->messageSem.semKey ) );
@@ -226,16 +249,16 @@ public:
 
 	~shmFifo()
 	{
-#ifdef USE_NATIVE_SHMEM
+#ifndef USE_QT_SHMEM
 		shmdt( m_data );
 #endif
 		// master?
 		if( m_master )
 		{
-#ifdef USE_NATIVE_SHMEM
+#ifndef USE_QT_SHMEM
 			shmctl( m_shmID, IPC_RMID, NULL );
 #endif
-#ifndef LMMS_BUILD_WIN32
+#ifndef USE_QT_SEMAPHORES
 			sem_destroy( m_dataSem );
 			sem_destroy( m_messageSem );
 #endif
@@ -263,7 +286,7 @@ public:
 	{
 		if( !isInvalid() && ++m_lockDepth == 1 )
 		{
-#ifdef LMMS_BUILD_WIN32
+#ifdef USE_QT_SEMAPHORES
 			m_dataSem.acquire();
 #else
 			sem_wait( m_dataSem );
@@ -278,7 +301,7 @@ public:
 		{
 			if( --m_lockDepth == 0 )
 			{
-#ifdef LMMS_BUILD_WIN32
+#ifdef USE_QT_SEMAPHORES
 				m_dataSem.release();
 #else
 				sem_post( m_dataSem );
@@ -292,7 +315,7 @@ public:
 	{
 		if( !isInvalid() )
 		{
-#ifdef LMMS_BUILD_WIN32
+#ifdef USE_QT_SEMAPHORES
 			m_messageSem.acquire();
 #else
 			sem_wait( m_messageSem );
@@ -303,7 +326,7 @@ public:
 	// increase message-semaphore
 	inline void messageSent( void )
 	{
-#ifdef LMMS_BUILD_WIN32
+#ifdef USE_QT_SEMAPHORES
 		m_messageSem.release();
 #else
 		sem_post( m_messageSem );
@@ -353,7 +376,7 @@ public:
 		{
 			return false;
 		}
-#ifdef LMMS_BUILD_WIN32
+#ifdef USE_QT_SEMAPHORES
 		lock();
 		const bool empty = ( m_data->startPtr == m_data->endPtr );
 		unlock();
@@ -447,14 +470,14 @@ private:
 	volatile bool m_invalid;
 	bool m_master;
 	key_t m_shmKey;
-#ifdef USE_NATIVE_SHMEM
-	int m_shmID;
-#else
+#ifdef USE_QT_SHMEM
 	QSharedMemory m_shmObj;
+#else
+	int m_shmID;
 #endif
 	size_t m_shmSize;
 	shmData * m_data;
-#ifdef LMMS_BUILD_WIN32
+#ifdef USE_QT_SEMAPHORES
 	QSystemSemaphore m_dataSem;
 	QSystemSemaphore m_messageSem;
 #else
@@ -755,10 +778,10 @@ private:
 
 	QMutex m_commMutex;
 	bool m_splitChannels;
-#ifdef USE_NATIVE_SHMEM
-	int m_shmID;
-#else
+#ifdef USE_QT_SHMEM
 	QSharedMemory m_shmObj;
+#else
+	int m_shmID;
 #endif
 	size_t m_shmSize;
 	float * m_shm;
@@ -845,7 +868,7 @@ private:
 	void setShmKey( key_t _key, int _size );
 	void doProcessing( void );
 
-#ifndef USE_NATIVE_SHMEM
+#ifdef USE_QT_SHMEM
 	QSharedMemory m_shmObj;
 #endif
 	float * m_shm;
@@ -972,7 +995,7 @@ remotePluginBase::message remotePluginBase::waitForMessage(
 remotePluginClient::remotePluginClient( key_t _shm_in, key_t _shm_out ) :
 	remotePluginBase( new shmFifo( _shm_in ),
 				new shmFifo( _shm_out ) ),
-#ifndef USE_NATIVE_SHMEM
+#ifdef USE_QT_SHMEM
 	m_shmObj(),
 #endif
 	m_shm( NULL ),
@@ -992,7 +1015,7 @@ remotePluginClient::~remotePluginClient()
 {
 	sendMessage( IdQuit );
 
-#ifdef USE_NATIVE_SHMEM
+#ifndef USE_QT_SHMEM
 	shmdt( m_shm );
 #endif
 }
@@ -1063,7 +1086,17 @@ bool remotePluginClient::processMessage( const message & _m )
 
 void remotePluginClient::setShmKey( key_t _key, int _size )
 {
-#ifdef USE_NATIVE_SHMEM
+#ifdef USE_QT_SHMEM
+	m_shmObj.setKey( QString::number( _key ) );
+	if( m_shmObj.attach() )
+	{
+		m_shm = (float *) m_shmObj.data();
+	}
+	else
+	{
+		fprintf( stderr, "failed getting shared memory\n" );
+	}
+#else
 	if( m_shm != NULL )
 	{
 		shmdt( m_shm );
@@ -1085,16 +1118,6 @@ void remotePluginClient::setShmKey( key_t _key, int _size )
 	{
 		m_shm = (float *) shmat( shm_id, 0, 0 );
 	}
-#else
-	m_shmObj.setKey( QString::number( _key ) );
-	if( m_shmObj.attach() )
-	{
-		m_shm = (float *) m_shmObj.data();
-	}
-	else
-	{
-		fprintf( stderr, "failed getting shared memory\n" );
-	}
 #endif
 }
 
@@ -1115,4 +1138,6 @@ void remotePluginClient::doProcessing( void )
 
 #endif
 
+#define QSTR_TO_STDSTR(s)	std::string( s.toUtf8().constData() )
+
 #endif

include/sample_buffer.h

@@ -2,7 +2,7 @@
  * sample_buffer.h - container-class sampleBuffer
  *
  * Copyright (c) 2005-2009 Tobias Doerffel <tobydox/at/users.sourceforge.net>
- * 
+ *
  * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
  *
  * This program is free software; you can redistribute it and/or
@@ -49,7 +49,7 @@ public:
 	class EXPORT handleState
 	{
 	public:
-		handleState( bool _varying_pitch = FALSE );
+		handleState( bool _varying_pitch = false );
 		virtual ~handleState();
 
 
@@ -66,16 +66,16 @@ public:
 	// constructor which either loads sample _audio_file or decodes
 	// base64-data out of string
 	sampleBuffer( const QString & _audio_file = QString(),
-						bool _is_base64_data = FALSE );
+						bool _is_base64_data = false );
 	sampleBuffer( const sampleFrame * _data, const f_cnt_t _frames );
 	sampleBuffer( const f_cnt_t _frames );
-	
+
 	virtual ~sampleBuffer();
 
 	bool play( sampleFrame * _ab, handleState * _state,
 				const fpp_t _frames,
 				const float _freq,
-				const bool _looped = FALSE );
+				const bool _looped = false );
 
 	void visualize( QPainter & _p, const QRect & _dr, const QRect & _clip );
 	inline void visualize( QPainter & _p, const QRect & _dr )
@@ -83,17 +83,17 @@ public:
 		visualize( _p, _dr, _dr );
 	}
 
-	inline const QString & audioFile( void ) const
+	inline const QString & audioFile() const
 	{
 		return m_audioFile;
 	}
 
-	inline f_cnt_t startFrame( void ) const
+	inline f_cnt_t startFrame() const
 	{
 		return m_startFrame;
 	}
 
-	inline f_cnt_t endFrame( void ) const
+	inline f_cnt_t endFrame() const
 	{
 		return m_endFrame;
 	}
@@ -112,22 +112,22 @@ public:
 		m_varLock.unlock();
 	}
 
-	inline f_cnt_t frames( void ) const
+	inline f_cnt_t frames() const
 	{
 		return m_frames;
 	}
 
-	inline float amplification( void ) const
+	inline float amplification() const
 	{
 		return m_amplification;
 	}
 
-	inline bool reversed( void ) const
+	inline bool reversed() const
 	{
 		return m_reversed;
 	}
 
-	inline float frequency( void ) const
+	inline float frequency() const
 	{
 		return m_frequency;
 	}
@@ -146,12 +146,12 @@ public:
 		m_varLock.unlock();
 	}
 
-	inline const sampleFrame * data( void ) const
+	inline const sampleFrame * data() const
 	{
 		return m_data;
 	}
 
-	QString openAudioFile( void ) const;
+	QString openAudioFile() const;
 
 	QString & toBase64( QString & _dst ) const;
 
@@ -170,7 +170,7 @@ public:
 	}
 
 	void normalizeSampleRate( const sample_rate_t _src_sr,
-						bool _keep_settings = FALSE );
+						bool _keep_settings = false );
 
 	inline sample_t userWaveSample( const float _sample ) const
 	{
@@ -207,7 +207,7 @@ public slots:
 
 
 private:
-	void update( bool _keep_settings = FALSE );
+	void update( bool _keep_settings = false );
 
 
 	f_cnt_t decodeSampleSF( const char * _f, int_sample_t * & _buf,
@@ -222,7 +222,6 @@ private:
 						ch_cnt_t & _channels,
 						sample_rate_t & _sample_rate );
 
-
 	QString m_audioFile;
 	sampleFrame * m_origData;
 	f_cnt_t m_origFrames;
@@ -245,7 +244,7 @@ private:
 
 
 signals:
-	void sampleUpdated( void );
+	void sampleUpdated();
 
 } ;
 

include/song.h

@@ -30,7 +30,7 @@
 
 #include "track_container.h"
 #include "automatable_model.h"
-#include "controller.h"
+#include "Controller.h"
 #include "meter_model.h"
 
 
@@ -177,11 +177,11 @@ public:
 		return false;
 	}
 
-	void addController( controller * _c );
-	void removeController( controller * _c );
+	void addController( Controller * _c );
+	void removeController( Controller * _c );
 	
 
-	const controllerVector & controllers( void ) const
+	const ControllerVector & controllers( void ) const
 	{
 		return m_controllers;
 	}
@@ -270,7 +270,7 @@ private:
 	intModel m_masterVolumeModel;
 	intModel m_masterPitchModel;
 
-	controllerVector m_controllers;
+	ControllerVector m_controllers;
 
 
 	QString m_fileName;
@@ -308,7 +308,8 @@ private:
 
 	friend class engine;
 	friend class songEditor;
-	friend class controllerRackView;
+	friend class mainWindow;
+	friend class ControllerRackView;
 
 signals:
 	void lengthChanged( int _tacts );

include/track.h

@@ -41,7 +41,6 @@
 class QMenu;
 class QPushButton;
 
-class bbTrack;
 class pixmapButton;
 class textFloat;
 class track;

lmms.rc.in

@@ -17,7 +17,7 @@ BEGIN
       VALUE "CompanyName",	"LMMS Developers\0"
       VALUE "FileDescription",	"Linux MultiMedia Studio\0"
       VALUE "FileVersion",	"@VERSION@\0"
-      VALUE "LegalCopyright",	"Copyright (c) 2004-2008 LMMS Developers\0"
+      VALUE "LegalCopyright",	"Copyright (c) 2004-2009 LMMS Developers\0"
       VALUE "OriginalFilename",	"lmms.exe\0"
       VALUE "ProductName",	"LMMS\0"
       VALUE "ProductVersion",	"@VERSION@\0"

plugins/audio_file_processor/audio_file_processor.cpp

@@ -1,7 +1,7 @@
 /*
  * audio_file_processor.cpp - instrument for using audio-files
  *
- * Copyright (c) 2004-2008 Tobias Doerffel <tobydox/at/users.sourceforge.net>
+ * Copyright (c) 2004-2009 Tobias Doerffel <tobydox/at/users.sourceforge.net>
  * 
  * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
  *
@@ -41,8 +41,6 @@
 #include "string_pair_drag.h"
 #include "mmp.h"
 
-
-#undef SINGLE_SOURCE_COMPILE
 #include "embed.cpp"
 
 
@@ -212,13 +210,13 @@ pluginView * audioFileProcessor::instantiateView( QWidget * _parent )
 
 
 void audioFileProcessor::setAudioFile( const QString & _audio_file,
-								bool _rename )
+													bool _rename )
 {
 	// is current channel-name equal to previous-filename??
-	if( _rename && 
+	if( _rename &&
 		( getInstrumentTrack()->name() ==
 			QFileInfo( m_sampleBuffer.audioFile() ).fileName() ||
-		   	m_sampleBuffer.audioFile() == "" ) )
+				m_sampleBuffer.audioFile().isEmpty() ) )
 	{
 		// then set it to new one
 		getInstrumentTrack()->setName( QFileInfo( _audio_file
@@ -424,7 +422,7 @@ void audioFileProcessorView::dropEvent( QDropEvent * _de )
 	}
 	else if( type == QString( "tco_%1" ).arg( track::SampleTrack ) )
 	{
-		multimediaProject mmp( value, FALSE );
+		multimediaProject mmp( value.toUtf8() );
 		castModel<audioFileProcessor>()->setAudioFile( mmp.content().
 				firstChild().toElement().attribute( "src" ) );
 		_de->accept();

plugins/bass_booster/bass_booster.cpp

@@ -1,7 +1,7 @@
 /*
  * bass_booster.cpp - bass-booster-effect-plugin
  *
- * Copyright (c) 2006-2008 Tobias Doerffel <tobydox/at/users.sourceforge.net>
+ * Copyright (c) 2006-2009 Tobias Doerffel <tobydox/at/users.sourceforge.net>
  * 
  * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
  *
@@ -25,8 +25,6 @@
 
 #include "bass_booster.h"
 
-
-#undef SINGLE_SOURCE_COMPILE
 #include "embed.cpp"
 
 

plugins/bit_invader/bit_invader.cpp

@@ -39,7 +39,6 @@
 #include "tooltip.h"
 #include "song.h"
 
-#undef SINGLE_SOURCE_COMPILE
 #include "embed.cpp"
 
 extern "C"

plugins/flp_import/flp_import.cpp

@@ -1,10 +1,10 @@
 /*
  * flp_import.cpp - support for importing FLP-files
  *
- * Copyright (c) 2006-2008 Tobias Doerffel <tobydox/at/users.sourceforge.net>
+ * Copyright (c) 2006-2009 Tobias Doerffel <tobydox/at/users.sourceforge.net>
  *
  * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
- * 
+ *
  * This program is free software; you can redistribute it and/or
  * modify it under the terms of the GNU General Public
  * License as published by the Free Software Foundation; either
@@ -22,7 +22,6 @@
  *
  */
 
-
 #include <QtXml/QDomDocument>
 #include <QtGui/QApplication>
 #include <QtGui/QProgressDialog>
@@ -105,9 +104,9 @@ static void dump_mem( const void * buffer, uint n_bytes )
 	uchar * cp = (uchar *) buffer;
 	for( uint k = 0; k < n_bytes; ++k )
 	{
-		printf( "%02x ", (unsigned int)cp[k] );//( cp[k] > 31 || cp[k] < 7 ) ? cp[k] : '.' );
+		qDebug( "%02x ", (unsigned int)cp[k] );//( cp[k] > 31 || cp[k] < 7 ) ? cp[k] : '.' );
 	}
-	printf( "\n" );
+	qDebug( "\n" );
 }
 
 
@@ -586,6 +585,8 @@ flpImport::~flpImport()
 }
 
 
+
+
 bool flpImport::tryImport( trackContainer * _tc )
 {
 	const int mappedFilter[] =
@@ -676,21 +677,21 @@ bool flpImport::tryImport( trackContainer * _tc )
 
 	if( readID() != makeID( 'F', 'L', 'h', 'd' ) )
 	{
-		printf( "flpImport::tryImport(): not a valid FL project\n" );
+		qWarning( "flpImport::tryImport(): not a valid FL project\n" );
 		return false;
 	}
 
 	const int header_len = read32LE();
 	if( header_len != 6 )
 	{
-		printf( "flpImport::tryImport(): invalid file format\n" );
+		qWarning( "flpImport::tryImport(): invalid file format\n" );
 		return false;
 	}
 
 	const int type = read16LE();
 	if( type != 0 )
 	{
-		printf( "flpImport::tryImport(): type %d format is not "
+		qWarning( "flpImport::tryImport(): type %d format is not "
 							"supported\n", type );
 		return false;
 	}
@@ -698,7 +699,7 @@ bool flpImport::tryImport( trackContainer * _tc )
 	p.numChannels = read16LE();
 	if( p.numChannels < 1 || p.numChannels > 1000 )
 	{
-		printf( "flpImport::tryImport(): invalid number of channels "
+		qWarning( "flpImport::tryImport(): invalid number of channels "
 						"(%d)\n", p.numChannels );
 		return false;
 	}
@@ -706,7 +707,7 @@ bool flpImport::tryImport( trackContainer * _tc )
 	const int ppq = read16LE();
 	if( ppq < 0 )
 	{
-		printf( "flpImport::tryImport(): invalid ppq\n" );
+		qWarning( "flpImport::tryImport(): invalid ppq\n" );
 		return false;
 	}
 
@@ -725,14 +726,14 @@ bool flpImport::tryImport( trackContainer * _tc )
 		const int len = read32LE();
 		if( file().atEnd() )
 		{
-			printf( "flpImport::tryImport(): unexpected "
-					"end of file\n" );
+			qWarning( "flpImport::tryImport(): unexpected "
+						"end of file\n" );
 			return false;
 		}
 		if( len < 0 || len >= 0x10000000 )
 		{
-			printf( "flpImport::tryImport(): invalid "
-					"chunk length %d\n", len );
+			qWarning( "flpImport::tryImport(): invalid "
+						"chunk length %d\n", len );
 			return false;
 		}
 		if( id == makeID( 'F', 'L', 'd', 't' ) )
@@ -753,7 +754,7 @@ bool flpImport::tryImport( trackContainer * _tc )
 		p.channels += FL_Channel();
 	}
 
-	printf( "channels: %d\n", p.numChannels );
+	qDebug( "channels: %d\n", p.numChannels );
 
 
 	char * text = NULL;
@@ -798,8 +799,8 @@ bool flpImport::tryImport( trackContainer * _tc )
 			text = new char[text_len+1];
 			if( readBlock( text, text_len ) <= 0 )
 			{
-				printf( "could not read string (len: %d)\n",
-						text_len );
+				qWarning( "could not read string (len: %d)\n",
+							text_len );
 			}
 
 			text[text_len] = 0;
@@ -815,32 +816,32 @@ bool flpImport::tryImport( trackContainer * _tc )
 		{
 			// BYTE EVENTS
 			case FLP_Byte:
-				printf( "undefined byte %d\n", data );
+				qDebug( "undefined byte %d\n", data );
 				break;
 
 			case FLP_NoteOn:
-				printf( "note on: %d\n", data );
+				qDebug( "note on: %d\n", data );
 				// data = pos   how to handle?
 				break;
 
 			case FLP_Vol:
-				printf( "vol %d\n", data );
+				qDebug( "vol %d\n", data );
 				break;
 
 			case FLP_Pan:
-				printf( "pan %d\n", data );
+				qDebug( "pan %d\n", data );
 				break;
 
 			case FLP_LoopActive:
-				printf( "active loop: %d\n", data );
+				qDebug( "active loop: %d\n", data );
 				break;
 
 			case FLP_ShowInfo:
-				printf( "show info: %d\n", data );
+				qDebug( "show info: %d\n", data );
 				break;
 
 			case FLP_Shuffle:
-				printf( "shuffle: %d\n", data );
+				qDebug( "shuffle: %d\n", data );
 				break;
 
 			case FLP_MainVol:
@@ -848,11 +849,11 @@ bool flpImport::tryImport( trackContainer * _tc )
 				break;
 
 			case FLP_PatLength:
-				printf( "pattern length: %d\n", data );
+				qDebug( "pattern length: %d\n", data );
 				break;
 
 			case FLP_BlockLength:
-				printf( "block length: %d\n", data );
+				qDebug( "block length: %d\n", data );
 				break;
 
 			case FLP_UseLoopPoints:
@@ -860,11 +861,11 @@ bool flpImport::tryImport( trackContainer * _tc )
 				break;
 
 			case FLP_LoopType:
-				printf( "loop type: %d\n", data );
+				qDebug( "loop type: %d\n", data );
 				break;
 
 			case FLP_ChanType:
-				printf( "channel type: %d\n", data );
+				qDebug( "channel type: %d\n", data );
 				if( cc )
 				{
 		switch( data )
@@ -895,7 +896,7 @@ if( p.currentEffectChannel <= NumFxChannels )
 			// WORD EVENTS
 			case FLP_NewChan:
 				cur_channel = data;
-				printf( "new channel: %d\n", data );
+				qDebug( "new channel: %d\n", data );
 				break;
 
 			case FLP_NewPat:
@@ -915,7 +916,7 @@ if( p.currentEffectChannel <= NumFxChannels )
 				break;
 
 			case FLP_FX:
-				printf( "FX: %d\n", data );
+				qDebug( "FX: %d\n", data );
 				break;
 
 			case FLP_Fade_Stereo:
@@ -927,11 +928,11 @@ if( p.currentEffectChannel <= NumFxChannels )
 				{
 					cc->sampleReverseStereo = true;
 				}
-				printf( "fade stereo: %d\n", data );
+				qDebug( "fade stereo: %d\n", data );
 				break;
 
 			case FLP_CutOff:
-				printf( "cutoff (sample): %d\n", data );
+				qDebug( "cutoff (sample): %d\n", data );
 				break;
 
 			case FLP_PreAmp:
@@ -939,11 +940,11 @@ if( p.currentEffectChannel <= NumFxChannels )
 				break;
 
 			case FLP_Decay:
-				printf( "decay (sample): %d\n", data );
+				qDebug( "decay (sample): %d\n", data );
 				break;
 
 			case FLP_Attack:
-				printf( "attack (sample): %d\n", data );
+				qDebug( "attack (sample): %d\n", data );
 				break;
 
 			case FLP_MainPitch:
@@ -951,23 +952,23 @@ if( p.currentEffectChannel <= NumFxChannels )
 				break;
 
 			case FLP_Resonance:
-				printf( "reso (sample): %d\n", data );
+				qDebug( "reso (sample): %d\n", data );
 				break;
 
 			case FLP_LoopBar:
-				printf( "loop bar: %d\n", data );
+				qDebug( "loop bar: %d\n", data );
 				break;
 
 			case FLP_StDel:
-				printf( "stdel (delay?): %d\n", data );
+				qDebug( "stdel (delay?): %d\n", data );
 				break;
 
 			case FLP_FX3:
-				printf( "FX 3: %d\n", data );
+				qDebug( "FX 3: %d\n", data );
 				break;
 
 			case FLP_ShiftDelay:
-				printf( "shift delay: %d\n", data );
+				qDebug( "shift delay: %d\n", data );
 				break;
 
 			case FLP_Dot:
@@ -999,11 +1000,11 @@ if( p.currentEffectChannel <= NumFxChannels )
 			}
 
 			case FLP_FXSine:
-				printf( "fx sine: %d\n", data );
+				qDebug( "fx sine: %d\n", data );
 				break;
 
 			case FLP_CutCutBy:
-				printf( "cut cut by: %d\n", data );
+				qDebug( "cut cut by: %d\n", data );
 				break;
 
 			case FLP_MiddleNote:
@@ -1011,15 +1012,15 @@ if( p.currentEffectChannel <= NumFxChannels )
 				break;
 
 			case FLP_DelayReso:
-				printf( "delay resonance: %d\n", data );
+				qDebug( "delay resonance: %d\n", data );
 				break;
 
 			case FLP_Reverb:
-				printf( "reverb (sample): %d\n", data );
+				qDebug( "reverb (sample): %d\n", data );
 				break;
 
 			case FLP_IntStretch:
-				printf( "int stretch (sample): %d\n", data );
+				qDebug( "int stretch (sample): %d\n", data );
 				break;
 
 			// TEXT EVENTS
@@ -1082,7 +1083,7 @@ if( p.currentEffectChannel <= NumFxChannels )
 
 			case FLP_Text_Version:
 			{
-				printf( "FLP version: %s\n", text );
+				qDebug( "FLP version: %s\n", text );
 				p.versionString = text;
 				QStringList l = p.versionString.split( '.' );
 				p.version = ( l[0].toInt() << 8 ) +
@@ -1103,36 +1104,32 @@ if( p.currentEffectChannel <= NumFxChannels )
 				mappedPluginTypes[QString( text ).toLower()] );
 					if( t > FL_Plugin::EffectPlugin )
 					{
-						printf( "recognized new "
-							"effect %s\n", text );
-						p.effects.push_back(
-							FL_Effect( t ) );
+						qDebug( "recognized new effect %s\n", text );
+						p.effects.push_back( FL_Effect( t ) );
 					}
 					else if( cc )
 					{
-						printf( "recognized new "
-							"plugin %s\n", text );
+						qDebug( "recognized new plugin %s\n", text );
 						cc->pluginType = t;
 					}
 					last_plugin_type = t;
 				}
 				else
 				{
-					printf( "unsupported plugin: %s!\n",
-									text );
+					qDebug( "unsupported plugin: %s!\n", text );
 				}
 				break;
 
 			case FLP_Text_EffectChanName:
-++p.currentEffectChannel;
-if( p.currentEffectChannel <= NumFxChannels )
-{
-	p.effectChannels[p.currentEffectChannel].name = text;
-}
+				++p.currentEffectChannel;
+				if( p.currentEffectChannel <= NumFxChannels )
+				{
+					p.effectChannels[p.currentEffectChannel].name = text;
+				}
 				break;
 
 			case FLP_Text_Delay:
-				printf( "delay data: " );
+				qDebug( "delay data: " );
 				// pi[1] seems to be volume or similiar and
 				// needs to be divided
 				// by p.versionSpecificFactor
@@ -1140,30 +1137,27 @@ if( p.currentEffectChannel <= NumFxChannels )
 				break;
 
 			case FLP_Text_TS404Params:
-				if( cc && cc->pluginType ==
-					FL_Plugin::UnknownPlugin &&
+				if( cc && cc->pluginType == FL_Plugin::UnknownPlugin &&
 						cc->pluginSettings == NULL )
 				{
 					cc->pluginSettings = new char[text_len];
-					memcpy( cc->pluginSettings, text,
-								text_len );
+					memcpy( cc->pluginSettings, text, text_len );
 					cc->pluginSettingsLength = text_len;
-					cc->pluginType =
-						FL_Plugin::TS404;
+					cc->pluginType = FL_Plugin::TS404;
 				}
 				break;
 
 			case FLP_Text_NewPlugin:
 				if( last_plugin_type > FL_Plugin::EffectPlugin )
 				{
-		FL_Effect * e = &p.effects.last();
-		e->fxChannel = puc[0];
-		e->fxPos = puc[4];
-					printf( "new effect: " );
+					FL_Effect * e = &p.effects.last();
+					e->fxChannel = puc[0];
+					e->fxPos = puc[4];
+					qDebug( "new effect: " );
 				}
 				else
 				{
-					printf( "new plugin: " );
+					qDebug( "new plugin: " );
 				}
 				dump_mem( text, text_len );
 				break;
@@ -1176,7 +1170,7 @@ if( p.currentEffectChannel <= NumFxChannels )
 								text_len );
 					cc->pluginSettingsLength = text_len;
 				}
-				printf( "plugin params: " );
+				qDebug( "plugin params: " );
 				dump_mem( text, text_len );
 				break;
 
@@ -1194,7 +1188,7 @@ if( p.currentEffectChannel <= NumFxChannels )
 				cc->arpGate = ( pi[14] * 100.0f ) / 48.0f;
 				cc->arpEnabled = pi[10] > 0;
 
-				printf( "channel params: " );
+				qDebug( "channel params: " );
 				dump_mem( text, text_len );
 				break;
 
@@ -1235,7 +1229,7 @@ if( p.currentEffectChannel <= NumFxChannels )
 //				e.lfoAmount = pi[11] / 128.0f;
 				cc->envelopes.push_back( e );
 
-				printf( "envelope and lfo params:\n" );
+				qDebug( "envelope and lfo params:\n" );
 				dump_mem( text, text_len );
 
 				break;
@@ -1256,7 +1250,7 @@ if( p.currentEffectChannel <= NumFxChannels )
 				cc->filterCut *= 0.5f;
 			}
 				}
-				printf( "basic chan params: " );
+				qDebug( "basic chan params: " );
 				dump_mem( text, text_len );
 				break;
 
@@ -1269,7 +1263,7 @@ if( p.currentEffectChannel <= NumFxChannels )
 				{
 					cc->filterCut *= 0.5;
 				}
-				printf( "old filter params: " );
+				qDebug( "old filter params: " );
 				dump_mem( text, text_len );
 				break;
 
@@ -1277,7 +1271,7 @@ if( p.currentEffectChannel <= NumFxChannels )
 			{
 				const int bpae = 12;
 				const int imax = text_len / bpae;
-				printf( "automation data (%d items)\n", imax );
+				qDebug( "automation data (%d items)\n", imax );
 				for( int i = 0; i < imax; ++i )
 				{
 					FL_Automation a;
@@ -1289,8 +1283,9 @@ if( p.currentEffectChannel <= NumFxChannels )
 					if( a.channel >= 0 &&
 						a.channel < p.numChannels )
 					{
-printf("add channel %d at %d  val %d  control:%d\n", a.channel, a.pos, a.value, a.control );
-				p.channels[a.channel].automationData += a;
+						qDebug( "add channel %d at %d  val %d  control:%d\n",
+									a.channel, a.pos, a.value, a.control );
+						p.channels[a.channel].automationData += a;
 					}
 //					dump_mem( text+i*bpae, bpae );
 				}
@@ -1315,15 +1310,22 @@ printf("add channel %d at %d  val %d  control:%d\n", a.channel, a.pos, a.value,
 					len /= (4*ppq) / DefaultTicksPerTact;
 					note n( len, pos, key, vol * 100 / 128,
 							pan*200 / 128 - 100 );
+					if( ch < p.numChannels )
+					{
 	p.channels[ch].notes.push_back( qMakePair( p.currentPattern, n ) );
-					printf( "note: " );
+					}
+					else
+					{
+						qDebug( "invalid " );
+					}
+					qDebug( "note: " );
 					dump_mem( text+i*bpn, bpn );
 				}
 				break;
 			}
 
 			case FLP_Text_ChanGroupName:
-				printf( "channel group name: %s\n", text );
+				qDebug( "channel group name: %s\n", text );
 				break;
 
 			// case 216: pi[2] /= p.versionSpecificFactor
@@ -1354,7 +1356,7 @@ p.effectChannels[ch].volume = ( val / p.versionSpecificFactor ) * 100 / 128;
 					}
 					else
 					{
-printf("FX-ch: %d  param: %x  value:%x\n", ch, param, val );
+qDebug( "FX-ch: %d  param: %x  value:%x\n", ch, param, val );
 					}
 				}
 				break;
@@ -1379,7 +1381,7 @@ if( pat > 2146 && pat <= 2278 )	// whatever these magic numbers are for...
 }
 else
 {
-	printf( "playlist item: " );
+	qDebug( "unknown playlist item: " );
 	dump_mem( text+i*bpi, bpi );
 }
 				}
@@ -1389,25 +1391,18 @@ else
 			default:
 				if( ev >= FLP_Text )
 				{
-					printf( "!! unhandled text (ev: %d, "
-							"len: %d): ",
-							ev, text_len );
+					qDebug( "!! unhandled text (ev: %d, len: %d): ",
+								ev, text_len );
 					dump_mem( text, text_len );
 				}
 				else
 				{
-					printf( "!! handling of FLP-event %d "
-						"not implemented yet "
-						"(data=%d).\n", ev, data );
+					qDebug( "!! handling of FLP-event %d not implemented yet "
+							"(data=%d).\n", ev, data );
 				}
 				break;
 		}
-/*		if( ev >= FLP_Text )
-		{
-			printf( "dump (ev: %d, len: %d): ", ev, text_len );
-			dump_mem( text, text_len );
-		}*/
-        }
+	}
 
 
 	// now create a project from FL_Project data structure
@@ -1424,7 +1419,7 @@ else
 
 
 	progressDialog.setMaximum( p.maxPatterns + p.channels.size() +
-							p.effects.size() );
+								p.effects.size() );
 	int cur_progress = 0;
 
 	// create BB tracks
@@ -1480,8 +1475,7 @@ else
 			case FL_Plugin::UnknownPlugin:
 			default:
 				it->instrumentPlugin =
-					t->loadInstrument(
-							"audiofileprocessor" );
+					t->loadInstrument( "audiofileprocessor" );
 				break;
 		}
 		processPluginParams( &( *it ) );
@@ -1504,8 +1498,7 @@ else
 					iss->m_filterResModel.minValue() );
 		iss->m_filterEnabledModel.setValue( it->filterEnabled );
 
-		for( QList<FL_Channel_Envelope>::iterator jt =
-							it->envelopes.begin();
+		for( QList<FL_Channel_Envelope>::iterator jt = it->envelopes.begin();
 					jt != it->envelopes.end(); ++jt )
 		{
 			if( jt->target != instrumentSoundShaping::NumTargets )
@@ -1554,8 +1547,7 @@ else
 		}
 
 		// process all notes
-		for( FL_Channel::noteVector::const_iterator jt =
-							it->notes.begin();
+		for( FL_Channel::noteVector::const_iterator jt = it->notes.begin();
 						jt != it->notes.end(); ++jt )
 		{
 			const int pat = jt->first;
@@ -1564,8 +1556,7 @@ else
 			{
 				continue;
 			}
-			pattern * p = dynamic_cast<pattern *>(
-							t->getTCO( pat ) );
+			pattern * p = dynamic_cast<pattern *>( t->getTCO( pat ) );
 			if( p != NULL )
 			{
 				p->addNote( jt->second, false );
@@ -1584,8 +1575,7 @@ else
 			{
 				case FL_Automation::ControlVolume:
 					m = t->volumeModel();
-					value *= ( 100.0f / 128.0f ) /
-							p.versionSpecificFactor;
+					value *= ( 100.0f / 128.0f ) / p.versionSpecificFactor;
 					break;
 				case FL_Automation::ControlPanning:
 					m = t->panningModel();
@@ -1613,9 +1603,9 @@ else
 					value = mappedFilter[jt->value];
 					break;
 				default:
-					printf( "handling automation data of "
-						"control %d not implemented "
-						"yet\n", jt->control );
+					qDebug( "handling automation data of "
+							"control %d not implemented "
+							"yet\n", jt->control );
 					break;
 			}
 			if( m )
@@ -1653,14 +1643,12 @@ p->putValue( jt->pos, value, false );
 		else
 		{
 			effKeys << effectKey( &( *it ), it->name );
-
 		}
 	}
 
 	for( int fx_ch = 0; fx_ch <= NumFxChannels ; ++fx_ch )
 	{
-		fxChannel * ch = engine::getFxMixer()->getEffectChannel(
-									fx_ch );
+		fxChannel * ch = engine::getFxMixer()->getEffectChannel( fx_ch );
 		if( !ch )
 		{
 			continue;
@@ -1723,9 +1711,8 @@ p->putValue( jt->pos, value, false );
 			continue;
 		}
 		effectChain * ec = &engine::getFxMixer()->
-					getEffectChannel( it->fxChannel )->
-								m_fxChain;
-		printf("adding %s to %d\n", effName.toAscii().constData(),
+					getEffectChannel( it->fxChannel )->m_fxChain;
+		qDebug( "adding %s to %d\n", effName.toUtf8().constData(),
 								it->fxChannel );
 		for( effectKeyList::iterator jt = effKeys.begin();
 						jt != effKeys.end(); ++jt )
@@ -1735,7 +1722,7 @@ p->putValue( jt->pos, value, false );
 				( jt->desc->sub_plugin_features != NULL &&
 					jt->name.contains( effName ) ) )
 			{
-				printf("instantiate %s\n", jt->desc->name );
+				qDebug( "instantiate %s\n", jt->desc->name );
 				effect * e = effect::instantiate(
 						jt->desc->name, ec, &( *jt ) );
 				ec->appendEffect( e );
@@ -1788,7 +1775,7 @@ p->putValue( jt->pos, value, false );
 
 void flpImport::processPluginParams( FL_Channel * _ch )
 {
-	printf( "plugin params for plugin %d (%d bytes): ", _ch->pluginType,
+	qDebug( "plugin params for plugin %d (%d bytes): ", _ch->pluginType,
 						_ch->pluginSettingsLength );
 	dump_mem( _ch->pluginSettings, _ch->pluginSettingsLength );
 	switch( _ch->pluginType )
@@ -1877,7 +1864,7 @@ void flpImport::processPluginParams( FL_Channel * _ch )
 
 		case FL_Plugin::UnknownPlugin:
 		default:
-			printf( "handling of plugin params not implemented "
+			qDebug( "handling of plugin params not implemented "
 					"for current plugin\n" );
 			break;
 	}

plugins/flp_import/unrtf/attr.c

@@ -108,10 +108,10 @@ attr_express_begin (int attr, const char* param) {
 	switch(attr) 
 	{
 	case ATTR_BOLD: 
-		outstring+=QString().sprintf(op->bold_begin); 
+		outstring+=QString().sprintf("%s", op->bold_begin); 
 		break;
 	case ATTR_ITALIC: 
-		outstring+=QString().sprintf(op->italic_begin); 
+		outstring+=QString().sprintf("%s", op->italic_begin); 
 		break;
 
 	/* Various underlines, they all resolve to HTML's <u> */
@@ -123,11 +123,11 @@ attr_express_begin (int attr, const char* param) {
 	case ATTR_2DOT_DASH_UL:
 	case ATTR_WORD_UL: 
 	case ATTR_UNDERLINE: 
-		outstring+=QString().sprintf(op->underline_begin); 
+		outstring+=QString().sprintf("%s", op->underline_begin); 
 		break;
 
 	case ATTR_DOUBLE_UL: 
-		outstring+=QString().sprintf(op->dbl_underline_begin); 
+		outstring+=QString().sprintf("%s", op->dbl_underline_begin); 
 		break;
 
 	case ATTR_FONTSIZE: 
@@ -148,18 +148,18 @@ attr_express_begin (int attr, const char* param) {
 		break;
 
 	case ATTR_SUPER: 
-		outstring+=QString().sprintf(op->superscript_begin); 
+		outstring+=QString().sprintf("%s", op->superscript_begin); 
 		break;
 	case ATTR_SUB: 
-		outstring+=QString().sprintf(op->subscript_begin); 
+		outstring+=QString().sprintf("%s", op->subscript_begin); 
 		break;
 
 	case ATTR_STRIKE: 
-		outstring+=QString().sprintf(op->strikethru_begin); 
+		outstring+=QString().sprintf("%s", op->strikethru_begin); 
 		break;
 
 	case ATTR_DBL_STRIKE: 
-		outstring+=QString().sprintf(op->dbl_strikethru_begin); 
+		outstring+=QString().sprintf("%s", op->dbl_strikethru_begin); 
 		break;
 
 	case ATTR_EXPAND: 
@@ -167,16 +167,16 @@ attr_express_begin (int attr, const char* param) {
 		break;
 
 	case ATTR_OUTLINE: 
-		outstring+=QString().sprintf(op->outline_begin); 
+		outstring+=QString().sprintf("%s", op->outline_begin); 
 		break;
 	case ATTR_SHADOW: 
-		outstring+=QString().sprintf(op->shadow_begin); 
+		outstring+=QString().sprintf("%s", op->shadow_begin); 
 		break;
 	case ATTR_EMBOSS: 
-		outstring+=QString().sprintf(op->emboss_begin); 
+		outstring+=QString().sprintf("%s", op->emboss_begin); 
 		break;
 	case ATTR_ENGRAVE: 
-		outstring+=QString().sprintf(op->engrave_begin); 
+		outstring+=QString().sprintf("%s", op->engrave_begin); 
 		break;
 
 	case ATTR_CAPS:
@@ -189,7 +189,7 @@ attr_express_begin (int attr, const char* param) {
 			simulate_smallcaps = TRUE;
 		else {
 			if (op->small_caps_begin)
-				outstring+=QString().sprintf(op->small_caps_begin); 
+				outstring+=QString().sprintf("%s", op->small_caps_begin); 
 		}
 		break;
 	}
@@ -209,10 +209,10 @@ attr_express_end (int attr, char *param)
 	switch(attr) 
 	{
 	case ATTR_BOLD: 
-		outstring+=QString().sprintf(op->bold_end); 
+		outstring+=QString().sprintf("%s", op->bold_end); 
 		break;
 	case ATTR_ITALIC: 
-		outstring+=QString().sprintf(op->italic_end); 
+		outstring+=QString().sprintf("%s", op->italic_end); 
 		break;
 
 	/* Various underlines, they all resolve to HTML's </u> */
@@ -224,11 +224,11 @@ attr_express_end (int attr, char *param)
 	case ATTR_2DOT_DASH_UL: 
 	case ATTR_WORD_UL: 
 	case ATTR_UNDERLINE: 
-		outstring+=QString().sprintf(op->underline_end); 
+		outstring+=QString().sprintf("%s", op->underline_end); 
 		break;
 
 	case ATTR_DOUBLE_UL: 
-		outstring+=QString().sprintf(op->dbl_underline_end); 
+		outstring+=QString().sprintf("%s", op->dbl_underline_end); 
 		break;
 
 	case ATTR_FONTSIZE: 
@@ -236,47 +236,47 @@ attr_express_end (int attr, char *param)
 		break;
 
 	case ATTR_FONTFACE: 
-		outstring+=QString().sprintf(op->font_end); 
+		outstring+=QString().sprintf("%s", op->font_end); 
 		break;
 
 	case ATTR_FOREGROUND: 
-		outstring+=QString().sprintf(op->foreground_end); 
+		outstring+=QString().sprintf("%s", op->foreground_end); 
 		break;
 	case ATTR_BACKGROUND: 
 		if (!simple_mode)
-		  outstring+=QString().sprintf(op->background_end);
+		  outstring+=QString().sprintf("%s", op->background_end);
 		break;
 
 	case ATTR_SUPER: 
-		outstring+=QString().sprintf(op->superscript_end); 
+		outstring+=QString().sprintf("%s", op->superscript_end); 
 		break;
 	case ATTR_SUB: 
-		outstring+=QString().sprintf(op->subscript_end); 
+		outstring+=QString().sprintf("%s", op->subscript_end); 
 		break;
 
 	case ATTR_STRIKE: 
-		outstring+=QString().sprintf(op->strikethru_end); 
+		outstring+=QString().sprintf("%s", op->strikethru_end); 
 		break;
 
 	case ATTR_DBL_STRIKE: 
-		outstring+=QString().sprintf(op->dbl_strikethru_end); 
+		outstring+=QString().sprintf("%s", op->dbl_strikethru_end); 
 		break;
 
 	case ATTR_OUTLINE: 
-		outstring+=QString().sprintf(op->outline_end); 
+		outstring+=QString().sprintf("%s", op->outline_end); 
 		break;
 	case ATTR_SHADOW: 
-		outstring+=QString().sprintf(op->shadow_end); 
+		outstring+=QString().sprintf("%s", op->shadow_end); 
 		break;
 	case ATTR_EMBOSS: 
-		outstring+=QString().sprintf(op->emboss_end); 
+		outstring+=QString().sprintf("%s", op->emboss_end); 
 		break;
 	case ATTR_ENGRAVE: 
-		outstring+=QString().sprintf(op->engrave_end); 
+		outstring+=QString().sprintf("%s", op->engrave_end); 
 		break;
 
 	case ATTR_EXPAND: 
-		outstring+=QString().sprintf(op->expand_end); 
+		outstring+=QString().sprintf("%s", op->expand_end); 
 		break;
 
 	case ATTR_CAPS:
@@ -289,7 +289,7 @@ attr_express_end (int attr, char *param)
 			simulate_smallcaps = FALSE;
 		else {
 			if (op->small_caps_end)
-				outstring+=QString().sprintf(op->small_caps_end); 
+				outstring+=QString().sprintf("%s", op->small_caps_end); 
 		}
 		break;
 	}

plugins/flp_import/unrtf/convert.c

@@ -650,8 +650,8 @@ starting_body ()
 {
 	if (!have_printed_body) {
 		if (!inline_mode) {
-			outstring+=QString().sprintf(op->header_end);
-			outstring+=QString().sprintf(op->body_begin);
+			outstring+=QString().sprintf("%s", op->header_end);
+			outstring+=QString().sprintf("%s", op->body_begin);
 		}
 		within_header = FALSE;
 		have_printed_body = TRUE;
@@ -926,7 +926,7 @@ process_info_group (Word *w)
 
 			if (!inline_mode) {
 				if (!strcmp("\\title", s)) {
-					outstring+=QString().sprintf(op->document_title_begin);
+					outstring+=QString().sprintf("%s", op->document_title_begin);
 					w2=child->next;
 					while (w2) {
 						char *s2 = word_string(w2);
@@ -951,18 +951,18 @@ process_info_group (Word *w)
 								s3 = op_translate_char (op, charset_type, charset_codepage, ch, numchar_table);
 								if (!s3 || !*s3)
 								{
-									outstring+=QString().sprintf(op->comment_begin);
+									outstring+=QString().sprintf("%s", op->comment_begin);
 									outstring+=QString().sprintf("char 0x%02x",ch);
-									outstring+=QString().sprintf(op->comment_end);
+									outstring+=QString().sprintf("%s", op->comment_end);
 
 								}
 								else
 								{
 									if (op->word_begin)
-										outstring+=QString().sprintf(op->word_begin);
+										outstring+=QString().sprintf("%s", op->word_begin);
 									outstring+=QString().sprintf("%s", s3);
 									if (op->word_end)
-										outstring+=QString().sprintf(op->word_end);
+										outstring+=QString().sprintf("%s", op->word_end);
 								}
 							}
 						}
@@ -970,10 +970,10 @@ process_info_group (Word *w)
 #endif
 						w2 = w2->next;
 					}
-					outstring+=QString().sprintf(op->document_title_end);
+					outstring+=QString().sprintf("%s", op->document_title_end);
 				}
 				else if (!strcmp("\\keywords", s)) {
-					outstring+=QString().sprintf(op->document_keywords_begin);
+					outstring+=QString().sprintf("%s", op->document_keywords_begin);
 					w2=child->next;
 					while (w2) {
 						char *s2 = word_string(w2);
@@ -981,10 +981,10 @@ process_info_group (Word *w)
 							outstring+=QString().sprintf("%s,", s2);
 						w2 = w2->next;
 					}
-					outstring+=QString().sprintf(op->document_keywords_end);
+					outstring+=QString().sprintf("%s", op->document_keywords_end);
 				}
 				else if (!strcmp("\\author", s)) {
-					outstring+=QString().sprintf(op->document_author_begin);
+					outstring+=QString().sprintf("%s", op->document_author_begin);
 					w2=child->next;
 					while (w2) {
 						char *s2 = word_string(w2);
@@ -992,7 +992,7 @@ process_info_group (Word *w)
 							outstring+=QString().sprintf("%s", s2);
 						w2 = w2->next;
 					}
-					outstring+=QString().sprintf(op->document_author_end);
+					outstring+=QString().sprintf("%s", op->document_author_end);
 				}
 				else if (!strcmp("\\comment", s)) {
 					outstring+=QString().sprintf("%s",op->comment_begin);
@@ -1312,9 +1312,9 @@ cmd_field (Word *w, int align, char has_param, int num) {
 							    w4=w4->next;
 						    if (w4) {
 							    s4=word_string(w4);
-							    outstring+=QString().sprintf(op->hyperlink_begin);
+							    outstring+=QString().sprintf("%s", op->hyperlink_begin);
 							    outstring+=QString().sprintf("%s", s4);
-							    outstring+=QString().sprintf(op->hyperlink_end);
+							    outstring+=QString().sprintf("%s", op->hyperlink_end);
 							    return TRUE;
 						    }
 					    	
@@ -1423,7 +1423,7 @@ cmd_tab (Word *w, int align, char has_param, int param)
 	int need= 8-(total_chars_this_line%8);
 	total_chars_this_line += need;
 	while(need>0) {
-		outstring+=QString().sprintf(op->forced_space);
+		outstring+=QString().sprintf("%s", op->forced_space);
 		need--;
 	}
 	outstring+=QString().sprintf("\n");
@@ -1643,7 +1643,7 @@ cmd_scaps (Word *w, int align, char has_param, int param) {
 static int 
 cmd_bullet (Word *w, int align, char has_param, int param) {
 	if (op->chars.bullet) {
-		outstring+=QString().sprintf(op->chars.bullet);
+		outstring+=QString().sprintf("%s", op->chars.bullet);
 		++total_chars_this_line; /* \tab */
 	}
 	return FALSE;
@@ -1658,7 +1658,7 @@ cmd_bullet (Word *w, int align, char has_param, int param) {
 static int 
 cmd_ldblquote (Word *w, int align, char has_param, int param) {
 	if (op->chars.left_dbl_quote) {
-		outstring+=QString().sprintf(op->chars.left_dbl_quote);
+		outstring+=QString().sprintf("%s", op->chars.left_dbl_quote);
 		++total_chars_this_line; /* \tab */
 	}
 	return FALSE;
@@ -1675,7 +1675,7 @@ cmd_ldblquote (Word *w, int align, char has_param, int param) {
 static int 
 cmd_rdblquote (Word *w, int align, char has_param, int param) {
 	if (op->chars.right_dbl_quote) {
-		outstring+=QString().sprintf(op->chars.right_dbl_quote);
+		outstring+=QString().sprintf("%s", op->chars.right_dbl_quote);
 		++total_chars_this_line; /* \tab */
 	}
 	return FALSE;
@@ -1691,7 +1691,7 @@ cmd_rdblquote (Word *w, int align, char has_param, int param) {
 static int 
 cmd_lquote (Word *w, int align, char has_param, int param) {
 	if (op->chars.left_quote) {
-		outstring+=QString().sprintf(op->chars.left_quote);
+		outstring+=QString().sprintf("%s", op->chars.left_quote);
 		++total_chars_this_line; /* \tab */
 	}
 	return FALSE;
@@ -1708,7 +1708,7 @@ cmd_lquote (Word *w, int align, char has_param, int param) {
 static int 
 cmd_nonbreaking_space (Word *w, int align, char has_param, int param) {
 	if (op->chars.nonbreaking_space) {
-		outstring+=QString().sprintf(op->chars.nonbreaking_space);
+		outstring+=QString().sprintf("%s", op->chars.nonbreaking_space);
 		++total_chars_this_line; /* \tab */
 	}
 	return FALSE;
@@ -1725,7 +1725,7 @@ cmd_nonbreaking_space (Word *w, int align, char has_param, int param) {
 static int 
 cmd_nonbreaking_hyphen (Word *w, int align, char has_param, int param) {
 	if (op->chars.nonbreaking_hyphen) {
-		outstring+=QString().sprintf(op->chars.nonbreaking_hyphen);
+		outstring+=QString().sprintf("%s", op->chars.nonbreaking_hyphen);
 		++total_chars_this_line; /* \tab */
 	}
 	return FALSE;
@@ -1742,7 +1742,7 @@ cmd_nonbreaking_hyphen (Word *w, int align, char has_param, int param) {
 static int 
 cmd_optional_hyphen (Word *w, int align, char has_param, int param) {
 	if (op->chars.optional_hyphen) {
-		outstring+=QString().sprintf(op->chars.optional_hyphen);
+		outstring+=QString().sprintf("%s", op->chars.optional_hyphen);
 		++total_chars_this_line; /* \tab */
 	}
 	return FALSE;
@@ -1758,7 +1758,7 @@ cmd_optional_hyphen (Word *w, int align, char has_param, int param) {
 static int 
 cmd_emdash (Word *w, int align, char has_param, int param) {
 	if (op->chars.emdash) {
-		outstring+=QString().sprintf(op->chars.emdash);
+		outstring+=QString().sprintf("%s", op->chars.emdash);
 		++total_chars_this_line; /* \tab */
 	}
 	return FALSE;
@@ -1775,7 +1775,7 @@ cmd_emdash (Word *w, int align, char has_param, int param) {
 static int 
 cmd_endash (Word *w, int align, char has_param, int param) {
 	if (op->chars.endash) {
-		outstring+=QString().sprintf(op->chars.endash);
+		outstring+=QString().sprintf("%s", op->chars.endash);
 		++total_chars_this_line; /* \tab */
 	}
 	return FALSE;
@@ -1792,7 +1792,7 @@ cmd_endash (Word *w, int align, char has_param, int param) {
 static int 
 cmd_rquote (Word *w, int align, char has_param, int param) {
 	if (op->chars.right_quote) {
-		outstring+=QString().sprintf(op->chars.right_quote);
+		outstring+=QString().sprintf("%s", op->chars.right_quote);
 		++total_chars_this_line; /* \tab */
 	}
 	return FALSE;
@@ -1808,7 +1808,7 @@ cmd_rquote (Word *w, int align, char has_param, int param) {
 static int 
 cmd_par (Word *w, int align, char has_param, int param) {
 	if (op->line_break) {
-		outstring+=QString().sprintf(op->line_break);
+		outstring+=QString().sprintf("%s", op->line_break);
 		total_chars_this_line = 0; /* \tab */
 	}
 	return FALSE;
@@ -1825,7 +1825,7 @@ cmd_par (Word *w, int align, char has_param, int param) {
 static int 
 cmd_line (Word *w, int align, char has_param, int param) {
 	if (op->line_break) {
-		outstring+=QString().sprintf(op->line_break);
+		outstring+=QString().sprintf("%s", op->line_break);
 		total_chars_this_line = 0; /* \tab */
 	}
 	return FALSE;
@@ -1841,7 +1841,7 @@ cmd_line (Word *w, int align, char has_param, int param) {
 
 static int cmd_page (Word *w, int align, char has_param, int param) {
 	if (op->page_break) {
-		outstring+=QString().sprintf(op->page_break);
+		outstring+=QString().sprintf("%s", op->page_break);
 		total_chars_this_line = 0; /* \tab */
 	}
 	return FALSE;
@@ -2337,7 +2337,7 @@ static int cmd_s (Word *w, int align, char has_param, int param) {
 static int cmd_sect (Word *w, int align, char has_param, int param) {
 	/* XX kludge */
 	if (op->paragraph_begin) {
-		outstring+=QString().sprintf(op->paragraph_begin);
+		outstring+=QString().sprintf("%s", op->paragraph_begin);
 	}
 	return FALSE;
 }
@@ -2351,9 +2351,9 @@ static int cmd_sect (Word *w, int align, char has_param, int param) {
 
 static int cmd_shp (Word *w, int align, char has_param, int param) {
 	if (op->comment_begin) {
-		outstring+=QString().sprintf(op->comment_begin);
+		outstring+=QString().sprintf("%s", op->comment_begin);
 		outstring+=QString().sprintf("Drawn Shape (ignored--not implemented yet)");
-		outstring+=QString().sprintf(op->comment_end);	/* daved 0.20.2 */
+		outstring+=QString().sprintf("%s", op->comment_end);	/* daved 0.20.2 */
 	}
 
 	return FALSE;
@@ -2404,17 +2404,17 @@ static int cmd_ansicpg (Word *w, int align, char has_param, int param) {
 		charset_codepage = &codepages[i];
 		if (charset_codepage->cp == param) {
 			if (op->comment_begin) {
-				outstring+=QString().sprintf(op->comment_begin);
+				outstring+=QString().sprintf("%s", op->comment_begin);
 				outstring+=QString().sprintf("document uses ANSI codepage %d character set", param);
-				outstring+=QString().sprintf(op->comment_end);
+				outstring+=QString().sprintf("%s", op->comment_end);
 			}
 			break;
 		}
 	}
 	if ((charset_codepage == NULL || charset_codepage->cp == 0) && op->comment_begin) {
-		outstring+=QString().sprintf(op->comment_begin);
+		outstring+=QString().sprintf("%s", op->comment_begin);
 		outstring+=QString().sprintf("document uses default ANSI codepage character set");
-		outstring+=QString().sprintf(op->comment_end);
+		outstring+=QString().sprintf("%s", op->comment_end);
 	}
 	return FALSE;
 }
@@ -3106,7 +3106,7 @@ enum { SMALL=0, BIG=1 };
 	if (simulate_smallcaps) {
 		if (*s >= 'a' && *s <= 'z') {
 			state=SMALL;
-			outstring+=QString().sprintf(op->smaller_begin);
+			outstring+=QString().sprintf("%s", op->smaller_begin);
 		}
 		else
 			state=BIG;
@@ -3136,13 +3136,13 @@ enum { SMALL=0, BIG=1 };
 			ch = *s;
 			if (ch >= 'a' && ch <= 'z') {
 				if (state==BIG)
-					outstring+=QString().sprintf(op->smaller_begin);
+					outstring+=QString().sprintf("%s", op->smaller_begin);
 				state=SMALL;
 			}
 			else
 			{
 				if (state==SMALL)
-					outstring+=QString().sprintf(op->smaller_end);
+					outstring+=QString().sprintf("%s", op->smaller_end);
 				state=BIG;
 			}
 		}
@@ -3168,7 +3168,7 @@ begin_table()
 	have_printed_cell_end = FALSE;
 	attrstack_push();
 	starting_body();
-	outstring+=QString().sprintf(op->table_begin);
+	outstring+=QString().sprintf("%s", op->table_begin);
 }
 
 
@@ -3186,12 +3186,12 @@ end_table ()
 	if (within_table) {
 		if (!have_printed_cell_end) {
 			attr_pop_dump();
-			outstring+=QString().sprintf(op->table_cell_end);
+			outstring+=QString().sprintf("%s", op->table_cell_end);
 		}
 		if (!have_printed_row_end) {
-			outstring+=QString().sprintf(op->table_row_end);
+			outstring+=QString().sprintf("%s", op->table_row_end);
 		}
-		outstring+=QString().sprintf(op->table_end);
+		outstring+=QString().sprintf("%s", op->table_end);
 		within_table=FALSE;
 		have_printed_row_begin = FALSE;
 		have_printed_cell_begin = FALSE;
@@ -3213,13 +3213,13 @@ void
 starting_text() {
 	if (within_table) {
 		if (!have_printed_row_begin) {
-			outstring+=QString().sprintf(op->table_row_begin);
+			outstring+=QString().sprintf("%s", op->table_row_begin);
 			have_printed_row_begin=TRUE;
 			have_printed_row_end=FALSE;
 			have_printed_cell_begin=FALSE;
 		}
 		if (!have_printed_cell_begin) {
-			outstring+=QString().sprintf(op->table_cell_begin);
+			outstring+=QString().sprintf("%s", op->table_cell_begin);
 			attrstack_express_all();
 			have_printed_cell_begin=TRUE;
 			have_printed_cell_end=FALSE;
@@ -3246,15 +3246,15 @@ starting_paragraph_align (int align)
 	switch (align) 
 	{
 	case ALIGN_CENTER:
-		outstring+=QString().sprintf(op->center_begin); 
+		outstring+=QString().sprintf("%s", op->center_begin); 
 		break;
 	case ALIGN_LEFT:
 		break;
 	case ALIGN_RIGHT:
-		outstring+=QString().sprintf(op->align_right_begin);
+		outstring+=QString().sprintf("%s", op->align_right_begin);
 		break;
 	case ALIGN_JUSTIFY:
-		outstring+=QString().sprintf(op->align_right_begin); /* This is WRONG! */
+		outstring+=QString().sprintf("%s", op->align_right_begin); /* This is WRONG! */
 		break;
 	}
 }
@@ -3273,16 +3273,16 @@ ending_paragraph_align (int align)
 {
 	switch (align) {
 	case ALIGN_CENTER:
-		outstring+=QString().sprintf(op->center_end);
+		outstring+=QString().sprintf("%s", op->center_end);
 		break;
 	case ALIGN_LEFT:
-	  /* outstring+=QString().sprintf(op->align_left_end); */
+	  /* outstring+=QString().sprintf("%s", op->align_left_end); */
 		break;
 	case ALIGN_RIGHT:
-		outstring+=QString().sprintf(op->align_right_end);
+		outstring+=QString().sprintf("%s", op->align_right_end);
 		break;
 	case ALIGN_JUSTIFY:
-		outstring+=QString().sprintf(op->justify_end);
+		outstring+=QString().sprintf("%s", op->justify_end);
 		break;
 	}
 }
@@ -3391,12 +3391,12 @@ word_print_core (Word *w)
 					total_chars_this_line += strlen(s);
 
 					if (op->word_begin)
-						outstring+=QString().sprintf(op->word_begin);
+						outstring+=QString().sprintf("%s", op->word_begin);
 
 					print_with_special_exprs (s);
 
 					if (op->word_end)
-						outstring+=QString().sprintf(op->word_end);
+						outstring+=QString().sprintf("%s", op->word_end);
 				}
 
 
@@ -3456,17 +3456,17 @@ word_print_core (Word *w)
 					is_cell_group=TRUE;
 					if (!have_printed_cell_begin) {
 						/* Need this with empty cells */
-						outstring+=QString().sprintf(op->table_cell_begin);
+						outstring+=QString().sprintf("%s", op->table_cell_begin);
 						attrstack_express_all();
 					}
 					attr_pop_dump();
-					outstring+=QString().sprintf(op->table_cell_end);
+					outstring+=QString().sprintf("%s", op->table_cell_end);
 					have_printed_cell_begin = FALSE;
 					have_printed_cell_end=TRUE;
 				}
 				else if (!strcmp (s, "row")) {
 					if (within_table) {
-						outstring+=QString().sprintf(op->table_row_end);
+						outstring+=QString().sprintf("%s", op->table_row_end);
 						have_printed_row_begin = FALSE;
 						have_printed_row_end=TRUE;
 					} else {
@@ -3496,10 +3496,10 @@ word_print_core (Word *w)
 						outstring+=QString().sprintf("%s",op->comment_end);
 					} else {
 						if (op->word_begin)
-							outstring+=QString().sprintf(op->word_begin);
+							outstring+=QString().sprintf("%s", op->word_begin);
 						outstring+=QString().sprintf("%s", s2);
 						if (op->word_end)
-							outstring+=QString().sprintf(op->word_end);
+							outstring+=QString().sprintf("%s", op->word_end);
 					}
 				}
 				else
@@ -3529,9 +3529,9 @@ word_print_core (Word *w)
 
 					if (!hip) {
 						if (debug_mode) {
-							outstring+=QString().sprintf(op->comment_begin);
+							outstring+=QString().sprintf("%s", op->comment_begin);
 							outstring+=QString().sprintf("Unfamiliar RTF command: %s (HashIndex not found)", s);
-							outstring+=QString().sprintf(op->comment_end); /* daved 0.20.2 */
+							outstring+=QString().sprintf("%s", op->comment_end); /* daved 0.20.2 */
 						}
 					}
 					else {
@@ -3610,9 +3610,9 @@ word_print_core (Word *w)
 	if (within_picture) {
 		if(pictfile) { 
 			fclose(pictfile);
-			outstring+=QString().sprintf(op->imagelink_begin);
+			outstring+=QString().sprintf("%s", op->imagelink_begin);
 			outstring+=QString().sprintf("%s", picture_path);
-			outstring+=QString().sprintf(op->imagelink_end);
+			outstring+=QString().sprintf("%s", op->imagelink_end);
 		}
 		within_picture=FALSE;
 	}
@@ -3665,8 +3665,8 @@ word_print (Word *w, QString & _s)
 
 	outstring = "";
 	if (!inline_mode) {
-		outstring+=QString().sprintf(op->document_begin);
-		outstring+=QString().sprintf(op->header_begin);
+		outstring+=QString().sprintf("%s", op->document_begin);
+		outstring+=QString().sprintf("%s", op->header_begin);
 	}
 
 	within_header=TRUE;
@@ -3677,8 +3677,8 @@ word_print (Word *w, QString & _s)
 	end_table();
 
 	if (!inline_mode) {
-		outstring+=QString().sprintf(op->body_end);
-		outstring+=QString().sprintf(op->document_end);
+		outstring+=QString().sprintf("%s", op->body_end);
+		outstring+=QString().sprintf("%s", op->document_end);
 	}
 	_s = outstring;
 }

plugins/flp_import/unrtf/output.c

@@ -238,49 +238,49 @@ op_begin_std_fontsize (OutputPersonality *op, int size)
 	switch (size) {
 	case 8:
 		if (op->fontsize8_begin) {
-			outstring+=QString().sprintf(op->fontsize8_begin);
+			outstring+=QString().sprintf("%s", op->fontsize8_begin);
 			found_std_expr = TRUE;
 		}
 		break;
 	case 10:
 		if (op->fontsize10_begin) {
-			outstring+=QString().sprintf(op->fontsize10_begin);
+			outstring+=QString().sprintf("%s", op->fontsize10_begin);
 			found_std_expr = TRUE;
 		}
 		break;
 	case 12:
 		if (op->fontsize12_begin) {
-			outstring+=QString().sprintf(op->fontsize12_begin);
+			outstring+=QString().sprintf("%s", op->fontsize12_begin);
 			found_std_expr = TRUE;
 		}
 		break;
 	case 14:
 		if (op->fontsize14_begin) {
-			outstring+=QString().sprintf(op->fontsize14_begin);
+			outstring+=QString().sprintf("%s", op->fontsize14_begin);
 			found_std_expr = TRUE;
 		}
 		break;
 	case 18:
 		if (op->fontsize18_begin) {
-			outstring+=QString().sprintf(op->fontsize18_begin);
+			outstring+=QString().sprintf("%s", op->fontsize18_begin);
 			found_std_expr = TRUE;
 		}
 		break;
 	case 24:
 		if (op->fontsize24_begin) {
-			outstring+=QString().sprintf(op->fontsize24_begin);
+			outstring+=QString().sprintf("%s", op->fontsize24_begin);
 			found_std_expr = TRUE;
 		}
 		break;
 	case 36:
 		if (op->fontsize36_begin) {
-			outstring+=QString().sprintf(op->fontsize36_begin);
+			outstring+=QString().sprintf("%s", op->fontsize36_begin);
 			found_std_expr = TRUE;
 		}
 		break;
 	case 48:
 		if (op->fontsize48_begin) {
-			outstring+=QString().sprintf(op->fontsize48_begin);
+			outstring+=QString().sprintf("%s", op->fontsize48_begin);
 			found_std_expr = TRUE;
 		}
 		break;
@@ -300,46 +300,46 @@ op_begin_std_fontsize (OutputPersonality *op, int size)
 			 * size.
 			 */
 			if (size<9 && op->fontsize8_begin) {
-				outstring+=QString().sprintf(op->fontsize8_begin);
+				outstring+=QString().sprintf("%s", op->fontsize8_begin);
 			} else 
 			if (size<11 && op->fontsize10_begin) {
-				outstring+=QString().sprintf(op->fontsize10_begin);
+				outstring+=QString().sprintf("%s", op->fontsize10_begin);
 			} else 
 			if (size<13 && op->fontsize12_begin) {
-				outstring+=QString().sprintf(op->fontsize12_begin);
+				outstring+=QString().sprintf("%s", op->fontsize12_begin);
 			} else 
 			if (size<16 && op->fontsize14_begin) {
-				outstring+=QString().sprintf(op->fontsize14_begin);
+				outstring+=QString().sprintf("%s", op->fontsize14_begin);
 			} else 
 			if (size<21 && op->fontsize18_begin) {
-				outstring+=QString().sprintf(op->fontsize18_begin);
+				outstring+=QString().sprintf("%s", op->fontsize18_begin);
 			} else 
 			if (size<30 && op->fontsize24_begin) {
-				outstring+=QString().sprintf(op->fontsize24_begin);
+				outstring+=QString().sprintf("%s", op->fontsize24_begin);
 			} else 
 			if (size<42 && op->fontsize36_begin) {
-				outstring+=QString().sprintf(op->fontsize36_begin);
+				outstring+=QString().sprintf("%s", op->fontsize36_begin);
 			} else 
 			if (size>40 && op->fontsize48_begin) {
-				outstring+=QString().sprintf(op->fontsize48_begin);
+				outstring+=QString().sprintf("%s", op->fontsize48_begin);
 			} else 
 			/* If we can't even produce a good approximation,
 			 * just try to get a font size near 12 point.
 			 */
 			if (op->fontsize12_begin)
-				outstring+=QString().sprintf(op->fontsize12_begin);
+				outstring+=QString().sprintf("%s", op->fontsize12_begin);
 			else
 			if (op->fontsize14_begin)
-				outstring+=QString().sprintf(op->fontsize14_begin);
+				outstring+=QString().sprintf("%s", op->fontsize14_begin);
 			else
 			if (op->fontsize10_begin)
-				outstring+=QString().sprintf(op->fontsize10_begin);
+				outstring+=QString().sprintf("%s", op->fontsize10_begin);
 			else
 			if (op->fontsize18_begin)
-				outstring+=QString().sprintf(op->fontsize18_begin);
+				outstring+=QString().sprintf("%s", op->fontsize18_begin);
 			else
 			if (op->fontsize8_begin)
-				outstring+=QString().sprintf(op->fontsize8_begin);
+				outstring+=QString().sprintf("%s", op->fontsize8_begin);
 			else
 				error_handler ("output personality lacks sufficient font size change capability");
 		}
@@ -367,49 +367,49 @@ op_end_std_fontsize (OutputPersonality *op, int size)
 	switch (size) {
 	case 8:
 		if (op->fontsize8_end) {
-			outstring+=QString().sprintf(op->fontsize8_end);
+			outstring+=QString().sprintf("%s", op->fontsize8_end);
 			found_std_expr = TRUE;
 		}
 		break;
 	case 10:
 		if (op->fontsize10_end) {
-			outstring+=QString().sprintf(op->fontsize10_end);
+			outstring+=QString().sprintf("%s", op->fontsize10_end);
 			found_std_expr = TRUE;
 		}
 		break;
 	case 12:
 		if (op->fontsize12_end) {
-			outstring+=QString().sprintf(op->fontsize12_end);
+			outstring+=QString().sprintf("%s", op->fontsize12_end);
 			found_std_expr = TRUE;
 		}
 		break;
 	case 14:
 		if (op->fontsize14_end) {
-			outstring+=QString().sprintf(op->fontsize14_end);
+			outstring+=QString().sprintf("%s", op->fontsize14_end);
 			found_std_expr = TRUE;
 		}
 		break;
 	case 18:
 		if (op->fontsize18_end) {
-			outstring+=QString().sprintf(op->fontsize18_end);
+			outstring+=QString().sprintf("%s", op->fontsize18_end);
 			found_std_expr = TRUE;
 		}
 		break;
 	case 24:
 		if (op->fontsize24_end) {
-			outstring+=QString().sprintf(op->fontsize24_end);
+			outstring+=QString().sprintf("%s", op->fontsize24_end);
 			found_std_expr = TRUE;
 		}
 		break;
 	case 36:
 		if (op->fontsize36_end) {
-			outstring+=QString().sprintf(op->fontsize36_end);
+			outstring+=QString().sprintf("%s", op->fontsize36_end);
 			found_std_expr = TRUE;
 		}
 		break;
 	case 48:
 		if (op->fontsize48_end) {
-			outstring+=QString().sprintf(op->fontsize48_end);
+			outstring+=QString().sprintf("%s", op->fontsize48_end);
 			found_std_expr = TRUE;
 		}
 		break;
@@ -429,46 +429,46 @@ op_end_std_fontsize (OutputPersonality *op, int size)
 			 * size.
 			 */
 			if (size<9 && op->fontsize8_end) {
-				outstring+=QString().sprintf(op->fontsize8_end);
+				outstring+=QString().sprintf("%s", op->fontsize8_end);
 			} else 
 			if (size<11 && op->fontsize10_end) {
-				outstring+=QString().sprintf(op->fontsize10_end);
+				outstring+=QString().sprintf("%s", op->fontsize10_end);
 			} else 
 			if (size<13 && op->fontsize12_end) {
-				outstring+=QString().sprintf(op->fontsize12_end);
+				outstring+=QString().sprintf("%s", op->fontsize12_end);
 			} else 
 			if (size<16 && op->fontsize14_end) {
-				outstring+=QString().sprintf(op->fontsize14_end);
+				outstring+=QString().sprintf("%s", op->fontsize14_end);
 			} else 
 			if (size<21 && op->fontsize18_end) {
-				outstring+=QString().sprintf(op->fontsize18_end);
+				outstring+=QString().sprintf("%s", op->fontsize18_end);
 			} else 
 			if (size<30 && op->fontsize24_end) {
-				outstring+=QString().sprintf(op->fontsize24_end);
+				outstring+=QString().sprintf("%s", op->fontsize24_end);
 			} else 
 			if (size<42 && op->fontsize36_end) {
-				outstring+=QString().sprintf(op->fontsize36_end);
+				outstring+=QString().sprintf("%s", op->fontsize36_end);
 			} else 
 			if (size>40 && op->fontsize48_end) {
-				outstring+=QString().sprintf(op->fontsize48_end);
+				outstring+=QString().sprintf("%s", op->fontsize48_end);
 			} else 
 			/* If we can't even produce a good approximation,
 			 * just try to get a font size near 12 point.
 			 */
 			if (op->fontsize12_end)
-				outstring+=QString().sprintf(op->fontsize12_end);
+				outstring+=QString().sprintf("%s", op->fontsize12_end);
 			else
 			if (op->fontsize14_end)
-				outstring+=QString().sprintf(op->fontsize14_end);
+				outstring+=QString().sprintf("%s", op->fontsize14_end);
 			else
 			if (op->fontsize10_end)
-				outstring+=QString().sprintf(op->fontsize10_end);
+				outstring+=QString().sprintf("%s", op->fontsize10_end);
 			else
 			if (op->fontsize18_end)
-				outstring+=QString().sprintf(op->fontsize18_end);
+				outstring+=QString().sprintf("%s", op->fontsize18_end);
 			else
 			if (op->fontsize8_end)
-				outstring+=QString().sprintf(op->fontsize8_end);
+				outstring+=QString().sprintf("%s", op->fontsize8_end);
 			else
 				error_handler ("output personality lacks sufficient font size change capability");
 		}

plugins/kicker/kicker.cpp

@@ -1,7 +1,7 @@
 /*
  * kicker.cpp - bassdrum-synthesizer
  *
- * Copyright (c) 2006-2008 Tobias Doerffel <tobydox/at/users.sourceforge.net>
+ * Copyright (c) 2006-2009 Tobias Doerffel <tobydox/at/users.sourceforge.net>
  * 
  * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
  *
@@ -33,7 +33,6 @@
 #include "note_play_handle.h"
 #include "sweep_oscillator.h"
 
-#undef SINGLE_SOURCE_COMPILE
 #include "embed.cpp"
 
 

plugins/ladspa_browser/ladspa_browser.cpp

@@ -37,13 +37,10 @@
 #include "tab_bar.h"
 #include "tab_button.h"
 
-
-#undef SINGLE_SOURCE_COMPILE
 #include "embed.cpp"
 
 
 
-
 extern "C"
 {
 

plugins/ladspa_effect/CMakeLists.txt

@@ -14,6 +14,10 @@ IF(WANT_CMT)
 ADD_SUBDIRECTORY(cmt)
 ENDIF(WANT_CMT)
 
+IF(WANT_CALF)
+ADD_SUBDIRECTORY(calf)
+ENDIF(WANT_CALF)
+
 
 INCLUDE(BuildPlugin)
 

plugins/ladspa_effect/calf/AUTHORS

@@ -0,0 +1,9 @@
+Krzysztof Foltman <wdev@foltman.com>
+Hermann Meyer <brummer-@web.de>
+Thor Harald Johansen <thj@thj.no>
+Thorsten Wilms <t_w_@freenet.de>
+Hans Baier <hansfbaier@googlemail.com>
+Torben Hohn <torbenh@gmx.de>
+
+Additional bugfixes/enhancement patches:
+David Täht <d@teklibre.com>

plugins/ladspa_effect/calf/CMakeLists.txt

@@ -0,0 +1,15 @@
+FILE(GLOB SOURCES *.cpp)
+ADD_LIBRARY(calf MODULE ${SOURCES})
+INCLUDE_DIRECTORIES(${CMAKE_SOURCE_DIR}/include ${CMAKE_CURRENT_SOURCE_DIR} ${CMAKE_CURRENT_SOURCE_DIR}/calf)
+INSTALL(TARGETS calf LIBRARY DESTINATION ${PLUGIN_DIR}/ladspa)
+ADD_DEFINITIONS(-DUSE_LADSPA=1)
+SET_TARGET_PROPERTIES(calf PROPERTIES PREFIX "")
+SET_TARGET_PROPERTIES(calf PROPERTIES COMPILE_FLAGS "-O2 -finline-limit=80 -funroll-loops")
+
+IF(LMMS_BUILD_WIN32)
+	ADD_CUSTOM_COMMAND(TARGET calf POST_BUILD COMMAND ${STRIP} ${CMAKE_CURRENT_BINARY_DIR}/calf.dll)
+ENDIF(LMMS_BUILD_WIN32)
+IF(NOT LMMS_BUILD_APPLE)
+	SET_TARGET_PROPERTIES(calf PROPERTIES LINK_FLAGS "${LINK_FLAGS} -shared -Wl,-no-undefined")
+ENDIF(NOT LMMS_BUILD_APPLE)
+

plugins/ladspa_effect/calf/COPYING

@@ -0,0 +1,504 @@
+		  GNU LESSER GENERAL PUBLIC LICENSE
+		       Version 2.1, February 1999
+
+ Copyright (C) 1991, 1999 Free Software Foundation, Inc.
+     59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ Everyone is permitted to copy and distribute verbatim copies
+ of this license document, but changing it is not allowed.
+
+[This is the first released version of the Lesser GPL.  It also counts
+ as the successor of the GNU Library Public License, version 2, hence
+ the version number 2.1.]
+
+			    Preamble
+
+  The licenses for most software are designed to take away your
+freedom to share and change it.  By contrast, the GNU General Public
+Licenses are intended to guarantee your freedom to share and change
+free software--to make sure the software is free for all its users.
+
+  This license, the Lesser General Public License, applies to some
+specially designated software packages--typically libraries--of the
+Free Software Foundation and other authors who decide to use it.  You
+can use it too, but we suggest you first think carefully about whether
+this license or the ordinary General Public License is the better
+strategy to use in any particular case, based on the explanations below.
+
+  When we speak of free software, we are referring to freedom of use,
+not price.  Our General Public Licenses are designed to make sure that
+you have the freedom to distribute copies of free software (and charge
+for this service if you wish); that you receive source code or can get
+it if you want it; that you can change the software and use pieces of
+it in new free programs; and that you are informed that you can do
+these things.
+
+  To protect your rights, we need to make restrictions that forbid
+distributors to deny you these rights or to ask you to surrender these
+rights.  These restrictions translate to certain responsibilities for
+you if you distribute copies of the library or if you modify it.
+
+  For example, if you distribute copies of the library, whether gratis
+or for a fee, you must give the recipients all the rights that we gave
+you.  You must make sure that they, too, receive or can get the source
+code.  If you link other code with the library, you must provide
+complete object files to the recipients, so that they can relink them
+with the library after making changes to the library and recompiling
+it.  And you must show them these terms so they know their rights.
+
+  We protect your rights with a two-step method: (1) we copyright the
+library, and (2) we offer you this license, which gives you legal
+permission to copy, distribute and/or modify the library.
+
+  To protect each distributor, we want to make it very clear that
+there is no warranty for the free library.  Also, if the library is
+modified by someone else and passed on, the recipients should know
+that what they have is not the original version, so that the original
+author's reputation will not be affected by problems that might be
+introduced by others.
+
+  Finally, software patents pose a constant threat to the existence of
+any free program.  We wish to make sure that a company cannot
+effectively restrict the users of a free program by obtaining a
+restrictive license from a patent holder.  Therefore, we insist that
+any patent license obtained for a version of the library must be
+consistent with the full freedom of use specified in this license.
+
+  Most GNU software, including some libraries, is covered by the
+ordinary GNU General Public License.  This license, the GNU Lesser
+General Public License, applies to certain designated libraries, and
+is quite different from the ordinary General Public License.  We use
+this license for certain libraries in order to permit linking those
+libraries into non-free programs.
+
+  When a program is linked with a library, whether statically or using
+a shared library, the combination of the two is legally speaking a
+combined work, a derivative of the original library.  The ordinary
+General Public License therefore permits such linking only if the
+entire combination fits its criteria of freedom.  The Lesser General
+Public License permits more lax criteria for linking other code with
+the library.
+
+  We call this license the "Lesser" General Public License because it
+does Less to protect the user's freedom than the ordinary General
+Public License.  It also provides other free software developers Less
+of an advantage over competing non-free programs.  These disadvantages
+are the reason we use the ordinary General Public License for many
+libraries.  However, the Lesser license provides advantages in certain
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+
+  For example, on rare occasions, there may be a special need to
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+case, there is little to gain by limiting the free library to free
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+
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+non-free programs enables many more people to use the whole GNU
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+
+  Although the Lesser General Public License is Less protective of the
+users' freedom, it does ensure that the user of a program that is
+linked with the Library has the freedom and the wherewithal to run
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+former contains code derived from the library, whereas the latter must
+be combined with the library in order to run.
+
+		  GNU LESSER GENERAL PUBLIC LICENSE
+   TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
+
+  0. This License Agreement applies to any software library or other
+program which contains a notice placed by the copyright holder or
+other authorized party saying it may be distributed under the terms of
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+covered by this License; they are outside its scope.  The act of
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+  1. You may copy and distribute verbatim copies of the Library's
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+Thus, it is not the intent of this section to claim rights or contest
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+In addition, mere aggregation of another work not based on the Library
+with the Library (or with a work based on the Library) on a volume of
+a storage or distribution medium does not bring the other work under
+the scope of this License.
+
+  3. You may opt to apply the terms of the ordinary GNU General Public
+License instead of this License to a given copy of the Library.  To do
+this, you must alter all the notices that refer to this License, so
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+
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+Section 6 states terms for distribution of such executables.
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+  When a "work that uses the Library" uses material from a header file
+that is part of the Library, the object code for the work may be a
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+
+  6. As an exception to the Sections above, you may also combine or
+link a "work that uses the Library" with the Library to produce a
+work containing portions of the Library, and distribute that work
+under terms of your choice, provided that the terms permit
+modification of the work for the customer's own use and reverse
+engineering for debugging such modifications.
+
+  You must give prominent notice with each copy of the work that the
+Library is used in it and that the Library and its use are covered by
+this License.  You must supply a copy of this License.  If the work
+during execution displays copyright notices, you must include the
+copyright notice for the Library among them, as well as a reference
+directing the user to the copy of this License.  Also, you must do one
+of these things:
+
+    a) Accompany the work with the complete corresponding
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+    specified materials from the same place.
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+
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+distribute.
+
+  7. You may place library facilities that are a work based on the
+Library side-by-side in a single library together with other library
+facilities not covered by this License, and distribute such a combined
+library, provided that the separate distribution of the work based on
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+permitted, and provided that you do these two things:
+
+    a) Accompany the combined library with a copy of the same work
+    based on the Library, uncombined with any other library
+    facilities.  This must be distributed under the terms of the
+    Sections above.
+
+    b) Give prominent notice with the combined library of the fact
+    that part of it is a work based on the Library, and explaining
+    where to find the accompanying uncombined form of the same work.
+
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+the Library except as expressly provided under this License.  Any
+attempt otherwise to copy, modify, sublicense, link with, or
+distribute the Library is void, and will automatically terminate your
+rights under this License.  However, parties who have received copies,
+or rights, from you under this License will not have their licenses
+terminated so long as such parties remain in full compliance.
+
+  9. You are not required to accept this License, since you have not
+signed it.  However, nothing else grants you permission to modify or
+distribute the Library or its derivative works.  These actions are
+prohibited by law if you do not accept this License.  Therefore, by
+modifying or distributing the Library (or any work based on the
+Library), you indicate your acceptance of this License to do so, and
+all its terms and conditions for copying, distributing or modifying
+the Library or works based on it.
+
+  10. Each time you redistribute the Library (or any work based on the
+Library), the recipient automatically receives a license from the
+original licensor to copy, distribute, link with or modify the Library
+subject to these terms and conditions.  You may not impose any further
+restrictions on the recipients' exercise of the rights granted herein.
+You are not responsible for enforcing compliance by third parties with
+this License.
+
+  11. If, as a consequence of a court judgment or allegation of patent
+infringement or for any other reason (not limited to patent issues),
+conditions are imposed on you (whether by court order, agreement or
+otherwise) that contradict the conditions of this License, they do not
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+all those who receive copies directly or indirectly through you, then
+the only way you could satisfy both it and this License would be to
+refrain entirely from distribution of the Library.
+
+If any portion of this section is held invalid or unenforceable under any
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+
+It is not the purpose of this section to induce you to infringe any
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+integrity of the free software distribution system which is
+implemented by public license practices.  Many people have made
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+to distribute software through any other system and a licensee cannot
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+
+This section is intended to make thoroughly clear what is believed to
+be a consequence of the rest of this License.
+
+  12. If the distribution and/or use of the Library is restricted in
+certain countries either by patents or by copyrighted interfaces, the
+original copyright holder who places the Library under this License may add
+an explicit geographical distribution limitation excluding those countries,
+so that distribution is permitted only in or among countries not thus
+excluded.  In such case, this License incorporates the limitation as if
+written in the body of this License.
+
+  13. The Free Software Foundation may publish revised and/or new
+versions of the Lesser General Public License from time to time.
+Such new versions will be similar in spirit to the present version,
+but may differ in detail to address new problems or concerns.
+
+Each version is given a distinguishing version number.  If the Library
+specifies a version number of this License which applies to it and
+"any later version", you have the option of following the terms and
+conditions either of that version or of any later version published by
+the Free Software Foundation.  If the Library does not specify a
+license version number, you may choose any version ever published by
+the Free Software Foundation.
+
+  14. If you wish to incorporate parts of the Library into other free
+programs whose distribution conditions are incompatible with these,
+write to the author to ask for permission.  For software which is
+copyrighted by the Free Software Foundation, write to the Free
+Software Foundation; we sometimes make exceptions for this.  Our
+decision will be guided by the two goals of preserving the free status
+of all derivatives of our free software and of promoting the sharing
+and reuse of software generally.
+
+			    NO WARRANTY
+
+  15. BECAUSE THE LIBRARY IS LICENSED FREE OF CHARGE, THERE IS NO
+WARRANTY FOR THE LIBRARY, TO THE EXTENT PERMITTED BY APPLICABLE LAW.
+EXCEPT WHEN OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR
+OTHER PARTIES PROVIDE THE LIBRARY "AS IS" WITHOUT WARRANTY OF ANY
+KIND, EITHER EXPRESSED OR IMPLIED, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
+PURPOSE.  THE ENTIRE RISK AS TO THE QUALITY AND PERFORMANCE OF THE
+LIBRARY IS WITH YOU.  SHOULD THE LIBRARY PROVE DEFECTIVE, YOU ASSUME
+THE COST OF ALL NECESSARY SERVICING, REPAIR OR CORRECTION.
+
+  16. IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN
+WRITING WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MAY MODIFY
+AND/OR REDISTRIBUTE THE LIBRARY AS PERMITTED ABOVE, BE LIABLE TO YOU
+FOR DAMAGES, INCLUDING ANY GENERAL, SPECIAL, INCIDENTAL OR
+CONSEQUENTIAL DAMAGES ARISING OUT OF THE USE OR INABILITY TO USE THE
+LIBRARY (INCLUDING BUT NOT LIMITED TO LOSS OF DATA OR DATA BEING
+RENDERED INACCURATE OR LOSSES SUSTAINED BY YOU OR THIRD PARTIES OR A
+FAILURE OF THE LIBRARY TO OPERATE WITH ANY OTHER SOFTWARE), EVEN IF
+SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH
+DAMAGES.
+
+		     END OF TERMS AND CONDITIONS
+
+           How to Apply These Terms to Your New Libraries
+
+  If you develop a new library, and you want it to be of the greatest
+possible use to the public, we recommend making it free software that
+everyone can redistribute and change.  You can do so by permitting
+redistribution under these terms (or, alternatively, under the terms of the
+ordinary General Public License).
+
+  To apply these terms, attach the following notices to the library.  It is
+safest to attach them to the start of each source file to most effectively
+convey the exclusion of warranty; and each file should have at least the
+"copyright" line and a pointer to where the full notice is found.
+
+    <one line to give the library's name and a brief idea of what it does.>
+    Copyright (C) <year>  <name of author>
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+
+Also add information on how to contact you by electronic and paper mail.
+
+You should also get your employer (if you work as a programmer) or your
+school, if any, to sign a "copyright disclaimer" for the library, if
+necessary.  Here is a sample; alter the names:
+
+  Yoyodyne, Inc., hereby disclaims all copyright interest in the
+  library `Frob' (a library for tweaking knobs) written by James Random Hacker.
+
+  <signature of Ty Coon>, 1 April 1990
+  Ty Coon, President of Vice
+
+That's all there is to it!
+
+

plugins/ladspa_effect/calf/COPYING.GPL

@@ -0,0 +1,340 @@
+		    GNU GENERAL PUBLIC LICENSE
+		       Version 2, June 1991
+
+ Copyright (C) 1989, 1991 Free Software Foundation, Inc.
+     59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ Everyone is permitted to copy and distribute verbatim copies
+ of this license document, but changing it is not allowed.
+
+			    Preamble
+
+  The licenses for most software are designed to take away your
+freedom to share and change it.  By contrast, the GNU General Public
+License is intended to guarantee your freedom to share and change free
+software--to make sure the software is free for all its users.  This
+General Public License applies to most of the Free Software
+Foundation's software and to any other program whose authors commit to
+using it.  (Some other Free Software Foundation software is covered by
+the GNU Library General Public License instead.)  You can apply it to
+your programs, too.
+
+  When we speak of free software, we are referring to freedom, not
+price.  Our General Public Licenses are designed to make sure that you
+have the freedom to distribute copies of free software (and charge for
+this service if you wish), that you receive source code or can get it
+if you want it, that you can change the software or use pieces of it
+in new free programs; and that you know you can do these things.
+
+  To protect your rights, we need to make restrictions that forbid
+anyone to deny you these rights or to ask you to surrender the rights.
+These restrictions translate to certain responsibilities for you if you
+distribute copies of the software, or if you modify it.
+
+  For example, if you distribute copies of such a program, whether
+gratis or for a fee, you must give the recipients all the rights that
+you have.  You must make sure that they, too, receive or can get the
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+rights.
+
+  We protect your rights with two steps: (1) copyright the software, and
+(2) offer you this license which gives you legal permission to copy,
+distribute and/or modify the software.
+
+  Also, for each author's protection and ours, we want to make certain
+that everyone understands that there is no warranty for this free
+software.  If the software is modified by someone else and passed on, we
+want its recipients to know that what they have is not the original, so
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+
+  Finally, any free program is threatened constantly by software
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+
+  The precise terms and conditions for copying, distribution and
+modification follow.
+
+		    GNU GENERAL PUBLIC LICENSE
+   TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
+
+  0. This License applies to any program or other work which contains
+a notice placed by the copyright holder saying it may be distributed
+under the terms of this General Public License.  The "Program", below,
+refers to any such program or work, and a "work based on the Program"
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+that is to say, a work containing the Program or a portion of it,
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+language.  (Hereinafter, translation is included without limitation in
+the term "modification".)  Each licensee is addressed as "you".
+
+Activities other than copying, distribution and modification are not
+covered by this License; they are outside its scope.  The act of
+running the Program is not restricted, and the output from the Program
+is covered only if its contents constitute a work based on the
+Program (independent of having been made by running the Program).
+Whether that is true depends on what the Program does.
+
+  1. You may copy and distribute verbatim copies of the Program's
+source code as you receive it, in any medium, provided that you
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+notices that refer to this License and to the absence of any warranty;
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+along with the Program.
+
+You may charge a fee for the physical act of transferring a copy, and
+you may at your option offer warranty protection in exchange for a fee.
+
+  2. You may modify your copy or copies of the Program or any portion
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+distribute such modifications or work under the terms of Section 1
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+
+    a) You must cause the modified files to carry prominent notices
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+
+    b) You must cause any work that you distribute or publish, that in
+    whole or in part contains or is derived from the Program or any
+    part thereof, to be licensed as a whole at no charge to all third
+    parties under the terms of this License.
+
+    c) If the modified program normally reads commands interactively
+    when run, you must cause it, when started running for such
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+    does not normally print such an announcement, your work based on
+    the Program is not required to print an announcement.)
+
+These requirements apply to the modified work as a whole.  If
+identifiable sections of that work are not derived from the Program,
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+
+Thus, it is not the intent of this section to claim rights or contest
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+
+  4. You may not copy, modify, sublicense, or distribute the Program
+except as expressly provided under this License.  Any attempt
+otherwise to copy, modify, sublicense or distribute the Program is
+void, and will automatically terminate your rights under this License.
+However, parties who have received copies, or rights, from you under
+this License will not have their licenses terminated so long as such
+parties remain in full compliance.
+
+  5. You are not required to accept this License, since you have not
+signed it.  However, nothing else grants you permission to modify or
+distribute the Program or its derivative works.  These actions are
+prohibited by law if you do not accept this License.  Therefore, by
+modifying or distributing the Program (or any work based on the
+Program), you indicate your acceptance of this License to do so, and
+all its terms and conditions for copying, distributing or modifying
+the Program or works based on it.
+
+  6. Each time you redistribute the Program (or any work based on the
+Program), the recipient automatically receives a license from the
+original licensor to copy, distribute or modify the Program subject to
+these terms and conditions.  You may not impose any further
+restrictions on the recipients' exercise of the rights granted herein.
+You are not responsible for enforcing compliance by third parties to
+this License.
+
+  7. If, as a consequence of a court judgment or allegation of patent
+infringement or for any other reason (not limited to patent issues),
+conditions are imposed on you (whether by court order, agreement or
+otherwise) that contradict the conditions of this License, they do not
+excuse you from the conditions of this License.  If you cannot
+distribute so as to satisfy simultaneously your obligations under this
+License and any other pertinent obligations, then as a consequence you
+may not distribute the Program at all.  For example, if a patent
+license would not permit royalty-free redistribution of the Program by
+all those who receive copies directly or indirectly through you, then
+the only way you could satisfy both it and this License would be to
+refrain entirely from distribution of the Program.
+
+If any portion of this section is held invalid or unenforceable under
+any particular circumstance, the balance of the section is intended to
+apply and the section as a whole is intended to apply in other
+circumstances.
+
+It is not the purpose of this section to induce you to infringe any
+patents or other property right claims or to contest validity of any
+such claims; this section has the sole purpose of protecting the
+integrity of the free software distribution system, which is
+implemented by public license practices.  Many people have made
+generous contributions to the wide range of software distributed
+through that system in reliance on consistent application of that
+system; it is up to the author/donor to decide if he or she is willing
+to distribute software through any other system and a licensee cannot
+impose that choice.
+
+This section is intended to make thoroughly clear what is believed to
+be a consequence of the rest of this License.
+
+  8. If the distribution and/or use of the Program is restricted in
+certain countries either by patents or by copyrighted interfaces, the
+original copyright holder who places the Program under this License
+may add an explicit geographical distribution limitation excluding
+those countries, so that distribution is permitted only in or among
+countries not thus excluded.  In such case, this License incorporates
+the limitation as if written in the body of this License.
+
+  9. The Free Software Foundation may publish revised and/or new versions
+of the General Public License from time to time.  Such new versions will
+be similar in spirit to the present version, but may differ in detail to
+address new problems or concerns.
+
+Each version is given a distinguishing version number.  If the Program
+specifies a version number of this License which applies to it and "any
+later version", you have the option of following the terms and conditions
+either of that version or of any later version published by the Free
+Software Foundation.  If the Program does not specify a version number of
+this License, you may choose any version ever published by the Free Software
+Foundation.
+
+  10. If you wish to incorporate parts of the Program into other free
+programs whose distribution conditions are different, write to the author
+to ask for permission.  For software which is copyrighted by the Free
+Software Foundation, write to the Free Software Foundation; we sometimes
+make exceptions for this.  Our decision will be guided by the two goals
+of preserving the free status of all derivatives of our free software and
+of promoting the sharing and reuse of software generally.
+
+			    NO WARRANTY
+
+  11. BECAUSE THE PROGRAM IS LICENSED FREE OF CHARGE, THERE IS NO WARRANTY
+FOR THE PROGRAM, TO THE EXTENT PERMITTED BY APPLICABLE LAW.  EXCEPT WHEN
+OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR OTHER PARTIES
+PROVIDE THE PROGRAM "AS IS" WITHOUT WARRANTY OF ANY KIND, EITHER EXPRESSED
+OR IMPLIED, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE.  THE ENTIRE RISK AS
+TO THE QUALITY AND PERFORMANCE OF THE PROGRAM IS WITH YOU.  SHOULD THE
+PROGRAM PROVE DEFECTIVE, YOU ASSUME THE COST OF ALL NECESSARY SERVICING,
+REPAIR OR CORRECTION.
+
+  12. IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING
+WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MAY MODIFY AND/OR
+REDISTRIBUTE THE PROGRAM AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES,
+INCLUDING ANY GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING
+OUT OF THE USE OR INABILITY TO USE THE PROGRAM (INCLUDING BUT NOT LIMITED
+TO LOSS OF DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY
+YOU OR THIRD PARTIES OR A FAILURE OF THE PROGRAM TO OPERATE WITH ANY OTHER
+PROGRAMS), EVEN IF SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGES.
+
+		     END OF TERMS AND CONDITIONS
+
+	    How to Apply These Terms to Your New Programs
+
+  If you develop a new program, and you want it to be of the greatest
+possible use to the public, the best way to achieve this is to make it
+free software which everyone can redistribute and change under these terms.
+
+  To do so, attach the following notices to the program.  It is safest
+to attach them to the start of each source file to most effectively
+convey the exclusion of warranty; and each file should have at least
+the "copyright" line and a pointer to where the full notice is found.
+
+    <one line to give the program's name and a brief idea of what it does.>
+    Copyright (C) <year>  <name of author>
+
+    This program is free software; you can redistribute it and/or modify
+    it under the terms of the GNU General Public License as published by
+    the Free Software Foundation; either version 2 of the License, or
+    (at your option) any later version.
+
+    This program is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+    GNU General Public License for more details.
+
+    You should have received a copy of the GNU General Public License
+    along with this program; if not, write to the Free Software
+    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+
+
+Also add information on how to contact you by electronic and paper mail.
+
+If the program is interactive, make it output a short notice like this
+when it starts in an interactive mode:
+
+    Gnomovision version 69, Copyright (C) year  name of author
+    Gnomovision comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
+    This is free software, and you are welcome to redistribute it
+    under certain conditions; type `show c' for details.
+
+The hypothetical commands `show w' and `show c' should show the appropriate
+parts of the General Public License.  Of course, the commands you use may
+be called something other than `show w' and `show c'; they could even be
+mouse-clicks or menu items--whatever suits your program.
+
+You should also get your employer (if you work as a programmer) or your
+school, if any, to sign a "copyright disclaimer" for the program, if
+necessary.  Here is a sample; alter the names:
+
+  Yoyodyne, Inc., hereby disclaims all copyright interest in the program
+  `Gnomovision' (which makes passes at compilers) written by James Hacker.
+
+  <signature of Ty Coon>, 1 April 1989
+  Ty Coon, President of Vice
+
+This General Public License does not permit incorporating your program into
+proprietary programs.  If your program is a subroutine library, you may
+consider it more useful to permit linking proprietary applications with the
+library.  If this is what you want to do, use the GNU Library General
+Public License instead of this License.

plugins/ladspa_effect/calf/ChangeLog

@@ -0,0 +1,170 @@
+Version 0.0.18.2
+
++ Organ: fix voice stealing of released notes, sort out GUI, add quadratic
+  mode for amplitude envelope (enabled by default) - sounds more natural
++ Monosynth: fix the bug that caused JACK to kick the client out due
+  to precalculating waves in a completely wrong place, fix portamento
+  for off-stack notes
++ Presets: 3 new presets for Organ, 4 for Monosynth, 2 for Reverb
+
+Version 0.0.18.1
+
++ Filter: fixed subtle redraw bugs
++ Icons: fixed packaging-incompatible paths
+
+Version 0.0.18
+
++ Filterclavier: new plugin (a MIDI controlled filter) by Hans Baier
++ DSSI: added a basic implementation of live graphs. The graphs have a
+  limited resolution (128 data points), and are rather inefficient
+  (as the graph data need to be transmitted via OSC to a different
+  process), but it's better than nothing
++ GUI: Torben Hohn's drawing optimizations (critical for Intel graphics
+  cards, but should also reduce CPU usage on other hardware)
++ Phaser: added frequency response graph
++ JACK host: discontinue the broken option -p; allow giving preset names
+  after a colon sign (reverb:DiscoVerb instead of -p DiscoVerb reverb)
++ Reverb: less modulation; tone controls; 2 more room types
++ MultiChorus: add double bandpass filter on input
++ GUI: added frequency grid
++ Organ: added progress reporting on load (works with JACK host and LV2)
++ JACK host: use sensible port names (possibly breaking new LASH sessions)
++ Organ: added polyphony limit
++ Small plugins: added support for polymorphic port extension to allow
+  the same plugins to be used for control and audio signals
++ DSSI: renamed all the plugins from "plugin LADSPA" to "plugin DSSI"
++ LADSPA: more reasonable default value hints, fixed locale issue in LRDF
++ JACK host: added icons by Thorsten Wilms (thanks!)
++ Organ, Monosynth: better memory usage
++ LV2: attempt at supporting configure-like parameters (key mapping curve
+  in Organ) by the new String Port extension
++ AutoHell: header files are not installed anymore (they are of little
+  use anyway)
++ AutoHell: configure script prints if --enable-experimental was specified
+
+Version 0.0.17
+
++ Compressor: new plugin by Thor Harald Johansen
++ GUI: control improvements (new LED control, improved VU meter, XML
+  improvements, line graph with dots and grid lines - no legend yet), move
+  autolayout code from the plugin libraries to makerdf executable,
++ Most plugins: use custom GUI layouts instead of autogenerated ones
++ Most plugins: add dry amount (for aux bus type uses)
++ Flanger, Filter, MultiChorus: added live graphs displaying frequency
+  response and (in case of MultiChorus) LFO positions
++ LV2 GUI: added a way to display live graphs in Ardour and Zynjacku/LV2Rack
+  (only works when the plugin and the GUI are in the same process)
++ Framework: general improvements/cleanups to reduce the chance of the
+  kind of errors that were introduced in 0.0.16 and reduce dependencies
++ Monosynth: removed soft clipper on output
+
+Version 0.0.16.3
+
++ Fixed compilation without LV2 core installed
+
+Version 0.0.16.2
+
++ Fixed DSSI GUI for MultiChorus
++ Fixed LV2 GUI for MultiChorus
++ Make knob control mouse wheel handling work better in Ingen
+
+Version 0.0.16
+
++ New MultiChorus plugin (stereo multitap chorus with maximum of 8 voices)
++ Experimental set of plugins for modular synthesizers like Ingen by 
+  Dave Robillard (enabled using --enable-experimental option in configure
+  script)
++ Minor improvements to other plugins (like Rotary Speaker)
++ More work on API documentation
+
+Version 0.0.15
+
++ Organ: new percussive section, using 2-operator FM synthesis for
+  monophonic or polyphonic percussive attack; added global transpose and
+  detune; rearrangement of controls between sections
++ Rotary Speaker: another attempt at making it useful (thanks FishB8)
++ JACK host: eliminate deadlock on exit
++ GUI: bipolar knobs now have a "dead zone" (magnet) in the middle point
++ GUI: dragging a knob with SHIFT held allows for fine adjustments
++ GUI: new controls - curve editor and keyboard
++ LV2: improved extension support (supports my "extended port properties"
+  extension now)
++ Added some API documentation
+
+Version 0.0.14
++ OSC: totally new OSC wrapper, to allow for realtime-safe parsing (doesn't
+  matter as far as functionality goes, will probably be rewritten again
+  anyway)
++ Everything: memory management fixes (should improve stability and
+  compatibility)
++ Organ: improved memory usage
++ GUI: improved bipolar knobs, added endless knobs
++ Presets: separate 'built-in' and 'user' presets (so that built-in presets
+  can be upgraded without affecting user's own presets)
++ Monosynth: new presets
+
+Version 0.0.13
++ Fixed several problems related to 64-bit environments and OpenSUSE (thanks
+oc2pus!)
++ Added NOCONFIGURE environment variable support to autogen.sh
+
+Version 0.0.12
++ RotarySpeaker: work in progress; enabled by default just in case it's
+  useful for anyone
++ Organ: reworked to add a complete subtractive synth section, a selection
+  of waveform (settable on a per-drawbar basis), individual settings of
+  phase, detune, panning, routing for each drawbar, as well as improved(?)
+  percussive section and vibrato/phaser section. It is usable (and sounds
+  good!), but some parameters, waveform set etc. may change in future. May
+  take up to 100 MB of RAM due to pre-calculated bandlimited waveforms.
++ Added half-complete implementation of LV2 (including GUI and events).
++ Lots of small "polishing" kind of fixes in many places (like proper
+  rounding of values in the GUIs, another set of hold/sostenuto fixes etc)
+
+Version 0.0.11
+
++ Fixed x86-64 bugs
++ JackHost: implemented LASH support
++ RotarySpeaker: fixed panning bug, implemented acceleration/decceleration
+  for "off" state
+
+Version 0.0.10
+
++ First attempt at DSSI GUI, does not support some features from JACK host,
+  but that's inevitable because of API limitations
++ Reverb: improvements (more parameters, fixed denormals)
++ Knob: added custom support for scroll wheel (instead of one inherited from
+  GtkRange)
+
+Version 0.0.9
+
++ started creating an XML-based GUI
++ LineGraph: new GTK+ control for displaying waveforms and filter response
+  graphs in Monosynth (and maybe others in future)
++ Monosynth: notch filter changes (made notch bandwidth proportional to Q,
+  just for fun, might be a bad idea)
++ Monosynth: more waveforms (these might be final?)
++ Monosynth: capped Sustain level to 0.999 so that decay time actually means
+  something with Sustain = 100% (not a great way to do it, but acceptable in
+  this case)
++ Monosynth: GUI refreshes less often (which means less CPU use)
++ Monosynth: less clicks on sounds using LP filter with very low cutoff 
+  (using ramp of 256 samples instead of 64 samples as before)
++ Knob: new GTK+ control based on GtkRange, with my primitive bitmap set
+  (generated with Python and Cairo)
++ Organ: added a GUI too, very provisional
++ Organ: fixed Hold pedal (doesn't release the notes which are still depressed)
++ RotarySpeaker: new effect (split off Organ)
++ all: denormal fixes (still some denormals present in reverb)
++ Reverb: better time setting (decay time somewhat corresponds to -60dB
+  attenuation time)
++ JackHost: -M switch allows for automatic connection to JACK MIDI event source
+  (use -M system:midi_capture_2 or -M 2 for autoconnection to
+  system:midi_capture_2; of course, the short numeric form only work for
+  system:midi_capture_ ports)
++ JackHost: -p switch selects a preset automatically
++ JackHost: better size setting algorithm
++ JackHost: duplicate client name (causing JACK to rename the client) doesn't
+  break autoconnecting functionality
++ autotools configuration update (detect Cairo and require newer GTK+)
++ more presets

plugins/ladspa_effect/calf/INSTALL

@@ -0,0 +1,254 @@
+Installation Instructions
+*************************
+
+Copyright (C) 1994, 1995, 1996, 1999, 2000, 2001, 2002, 2004, 2005 Free
+Software Foundation, Inc.
+Copyright (C) 2007-2008 Krzysztof Foltman
+
+This file is free documentation; the Free Software Foundation gives
+unlimited permission to copy, distribute and modify it.
+
+Prerequisites
+=============
+
+To compile and install Calf, you need:
+
+- POSIX-compliant operating system
+- G++ version 4.0 or higher (tested with 4.1.3)
+- GTK+2 headers and libraries (glib 2.10, gtk+ 2.12)
+- Cairo headers and libraries
+- Glade 2 headers and libraries
+
+Optional but recommended:
+- JACK header and libraries (tested with 0.109.0)
+- LADSPA header
+- DSSI header
+- LV2 core
+
+Basic Installation
+==================
+
+These are generic installation instructions.
+
+   The `configure' shell script attempts to guess correct values for
+various system-dependent variables used during compilation.  It uses
+those values to create a `Makefile' in each directory of the package.
+It may also create one or more `.h' files containing system-dependent
+definitions.  Finally, it creates a shell script `config.status' that
+you can run in the future to recreate the current configuration, and a
+file `config.log' containing compiler output (useful mainly for
+debugging `configure').
+
+   It can also use an optional file (typically called `config.cache'
+and enabled with `--cache-file=config.cache' or simply `-C') that saves
+the results of its tests to speed up reconfiguring.  (Caching is
+disabled by default to prevent problems with accidental use of stale
+cache files.)
+
+   If you need to do unusual things to compile the package, please try
+to figure out how `configure' could check whether to do them, and mail
+diffs or instructions to the address given in the `README' so they can
+be considered for the next release.  If you are using the cache, and at
+some point `config.cache' contains results you don't want to keep, you
+may remove or edit it.
+
+   The file `configure.ac' (or `configure.in') is used to create
+`configure' by a program called `autoconf'.  You only need
+`configure.ac' if you want to change it or regenerate `configure' using
+a newer version of `autoconf'.
+
+The simplest way to compile this package is:
+
+  1. `cd' to the directory containing the package's source code and type
+     `./configure' to configure the package for your system.  If you're
+     using `csh' on an old version of System V, you might need to type
+     `sh ./configure' instead to prevent `csh' from trying to execute
+     `configure' itself.
+
+     Running `configure' takes awhile.  While running, it prints some
+     messages telling which features it is checking for.
+
+  2. Type `make' to compile the package.
+
+  3. Optionally, type `make check' to run any self-tests that come with
+     the package.
+
+  4. Type `make install' to install the programs and any data files and
+     documentation.
+
+  5. You can remove the program binaries and object files from the
+     source code directory by typing `make clean'.  To also remove the
+     files that `configure' created (so you can compile the package for
+     a different kind of computer), type `make distclean'.  There is
+     also a `make maintainer-clean' target, but that is intended mainly
+     for the package's developers.  If you use it, you may have to get
+     all sorts of other programs in order to regenerate files that came
+     with the distribution.
+
+Compilers and Options
+=====================
+
+Some systems require unusual options for compilation or linking that the
+`configure' script does not know about.  Run `./configure --help' for
+details on some of the pertinent environment variables.
+
+   You can give `configure' initial values for configuration parameters
+by setting variables in the command line or in the environment.  Here
+is an example:
+
+     ./configure CC=c89 CFLAGS=-O2 LIBS=-lposix
+
+   *Note Defining Variables::, for more details.
+
+Compiling For Multiple Architectures
+====================================
+
+You can compile the package for more than one kind of computer at the
+same time, by placing the object files for each architecture in their
+own directory.  To do this, you must use a version of `make' that
+supports the `VPATH' variable, such as GNU `make'.  `cd' to the
+directory where you want the object files and executables to go and run
+the `configure' script.  `configure' automatically checks for the
+source code in the directory that `configure' is in and in `..'.
+
+   If you have to use a `make' that does not support the `VPATH'
+variable, you have to compile the package for one architecture at a
+time in the source code directory.  After you have installed the
+package for one architecture, use `make distclean' before reconfiguring
+for another architecture.
+
+Installation Names
+==================
+
+By default, `make install' installs the package's commands under
+`/usr/local/bin', include files under `/usr/local/include', etc.  You
+can specify an installation prefix other than `/usr/local' by giving
+`configure' the option `--prefix=PREFIX'.
+
+   You can specify separate installation prefixes for
+architecture-specific files and architecture-independent files.  If you
+pass the option `--exec-prefix=PREFIX' to `configure', the package uses
+PREFIX as the prefix for installing programs and libraries.
+Documentation and other data files still use the regular prefix.
+
+   In addition, if you use an unusual directory layout you can give
+options like `--bindir=DIR' to specify different values for particular
+kinds of files.  Run `configure --help' for a list of the directories
+you can set and what kinds of files go in them.
+
+   If the package supports it, you can cause programs to be installed
+with an extra prefix or suffix on their names by giving `configure' the
+option `--program-prefix=PREFIX' or `--program-suffix=SUFFIX'.
+
+Optional Features
+=================
+
+Some packages pay attention to `--enable-FEATURE' options to
+`configure', where FEATURE indicates an optional part of the package.
+They may also pay attention to `--with-PACKAGE' options, where PACKAGE
+is something like `gnu-as' or `x' (for the X Window System).  The
+`README' should mention any `--enable-' and `--with-' options that the
+package recognizes.
+
+   For packages that use the X Window System, `configure' can usually
+find the X include and library files automatically, but if it doesn't,
+you can use the `configure' options `--x-includes=DIR' and
+`--x-libraries=DIR' to specify their locations.
+
+Specifying the System Type
+==========================
+
+There may be some features `configure' cannot figure out automatically,
+but needs to determine by the type of machine the package will run on.
+Usually, assuming the package is built to be run on the _same_
+architectures, `configure' can figure that out, but if it prints a
+message saying it cannot guess the machine type, give it the
+`--build=TYPE' option.  TYPE can either be a short name for the system
+type, such as `sun4', or a canonical name which has the form:
+
+     CPU-COMPANY-SYSTEM
+
+where SYSTEM can have one of these forms:
+
+     OS KERNEL-OS
+
+   See the file `config.sub' for the possible values of each field.  If
+`config.sub' isn't included in this package, then this package doesn't
+need to know the machine type.
+
+   If you are _building_ compiler tools for cross-compiling, you should
+use the option `--target=TYPE' to select the type of system they will
+produce code for.
+
+   If you want to _use_ a cross compiler, that generates code for a
+platform different from the build platform, you should specify the
+"host" platform (i.e., that on which the generated programs will
+eventually be run) with `--host=TYPE'.
+
+Sharing Defaults
+================
+
+If you want to set default values for `configure' scripts to share, you
+can create a site shell script called `config.site' that gives default
+values for variables like `CC', `cache_file', and `prefix'.
+`configure' looks for `PREFIX/share/config.site' if it exists, then
+`PREFIX/etc/config.site' if it exists.  Or, you can set the
+`CONFIG_SITE' environment variable to the location of the site script.
+A warning: not all `configure' scripts look for a site script.
+
+Defining Variables
+==================
+
+Variables not defined in a site shell script can be set in the
+environment passed to `configure'.  However, some packages may run
+configure again during the build, and the customized values of these
+variables may be lost.  In order to avoid this problem, you should set
+them in the `configure' command line, using `VAR=value'.  For example:
+
+     ./configure CC=/usr/local2/bin/gcc
+
+causes the specified `gcc' to be used as the C compiler (unless it is
+overridden in the site shell script).  Here is a another example:
+
+     /bin/bash ./configure CONFIG_SHELL=/bin/bash
+
+Here the `CONFIG_SHELL=/bin/bash' operand causes subsequent
+configuration-related scripts to be executed by `/bin/bash'.
+
+`configure' Invocation
+======================
+
+`configure' recognizes the following options to control how it operates.
+
+`--help'
+`-h'
+     Print a summary of the options to `configure', and exit.
+
+`--version'
+`-V'
+     Print the version of Autoconf used to generate the `configure'
+     script, and exit.
+
+`--cache-file=FILE'
+     Enable the cache: use and save the results of the tests in FILE,
+     traditionally `config.cache'.  FILE defaults to `/dev/null' to
+     disable caching.
+
+`--config-cache'
+`-C'
+     Alias for `--cache-file=config.cache'.
+
+`--quiet'
+`--silent'
+`-q'
+     Do not print messages saying which checks are being made.  To
+     suppress all normal output, redirect it to `/dev/null' (any error
+     messages will still be shown).
+
+`--srcdir=DIR'
+     Look for the package's source code in directory DIR.  Usually
+     `configure' can determine that directory automatically.
+
+`configure' also accepts some other, not widely useful, options.  Run
+`configure --help' for more details.
+

plugins/ladspa_effect/calf/README

@@ -0,0 +1,49 @@
+Calf is a pack of audio plugins - effects and instruments, currently in
+development. The goal is to create a set of plugins using decent algorithms
+and parameter settings, available in a form which is compatible with as many
+open source applications as possible.
+
+How to use Calf plugins:
+
+* LADSPA plugins
+
+Calf is installed as calf.so library in your LADSPA directory (typically
+/usr/lib/ladspa). It means that typical LADSPA host should be able to find
+Calf's plugins.
+
+* DSSI plugins
+
+Calf .so module is also installed in your DSSI plugin directory, which means
+your DSSI host (like jack-dssi-host or rosegarden) should find it and
+include its plugins in the plugin list. 
+
+* JACK client application
+
+You can also use Calf plugins as separate applications, connecting to other
+applications using JACK Audio Connection Kit (version 0.103 or newer is
+required). To run the client, type:
+
+        calfjackhost monosynth !
+
+(! means "connect", last "!" means "connect to output")
+
+Other examples:
+
+        calfjackhost monosynth ! vintagedelay ! flanger !
+
+(runs monosynth into vintagedelay and vintagedelay into flanger, then to
+output)
+
+        calfjackhost ! reverb !
+
+(takes signal from system:capture_1 and _2, puts it through reverb, and then
+sends to system:playback_1 and _2)
+
+You can also change client name or input/output port names with command-line
+options (type calfjackhost --help). Use qjackctl, patchage or jack_connect
+to connect the Calf JACK client to your sound card or other applications, if
+"!" is inadequate for any reason (if I didn't explain it properly, or if it
+doesn't provide the connectivity options needed).
+
+Keep in mind this project is in the early development phase. It is usable
+for certain purposes, but drop me a note if you need something.

plugins/ladspa_effect/calf/TODO

@@ -0,0 +1,40 @@
+1. More effects
+
+- auto-wah (might be integrated into filter)
+- envelope follower
+- better reverb (more features, use nested allpasses, use 1-pole 
+  1-zero allpass instead of fractional delays)
+- dynamics processing (Thor already did the compressor)
+- distortion?
+- windy rotary speakery stuff
+- filter: more types
+
+2. Some instruments
+
+- some virtual analogue thing (something larger than Monosynth)
+- FM (by reusing my MMX code, or something)
+
+3. DSP library
+
+- profiling framework
+- optimized code (the one I have now only pretends to be optimized :) )
+- underflow handling
+
+4. Wrappers
+
+- LADSPA: proper rdf (get clearance from drobilla ;) )
+- better jack host (controls etc)
+- BSE
+- buzztard
+- Linux VST
+- LV2
+  Message Context (for Organ)
+  EPP (the rest of them)
+  Mixing Controls
+
+5. Organization stuff (autotools etc)
+
+- correct compilation and installation of LADSPA plugins (current version is a hack!)
+- switch to -O3
+- get to work on 64-bit architectures
+- i18n (gettext or whatever)

plugins/ladspa_effect/calf/calf/audio_fx.h

@@ -0,0 +1,790 @@
+/* Calf DSP Library
+ * Reusable audio effect classes.
+ *
+ * Copyright (C) 2001-2007 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA  02110-1301  USA
+ */
+#ifndef __CALF_AUDIOFX_H
+#define __CALF_AUDIOFX_H
+
+#include <complex>
+#include <iostream>
+#include <calf/biquad.h>
+#include <calf/onepole.h>
+#include "primitives.h"
+#include "delay.h"
+#include "fixed_point.h"
+#include "inertia.h"
+
+namespace dsp {
+#if 0
+}; to keep editor happy
+#endif
+
+/**
+ * Audio effect base class. Not really useful until it gets more developed.
+ */
+class audio_effect
+{
+public:
+    virtual void setup(int sample_rate)=0;
+    virtual ~audio_effect() {}
+};
+
+class modulation_effect: public audio_effect
+{
+protected:
+    int sample_rate;
+    float rate, wet, dry, odsr;
+    gain_smoothing gs_wet, gs_dry;
+public:
+    fixed_point<unsigned int, 20> phase, dphase;
+    float get_rate() {
+        return rate;
+    }
+    void set_rate(float rate) {
+        this->rate = rate;
+        dphase = rate/sample_rate*4096;        
+    }
+    float get_wet() {
+        return wet;
+    }
+    void set_wet(float wet) {
+        this->wet = wet;
+        gs_wet.set_inertia(wet);
+    }
+    float get_dry() {
+        return dry;
+    }
+    void set_dry(float dry) {
+        this->dry = dry;
+        gs_dry.set_inertia(dry);
+    }
+    void reset_phase(float req_phase)
+    {
+        phase = req_phase * 4096.0;
+    }
+    void inc_phase(float req_phase)
+    {
+        phase += fixed_point<unsigned int, 20>(req_phase * 4096.0);
+    }
+    void setup(int sample_rate)
+    {
+        this->sample_rate = sample_rate;
+        this->odsr = 1.0 / sample_rate;
+        phase = 0;
+        set_rate(get_rate());
+    }
+};
+
+/**
+ * A monophonic phaser. If you want stereo, combine two :)
+ * Also, gave up on using template args for signal type.
+ */
+template<int MaxStages>
+class simple_phaser: public modulation_effect
+{
+protected:
+    float base_frq, mod_depth, fb;
+    float state;
+    int cnt, stages;
+    dsp::onepole<float, float> stage1;
+    float x1[MaxStages], y1[MaxStages];
+public:
+    simple_phaser()
+    {
+        set_base_frq(1000);
+        set_mod_depth(1000);
+        set_fb(0);
+        state = 0;
+        cnt = 0;
+        stages = 0;
+        set_stages(6);
+    }
+    float get_base_frq() {
+        return base_frq;
+    }
+    void set_base_frq(float _base_frq) {
+        base_frq = _base_frq;
+    }
+    int get_stages() {
+        return stages;
+    }
+    void set_stages(int _stages) {
+        if (_stages > stages)
+        {
+            for (int i = stages; i < _stages; i++)
+            {
+                x1[i] = x1[stages-1];
+                y1[i] = y1[stages-1];
+            }
+        }
+        stages = _stages;
+    }
+    float get_mod_depth() {
+        return mod_depth;
+    }
+    void set_mod_depth(float _mod_depth) {
+        mod_depth = _mod_depth;
+    }
+    float get_fb() {
+        return fb;
+    }
+    void set_fb(float fb) {
+        this->fb = fb;
+    }
+    virtual void setup(int sample_rate) {
+        modulation_effect::setup(sample_rate);
+        reset();
+    }
+    void reset()
+    {
+        cnt = 0;
+        state = 0;
+        phase.set(0);
+        for (int i = 0; i < MaxStages; i++)
+            x1[i] = y1[i] = 0;
+        control_step();
+    }
+    inline void control_step()
+    {
+        cnt = 0;
+        int v = phase.get() + 0x40000000;
+        int sign = v >> 31;
+        v ^= sign;
+        // triangle wave, range from 0 to INT_MAX
+        double vf = (double)((v >> 16) * (1.0 / 16384.0) - 1);
+        
+        float freq = base_frq * pow(2.0, vf * mod_depth / 1200.0);
+        freq = dsp::clip<float>(freq, 10.0, 0.49 * sample_rate);
+        stage1.set_ap_w(freq * (M_PI / 2.0) * odsr);
+        phase += dphase * 32;
+        for (int i = 0; i < stages; i++)
+        {
+            dsp::sanitize(x1[i]);
+            dsp::sanitize(y1[i]);
+        }
+        dsp::sanitize(state);
+    }
+    void process(float *buf_out, float *buf_in, int nsamples) {
+        for (int i=0; i<nsamples; i++) {
+            cnt++;
+            if (cnt == 32)
+                control_step();
+            float in = *buf_in++;
+            float fd = in + state * fb;
+            for (int j = 0; j < stages; j++)
+                fd = stage1.process_ap(fd, x1[j], y1[j]);
+            state = fd;
+            
+            float sdry = in * gs_dry.get();
+            float swet = fd * gs_wet.get();
+            *buf_out++ = sdry + swet;
+        }
+    }
+    float freq_gain(float freq, float sr)
+    {
+        typedef std::complex<double> cfloat;
+        freq *= 2.0 * M_PI / sr;
+        cfloat z = 1.0 / exp(cfloat(0.0, freq)); // z^-1
+        
+        cfloat p = cfloat(1.0);
+        cfloat stg = stage1.h_z(z);
+        
+        for (int i = 0; i < stages; i++)
+            p = p * stg;
+        
+        p = p / (cfloat(1.0) - cfloat(fb) * p);        
+        return std::abs(cfloat(gs_dry.get_last()) + cfloat(gs_wet.get_last()) * p);
+    }
+};
+
+/**
+ * Base class for chorus and flanger. Wouldn't be needed if it wasn't
+ * for odd behaviour of GCC when deriving templates from template
+ * base classes (not seeing fields from base classes!).
+ */
+class chorus_base: public modulation_effect
+{
+protected:
+    int min_delay_samples, mod_depth_samples;
+    float min_delay, mod_depth;
+    sine_table<int, 4096, 65536> sine;
+public:
+    float get_min_delay() {
+        return min_delay;
+    }
+    void set_min_delay(float min_delay) {
+        this->min_delay = min_delay;
+        this->min_delay_samples = (int)(min_delay * 65536.0 * sample_rate);
+    }
+    float get_mod_depth() {
+        return mod_depth;
+    }
+    void set_mod_depth(float mod_depth) {
+        this->mod_depth = mod_depth;
+        // 128 because it's then multiplied by (hopefully) a value of 32768..-32767
+        this->mod_depth_samples = (int)(mod_depth * 32.0 * sample_rate);
+    }
+};
+
+/**
+ * Single-tap chorus without feedback.
+ * Perhaps MaxDelay should be a bit longer!
+ */
+template<class T, int MaxDelay=512>
+class simple_chorus: public chorus_base
+{
+protected:
+    simple_delay<MaxDelay,T> delay;
+public:    
+    simple_chorus() {
+        rate = 0.63f;
+        dry = 0.5f;
+        wet = 0.5f;
+        min_delay = 0.005f;
+        mod_depth = 0.0025f;
+        setup(44100);
+    }
+    void reset() {
+        delay.reset();
+    }
+    virtual void setup(int sample_rate) {
+        modulation_effect::setup(sample_rate);
+        delay.reset();
+        set_min_delay(get_min_delay());
+        set_mod_depth(get_mod_depth());
+    }
+    template<class OutIter, class InIter>
+    void process(OutIter buf_out, InIter buf_in, int nsamples) {
+        int mds = min_delay_samples + mod_depth_samples * 1024 + 2*65536;
+        int mdepth = mod_depth_samples;
+        for (int i=0; i<nsamples; i++) {
+            phase += dphase;
+            unsigned int ipart = phase.ipart();
+            
+            float in = *buf_in++;
+            int lfo = phase.lerp_by_fract_int<int, 14, int>(sine.data[ipart], sine.data[ipart+1]);
+            int v = mds + (mdepth * lfo >> 6);
+            // if (!(i & 7)) printf("%d\n", v);
+            int ifv = v >> 16;
+            delay.put(in);
+            T fd; // signal from delay's output
+            delay.get_interp(fd, ifv, (v & 0xFFFF)*(1.0/65536.0));
+            T sdry = in * gs_dry.get();
+            T swet = fd * gs_wet.get();
+            *buf_out++ = sdry + swet;
+        }
+    }
+};
+
+/**
+ * Single-tap flanger (chorus plus feedback).
+ */
+template<class T, int MaxDelay=1024>
+class simple_flanger: public chorus_base
+{
+protected:
+    simple_delay<MaxDelay,T> delay;
+    float fb;
+    int last_delay_pos, last_actual_delay_pos;
+    int ramp_pos, ramp_delay_pos;
+public:
+    simple_flanger()
+    : fb(0) {}
+    void reset() {
+        delay.reset();
+        last_delay_pos = last_actual_delay_pos = ramp_delay_pos = 0;
+        ramp_pos = 1024;
+    }
+    virtual void setup(int sample_rate) {
+        this->sample_rate = sample_rate;
+        this->odsr = 1.0 / sample_rate;
+        delay.reset();
+        phase = 0;
+        set_rate(get_rate());
+        set_min_delay(get_min_delay());
+    }
+    float get_fb() {
+        return fb;
+    }
+    void set_fb(float fb) {
+        this->fb = fb;
+    }
+    template<class OutIter, class InIter>
+    void process(OutIter buf_out, InIter buf_in, int nsamples) {
+        if (!nsamples)
+            return;
+        int mds = this->min_delay_samples + this->mod_depth_samples * 1024 + 2 * 65536;
+        int mdepth = this->mod_depth_samples;
+        int delay_pos;
+        unsigned int ipart = this->phase.ipart();
+        int lfo = phase.lerp_by_fract_int<int, 14, int>(this->sine.data[ipart], this->sine.data[ipart+1]);
+        delay_pos = mds + (mdepth * lfo >> 6);
+        
+        if (delay_pos != last_delay_pos || ramp_pos < 1024)
+        {
+            if (delay_pos != last_delay_pos) {
+                // we need to ramp from what the delay tap length actually was, 
+                // not from old (ramp_delay_pos) or desired (delay_pos) tap length
+                ramp_delay_pos = last_actual_delay_pos;
+                ramp_pos = 0;
+            }
+            
+            int64_t dp = 0;
+            for (int i=0; i<nsamples; i++) {
+                float in = *buf_in++;
+                T fd; // signal from delay's output
+                dp = (((int64_t)ramp_delay_pos) * (1024 - ramp_pos) + ((int64_t)delay_pos) * ramp_pos) >> 10;
+                ramp_pos++;
+                if (ramp_pos > 1024) ramp_pos = 1024;
+                this->delay.get_interp(fd, dp >> 16, (dp & 0xFFFF)*(1.0/65536.0));
+                sanitize(fd);
+                T sdry = in * this->dry;
+                T swet = fd * this->wet;
+                *buf_out++ = sdry + swet;
+                this->delay.put(in+fb*fd);
+
+                this->phase += this->dphase;
+                ipart = this->phase.ipart();
+                lfo = phase.lerp_by_fract_int<int, 14, int>(this->sine.data[ipart], this->sine.data[ipart+1]);
+                delay_pos = mds + (mdepth * lfo >> 6);
+            }
+            last_actual_delay_pos = dp;
+        }
+        else {
+            for (int i=0; i<nsamples; i++) {
+                float in = *buf_in++;
+                T fd; // signal from delay's output
+                this->delay.get_interp(fd, delay_pos >> 16, (delay_pos & 0xFFFF)*(1.0/65536.0));
+                sanitize(fd);
+                T sdry = in * this->gs_dry.get();
+                T swet = fd * this->gs_wet.get();
+                *buf_out++ = sdry + swet;
+                this->delay.put(in+fb*fd);
+
+                this->phase += this->dphase;
+                ipart = this->phase.ipart();
+                lfo = phase.lerp_by_fract_int<int, 14, int>(this->sine.data[ipart], this->sine.data[ipart+1]);
+                delay_pos = mds + (mdepth * lfo >> 6);
+            }
+            last_actual_delay_pos = delay_pos;
+        }
+        last_delay_pos = delay_pos;
+    }
+    float freq_gain(float freq, float sr)
+    {
+        typedef std::complex<double> cfloat;
+        freq *= 2.0 * M_PI / sr;
+        cfloat z = 1.0 / exp(cfloat(0.0, freq)); // z^-1
+        
+        float ldp = last_delay_pos / 65536.0;
+        float fldp = floor(ldp);
+        cfloat zn = std::pow(z, fldp); // z^-N
+        cfloat zn1 = zn * z; // z^-(N+1)
+        // simulate a lerped comb filter - H(z) = 1 / (1 + fb * (lerp(z^-N, z^-(N+1), fracpos))), N = int(pos), fracpos = pos - int(pos)
+        cfloat delayed = zn + (zn1 - zn) * cfloat(ldp - fldp);
+        cfloat h = cfloat(delayed) / (cfloat(1.0) - cfloat(fb) * delayed);
+        // mix with dry signal
+        float v = std::abs(cfloat(gs_dry.get_last()) + cfloat(gs_wet.get_last()) * h);
+        return v;
+    }
+};
+
+/**
+ * A classic allpass loop reverb with modulated allpass filter.
+ * Just started implementing it, so there is no control over many
+ * parameters.
+ */
+template<class T>
+class reverb: public audio_effect
+{
+    simple_delay<2048, T> apL1, apL2, apL3, apL4, apL5, apL6;
+    simple_delay<2048, T> apR1, apR2, apR3, apR4, apR5, apR6;
+    fixed_point<unsigned int, 25> phase, dphase;
+    sine_table<int, 128, 10000> sine;
+    onepole<T> lp_left, lp_right;
+    T old_left, old_right;
+    int type;
+    float time, fb, cutoff, diffusion;
+    int tl[6], tr[6];
+    float ldec[6], rdec[6];
+    
+    int sr;
+public:
+    reverb()
+    {
+        phase = 0.0;
+        time = 1.0;
+        cutoff = 9000;
+        type = 2;
+        diffusion = 1.f;
+        setup(44100);
+    }
+    virtual void setup(int sample_rate) {
+        sr = sample_rate;
+        set_time(time);
+        set_cutoff(cutoff);
+        phase = 0.0;
+        dphase = 0.5*128/sr;
+        update_times();
+    }
+    void update_times()
+    {
+        switch(type)
+        {
+        case 0:
+            tl[0] =  397 << 16, tr[0] =  383 << 16;
+            tl[1] =  457 << 16, tr[1] =  429 << 16;
+            tl[2] =  549 << 16, tr[2] =  631 << 16;
+            tl[3] =  649 << 16, tr[3] =  756 << 16;
+            tl[4] =  773 << 16, tr[4] =  803 << 16;
+            tl[5] =  877 << 16, tr[5] =  901 << 16;
+            break;
+        case 1:
+            tl[0] =  697 << 16, tr[0] =  783 << 16;
+            tl[1] =  957 << 16, tr[1] =  929 << 16;
+            tl[2] =  649 << 16, tr[2] =  531 << 16;
+            tl[3] = 1049 << 16, tr[3] = 1177 << 16;
+            tl[4] =  473 << 16, tr[4] =  501 << 16;
+            tl[5] =  587 << 16, tr[5] =  681 << 16;
+            break;
+        case 2:
+        default:
+            tl[0] =  697 << 16, tr[0] =  783 << 16;
+            tl[1] =  957 << 16, tr[1] =  929 << 16;
+            tl[2] =  649 << 16, tr[2] =  531 << 16;
+            tl[3] = 1249 << 16, tr[3] = 1377 << 16;
+            tl[4] = 1573 << 16, tr[4] = 1671 << 16;
+            tl[5] = 1877 << 16, tr[5] = 1781 << 16;
+            break;
+        case 3:
+            tl[0] = 1097 << 16, tr[0] = 1087 << 16;
+            tl[1] = 1057 << 16, tr[1] = 1031 << 16;
+            tl[2] = 1049 << 16, tr[2] = 1039 << 16;
+            tl[3] = 1083 << 16, tr[3] = 1055 << 16;
+            tl[4] = 1075 << 16, tr[4] = 1099 << 16;
+            tl[5] = 1003 << 16, tr[5] = 1073 << 16;
+            break;
+        case 4:
+            tl[0] =  197 << 16, tr[0] =  133 << 16;
+            tl[1] =  357 << 16, tr[1] =  229 << 16;
+            tl[2] =  549 << 16, tr[2] =  431 << 16;
+            tl[3] =  949 << 16, tr[3] = 1277 << 16;
+            tl[4] = 1173 << 16, tr[4] = 1671 << 16;
+            tl[5] = 1477 << 16, tr[5] = 1881 << 16;
+            break;
+        case 5:
+            tl[0] =  197 << 16, tr[0] =  133 << 16;
+            tl[1] =  257 << 16, tr[1] =  179 << 16;
+            tl[2] =  549 << 16, tr[2] =  431 << 16;
+            tl[3] =  619 << 16, tr[3] =  497 << 16;
+            tl[4] = 1173 << 16, tr[4] = 1371 << 16;
+            tl[5] = 1577 << 16, tr[5] = 1881 << 16;
+            break;
+        }
+        
+        float fDec=1000 + 2400.f * diffusion;
+        for (int i = 0 ; i < 6; i++) {
+            ldec[i]=exp(-float(tl[i] >> 16) / fDec), 
+            rdec[i]=exp(-float(tr[i] >> 16) / fDec);
+        }
+    }
+    float get_time() {
+        return time;
+    }
+    void set_time(float time) {
+        this->time = time;
+        // fb = pow(1.0f/4096.0f, (float)(1700/(time*sr)));
+        fb = 1.0 - 0.3 / (time * sr / 44100.0);
+    }
+    float get_type() {
+        return type;
+    }
+    void set_type(int type) {
+        this->type = type;
+        update_times();
+    }
+    float get_diffusion() {
+        return diffusion;
+    }
+    void set_diffusion(float diffusion) {
+        this->diffusion = diffusion;
+        update_times();
+    }
+    void set_type_and_diffusion(int type, float diffusion) {
+        this->type = type;
+        this->diffusion = diffusion;
+        update_times();
+    }
+    float get_fb()
+    {
+        return this->fb;
+    }
+    void set_fb(float fb)
+    {
+        this->fb = fb;
+    }
+    float get_cutoff() {
+        return cutoff;
+    }
+    void set_cutoff(float cutoff) {
+        this->cutoff = cutoff;
+        lp_left.set_lp(cutoff,sr);
+        lp_right.set_lp(cutoff,sr);
+    }
+    void reset()
+    {
+        apL1.reset();apR1.reset();
+        apL2.reset();apR2.reset();
+        apL3.reset();apR3.reset();
+        apL4.reset();apR4.reset();
+        apL5.reset();apR5.reset();
+        apL6.reset();apR6.reset();
+        lp_left.reset();lp_right.reset();
+        old_left = 0; old_right = 0;
+    }
+    void process(T &left, T &right)
+    {
+        unsigned int ipart = phase.ipart();
+        
+        // the interpolated LFO might be an overkill here
+        int lfo = phase.lerp_by_fract_int<int, 14, int>(sine.data[ipart], sine.data[ipart+1]) >> 2;
+        phase += dphase;
+        
+        left += old_right;
+        left = apL1.process_allpass_comb_lerp16(left, tl[0] - 45*lfo, ldec[0]);
+        left = apL2.process_allpass_comb_lerp16(left, tl[1] + 47*lfo, ldec[1]);
+        float out_left = left;
+        left = apL3.process_allpass_comb_lerp16(left, tl[2] + 54*lfo, ldec[2]);
+        left = apL4.process_allpass_comb_lerp16(left, tl[3] - 69*lfo, ldec[3]);
+        left = apL5.process_allpass_comb_lerp16(left, tl[4] + 69*lfo, ldec[4]);
+        left = apL6.process_allpass_comb_lerp16(left, tl[5] - 46*lfo, ldec[5]);
+        old_left = lp_left.process(left * fb);
+        sanitize(old_left);
+
+        right += old_left;
+        right = apR1.process_allpass_comb_lerp16(right, tr[0] - 45*lfo, rdec[0]);
+        right = apR2.process_allpass_comb_lerp16(right, tr[1] + 47*lfo, rdec[1]);
+        float out_right = right;
+        right = apR3.process_allpass_comb_lerp16(right, tr[2] + 54*lfo, rdec[2]);
+        right = apR4.process_allpass_comb_lerp16(right, tr[3] - 69*lfo, rdec[3]);
+        right = apR5.process_allpass_comb_lerp16(right, tr[4] + 69*lfo, rdec[4]);
+        right = apR6.process_allpass_comb_lerp16(right, tr[5] - 46*lfo, rdec[5]);
+        old_right = lp_right.process(right * fb);
+        sanitize(old_right);
+        
+        left = out_left, right = out_right;
+    }
+    void extra_sanitize()
+    {
+        lp_left.sanitize();
+        lp_right.sanitize();
+    }
+};
+
+class filter_module_iface
+{
+public:
+    virtual void  calculate_filter(float freq, float q, int mode, float gain = 1.0) = 0;
+    virtual void  filter_activate() = 0;
+    virtual void  sanitize() = 0;
+    virtual int   process_channel(uint16_t channel_no, float *in, float *out, uint32_t numsamples, int inmask) = 0;
+    virtual float freq_gain(int subindex, float freq, float srate) = 0;
+
+    virtual ~filter_module_iface() {}
+};
+
+
+class biquad_filter_module: public filter_module_iface
+{
+private:
+    dsp::biquad_d1<float> left[3], right[3];
+    int order;
+    
+public:    
+    uint32_t srate;
+    
+    enum { mode_12db_lp = 0, mode_24db_lp = 1, mode_36db_lp = 2, 
+           mode_12db_hp = 3, mode_24db_hp = 4, mode_36db_hp = 5,
+           mode_6db_bp  = 6, mode_12db_bp = 7, mode_18db_bp = 8,
+           mode_6db_br  = 9, mode_12db_br = 10, mode_18db_br = 11,
+           mode_count
+    };
+    
+public:
+    biquad_filter_module() : order(0) {}
+    
+    void calculate_filter(float freq, float q, int mode, float gain = 1.0)
+    {
+        if (mode <= mode_36db_lp) {
+            order = mode + 1;
+            left[0].set_lp_rbj(freq, pow(q, 1.0 / order), srate, gain);
+        } else if ( mode_12db_hp <= mode && mode <= mode_36db_hp ) {
+            order = mode - mode_12db_hp + 1;
+            left[0].set_hp_rbj(freq, pow(q, 1.0 / order), srate, gain);
+        } else if ( mode_6db_bp <= mode && mode <= mode_18db_bp ) {
+            order = mode - mode_6db_bp + 1;
+            left[0].set_bp_rbj(freq, pow(q, 1.0 / order), srate, gain);
+        } else { // mode_6db_br <= mode <= mode_18db_br
+            order = mode - mode_6db_br + 1;
+            left[0].set_br_rbj(freq, order * 0.1 * q, srate, gain);
+        }
+        
+        right[0].copy_coeffs(left[0]);
+        for (int i = 1; i < order; i++) {
+            left[i].copy_coeffs(left[0]);
+            right[i].copy_coeffs(left[0]);
+        }
+    }
+    
+    void filter_activate()
+    {
+        for (int i=0; i < order; i++) {
+            left[i].reset();
+            right[i].reset();
+        }
+    }
+    
+    void  sanitize()
+    {
+        for (int i=0; i < order; i++) {
+            left[i].sanitize();
+            right[i].sanitize();
+        }
+    }
+    
+    inline int process_channel(uint16_t channel_no, float *in, float *out, uint32_t numsamples, int inmask) {
+        dsp::biquad_d1<float> *filter;
+        switch (channel_no) {
+        case 0:
+            filter = left;
+            break;
+            
+        case 1:
+            filter = right;
+            break;
+        
+        default:
+            assert(false);
+            return 0;
+        }
+        
+        if (inmask) {
+            switch(order) {
+                case 1:
+                    for (uint32_t i = 0; i < numsamples; i++)
+                        out[i] = filter[0].process(in[i]);
+                    break;
+                case 2:
+                    for (uint32_t i = 0; i < numsamples; i++)
+                        out[i] = filter[1].process(filter[0].process(in[i]));
+                    break;
+                case 3:
+                    for (uint32_t i = 0; i < numsamples; i++)
+                        out[i] = filter[2].process(filter[1].process(filter[0].process(in[i])));
+                    break;
+            }
+        } else {
+            if (filter[order - 1].empty())
+                return 0;
+            switch(order) {
+                case 1:
+                    for (uint32_t i = 0; i < numsamples; i++)
+                        out[i] = filter[0].process_zeroin();
+                    break;
+                case 2:
+                    if (filter[0].empty())
+                        for (uint32_t i = 0; i < numsamples; i++)
+                            out[i] = filter[1].process_zeroin();
+                    else
+                        for (uint32_t i = 0; i < numsamples; i++)
+                            out[i] = filter[1].process(filter[0].process_zeroin());
+                    break;
+                case 3:
+                    if (filter[1].empty())
+                        for (uint32_t i = 0; i < numsamples; i++)
+                            out[i] = filter[2].process_zeroin();
+                    else
+                        for (uint32_t i = 0; i < numsamples; i++)
+                            out[i] = filter[2].process(filter[1].process(filter[0].process_zeroin()));
+                    break;
+            }
+        }
+        for (int i = 0; i < order; i++)
+            filter[i].sanitize();
+        return filter[order - 1].empty() ? 0 : inmask;
+    }
+    
+    float freq_gain(int subindex, float freq, float srate)
+    {
+        float level = 1.0;
+        for (int j = 0; j < order; j++)
+            level *= left[j].freq_gain(freq, srate);
+        return level;
+    }
+};
+
+class two_band_eq
+{
+private:
+    dsp::onepole<float> lowcut, highcut;
+    float low_gain, high_gain;
+
+public:
+    void reset()
+    {
+        lowcut.reset();
+        highcut.reset();
+    }
+    
+    inline float process(float v)
+    {
+        v = dsp::lerp(lowcut.process_hp(v), v, low_gain);
+        v = dsp::lerp(highcut.process_lp(v), v, high_gain);
+        return v;
+    }
+    
+    inline void copy_coeffs(const two_band_eq &src)
+    {
+        lowcut.copy_coeffs(src.lowcut);
+        highcut.copy_coeffs(src.highcut);
+        low_gain = src.low_gain;
+        high_gain = src.high_gain;
+    }
+    
+    void sanitize()
+    {
+        lowcut.sanitize();
+        highcut.sanitize();
+    }
+    
+    void set(float _low_freq, float _low_gain, float _high_freq, float _high_gain, float sr)
+    {
+        lowcut.set_hp(_low_freq, sr);
+        highcut.set_lp(_high_freq, sr);
+        low_gain = _low_gain;
+        high_gain = _high_gain;
+    }
+};
+
+#if 0
+{ to keep editor happy
+#endif
+}
+
+#endif

plugins/ladspa_effect/calf/calf/biquad.h

@@ -0,0 +1,580 @@
+/* Calf DSP Library
+ * Biquad filters
+ * Copyright (C) 2001-2007 Krzysztof Foltman
+ *
+ * Most of code in this file is based on freely
+ * available other work of other people (filter equations).
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA  02110-1301  USA
+ */
+#ifndef __CALF_BIQUAD_H
+#define __CALF_BIQUAD_H
+
+#include <complex>
+#include "primitives.h"
+
+namespace dsp {
+
+/**
+ * Coefficients for two-pole two-zero filter, for floating point values,
+ * plus a bunch of functions to set them to typical values.
+ * 
+ * Coefficient calculation is based on famous Robert Bristow-Johnson's equations,
+ * except where it's not. 
+ * The coefficient calculation is NOT mine, the only exception is the lossy 
+ * optimization in Zoelzer and rbj HP filter code.
+ * 
+ * See http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt for reference.
+ * 
+ * don't use this for integers because it won't work
+ */
+template<class Coeff = float>
+class biquad_coeffs
+{
+public:
+    // filter coefficients
+    Coeff a0, a1, a2, b1, b2;
+    typedef std::complex<double> cfloat;
+
+    biquad_coeffs()
+    {
+        set_null();
+    }
+    
+    inline void set_null()
+    {
+        a0 = 1.0;
+        b1 = b2 = a1 = a2 = 0.f;
+    }
+    
+    /** Lowpass filter based on Robert Bristow-Johnson's equations
+     * Perhaps every synth code that doesn't use SVF uses these
+     * equations :)
+     * @param fc     resonant frequency
+     * @param q      resonance (gain at fc)
+     * @param sr     sample rate
+     * @param gain   amplification (gain at 0Hz)
+     */
+    inline void set_lp_rbj(float fc, float q, float sr, float gain = 1.0)
+    {
+        float omega=(float)(2*M_PI*fc/sr);
+        float sn=sin(omega);
+        float cs=cos(omega);
+        float alpha=(float)(sn/(2*q));
+        float inv=(float)(1.0/(1.0+alpha));
+
+        a2 = a0 =  (float)(gain*inv*(1 - cs)*0.5f);
+        a1 =  a0 + a0;
+        b1 =  (float)(-2*cs*inv);
+        b2 =  (float)((1 - alpha)*inv);
+    }
+
+    // different lowpass filter, based on Zoelzer's equations, modified by
+    // me (kfoltman) to use polynomials to approximate tangent function
+    // not very accurate, but perhaps good enough for synth work :)
+    // odsr is "one divided by samplerate"
+    // from how it looks, it perhaps uses bilinear transform - but who knows :)
+    inline void set_lp_zoelzer(float fc, float q, float odsr, float gain=1.0)
+    {
+        Coeff omega=(Coeff)(M_PI*fc*odsr);
+        Coeff omega2=omega*omega;
+        Coeff K=omega*(1+omega2*omega2*Coeff(1.0/1.45));
+        Coeff KK=K*K;
+        Coeff QK=q*(KK+1.f);
+        Coeff iQK=1.0f/(QK+K);
+        Coeff inv=q*iQK;
+        b2 =  (Coeff)(iQK*(QK-K));
+        b1 =  (Coeff)(2.f*(KK-1.f)*inv);
+        a2 = a0 =  (Coeff)(inv*gain*KK);
+        a1 =  a0 + a0;
+    }
+
+    /** Highpass filter based on Robert Bristow-Johnson's equations
+     * @param fc     resonant frequency
+     * @param q      resonance (gain at fc)
+     * @param sr     sample rate
+     * @param gain   amplification (gain at sr/2)
+     */
+    inline void set_hp_rbj(float fc, float q, float esr, float gain=1.0)
+    {
+        Coeff omega=(float)(2*M_PI*fc/esr);
+        Coeff sn=sin(omega);
+        Coeff cs=cos(omega);
+        Coeff alpha=(float)(sn/(2*q));
+
+        float inv=(float)(1.0/(1.0+alpha));
+
+        a0 =  (Coeff)(gain*inv*(1 + cs)/2);
+        a1 =  -2.f * a0;
+        a2 =  a0;
+        b1 =  (Coeff)(-2*cs*inv);
+        b2 =  (Coeff)((1 - alpha)*inv);
+    }
+
+    // this replaces sin/cos with polynomial approximation
+    inline void set_hp_rbj_optimized(float fc, float q, float esr, float gain=1.0)
+    {
+        Coeff omega=(float)(2*M_PI*fc/esr);
+        Coeff sn=omega+omega*omega*omega*(1.0/6.0)+omega*omega*omega*omega*omega*(1.0/120);
+        Coeff cs=1-omega*omega*(1.0/2.0)+omega*omega*omega*omega*(1.0/24);
+        Coeff alpha=(float)(sn/(2*q));
+
+        float inv=(float)(1.0/(1.0+alpha));
+
+        a0 =  (Coeff)(gain*inv*(1 + cs)*(1.0/2.0));
+        a1 =  -2.f * a0;
+        a2 =  a0;
+        b1 =  (Coeff)(-2*cs*inv);
+        b2 =  (Coeff)((1 - alpha)*inv);
+    }
+    
+    /** Bandpass filter based on Robert Bristow-Johnson's equations (normalized to 1.0 at center frequency)
+     * @param fc     center frequency (gain at fc = 1.0)
+     * @param q      =~ fc/bandwidth (not quite, but close)  - 1/Q = 2*sinh(ln(2)/2*BW*w0/sin(w0))
+     * @param sr     sample rate
+     * @param gain   amplification (gain at sr/2)
+     */
+    inline void set_bp_rbj(double fc, double q, double esr, double gain=1.0)
+    {
+        float omega=(float)(2*M_PI*fc/esr);
+        float sn=sin(omega);
+        float cs=cos(omega);
+        float alpha=(float)(sn/(2*q));
+
+        float inv=(float)(1.0/(1.0+alpha));
+
+        a0 =  (float)(gain*inv*alpha);
+        a1 =  0.f;
+        a2 =  (float)(-gain*inv*alpha);
+        b1 =  (float)(-2*cs*inv);
+        b2 =  (float)((1 - alpha)*inv);
+    }
+    
+    // rbj's bandreject
+    inline void set_br_rbj(double fc, double q, double esr, double gain=1.0)
+    {
+        float omega=(float)(2*M_PI*fc/esr);
+        float sn=sin(omega);
+        float cs=cos(omega);
+        float alpha=(float)(sn/(2*q));
+
+        float inv=(float)(1.0/(1.0+alpha));
+
+        a0 =  (Coeff)(gain*inv);
+        a1 =  (Coeff)(-gain*inv*2*cs);
+        a2 =  (Coeff)(gain*inv);
+        b1 =  (Coeff)(-2*cs*inv);
+        b2 =  (Coeff)((1 - alpha)*inv);
+    }
+    // this is mine (and, I guess, it sucks/doesn't work)
+    void set_allpass(float freq, float pole_r, float sr)
+    {
+        float a=prewarp(freq, sr);
+        float q=pole_r;
+        set_bilinear(a*a+q*q, -2.0f*a, 1, a*a+q*q, 2.0f*a, 1);
+    }
+    /// prewarping for bilinear transform, maps given digital frequency to analog counterpart for analog filter design
+    static inline float prewarp(float freq, float sr)
+    {
+        if (freq>sr*0.49) freq=(float)(sr*0.49);
+        return (float)(tan(M_PI*freq/sr));
+    }
+    /// convert analog angular frequency value to digital
+    static inline float unwarp(float omega, float sr)
+    {
+        float T = 1.0 / sr;
+        return (2 / T) * atan(omega * T / 2);
+    }
+    /// convert analog filter time constant to digital counterpart
+    static inline float unwarpf(float t, float sr)
+    {
+        // this is most likely broken and works by pure accident!
+        float omega = 1.0 / t;
+        omega = unwarp(omega, sr);
+        // I really don't know why does it have to be M_PI and not 2 * M_PI!
+        float f = M_PI / omega;
+        return f / sr;
+    }
+    /// set digital filter parameters based on given analog filter parameters
+    void set_bilinear(float aa0, float aa1, float aa2, float ab0, float ab1, float ab2)
+    {
+        float q=(float)(1.0/(ab0+ab1+ab2));
+        a0 = (aa0+aa1+aa2)*q;
+        a1 = 2*(aa0-aa2)*q;
+        a2 = (aa0-aa1+aa2)*q;
+        b1 = 2*(ab0-ab2)*q;
+        b2 = (ab0-ab1+ab2)*q;
+    }
+    
+    /// RBJ peaking EQ
+    /// @param freq   peak frequency
+    /// @param q      q (correlated to freq/bandwidth, @see set_bp_rbj)
+    /// @param peak   peak gain (1.0 means no peak, >1.0 means a peak, less than 1.0 is a dip)
+    inline void set_peakeq_rbj(float freq, float q, float peak, float sr)
+    {
+        float A = sqrt(peak);
+        float w0 = freq * 2 * M_PI * (1.0 / sr);
+        float alpha = sin(w0) / (2 * q);
+        float ib0 = 1.0 / (1 + alpha/A);
+        a1 = b1 = -2*cos(w0) * ib0;
+        a0 = ib0 * (1 + alpha*A);
+        a2 = ib0 * (1 - alpha*A);
+        b2 = ib0 * (1 - alpha/A);
+    }
+    
+    /// RBJ low shelf EQ - amplitication of 'peak' at 0 Hz and of 1.0 (0dB) at sr/2 Hz
+    /// @param freq   corner frequency (gain at freq is sqrt(peak))
+    /// @param q      q (relates bandwidth and peak frequency), the higher q, the louder the resonant peak (situated below fc) is
+    /// @param peak   shelf gain (1.0 means no peak, >1.0 means a peak, less than 1.0 is a dip)
+    inline void set_lowshelf_rbj(float freq, float q, float peak, float sr)
+    {
+        float A = sqrt(peak);
+        float w0 = freq * 2 * M_PI * (1.0 / sr);
+        float alpha = sin(w0) / (2 * q);
+        float cw0 = cos(w0);
+        float tmp = 2 * sqrt(A) * alpha;
+        float b0 = 0.f, ib0 = 0.f;
+        
+        a0 =    A*( (A+1) - (A-1)*cw0 + tmp);
+        a1 =  2*A*( (A-1) - (A+1)*cw0);
+        a2 =    A*( (A+1) - (A-1)*cw0 - tmp);
+        b0 =        (A+1) + (A-1)*cw0 + tmp;
+        b1 =   -2*( (A-1) + (A+1)*cw0);
+        b2 =        (A+1) + (A-1)*cw0 - tmp;
+        
+        ib0 = 1.0 / b0;
+        b1 *= ib0;
+        b2 *= ib0;
+        a0 *= ib0;
+        a1 *= ib0;
+        a2 *= ib0;
+    }
+    
+    /// RBJ high shelf EQ - amplitication of 0dB at 0 Hz and of peak at sr/2 Hz
+    /// @param freq   corner frequency (gain at freq is sqrt(peak))
+    /// @param q      q (relates bandwidth and peak frequency), the higher q, the louder the resonant peak (situated above fc) is
+    /// @param peak   shelf gain (1.0 means no peak, >1.0 means a peak, less than 1.0 is a dip)
+    inline void set_highshelf_rbj(float freq, float q, float peak, float sr)
+    {
+        float A = sqrt(peak);
+        float w0 = freq * 2 * M_PI * (1.0 / sr);
+        float alpha = sin(w0) / (2 * q);
+        float cw0 = cos(w0);
+        float tmp = 2 * sqrt(A) * alpha;
+        float b0 = 0.f, ib0 = 0.f;
+        
+        a0 =    A*( (A+1) + (A-1)*cw0 + tmp);
+        a1 = -2*A*( (A-1) + (A+1)*cw0);
+        a2 =    A*( (A+1) + (A-1)*cw0 - tmp);
+        b0 =        (A+1) - (A-1)*cw0 + tmp;
+        b1 =    2*( (A-1) - (A+1)*cw0);
+        b2 =        (A+1) - (A-1)*cw0 - tmp;
+        
+        ib0 = 1.0 / b0;
+        b1 *= ib0;
+        b2 *= ib0;
+        a0 *= ib0;
+        a1 *= ib0;
+        a2 *= ib0;
+    }
+    
+    /// copy coefficients from another biquad
+    template<class U>
+    inline void copy_coeffs(const biquad_coeffs<U> &src)
+    {
+        a0 = src.a0;
+        a1 = src.a1;
+        a2 = src.a2;
+        b1 = src.b1;
+        b2 = src.b2;
+    }
+    
+    /// Return the filter's gain at frequency freq
+    /// @param freq   Frequency to look up
+    /// @param sr     Filter sample rate (used to convert frequency to angular frequency)
+    float freq_gain(float freq, float sr)
+    {
+        typedef std::complex<double> cfloat;
+        freq *= 2.0 * M_PI / sr;
+        cfloat z = 1.0 / exp(cfloat(0.0, freq));
+        
+        return std::abs(h_z(z));
+    }
+    
+    /// Return H(z) the filter's gain at frequency freq
+    /// @param z   Z variable (e^jw)
+    cfloat h_z(const cfloat &z)
+    {
+        
+        return (cfloat(a0) + double(a1) * z + double(a2) * z*z) / (cfloat(1.0) + double(b1) * z + double(b2) * z*z);
+    }
+    
+};
+
+/**
+ * Two-pole two-zero filter, for floating point values.
+ * Uses "traditional" Direct I form (separate FIR and IIR halves).
+ * don't use this for integers because it won't work
+ */
+template<class Coeff = float, class T = float>
+struct biquad_d1: public biquad_coeffs<Coeff>
+{
+    using biquad_coeffs<Coeff>::a0;
+    using biquad_coeffs<Coeff>::a1;
+    using biquad_coeffs<Coeff>::a2;
+    using biquad_coeffs<Coeff>::b1;
+    using biquad_coeffs<Coeff>::b2;
+    /// input[n-1]
+    T x1; 
+    /// input[n-2]
+    T x2; 
+    /// output[n-1]
+    T y1; 
+    /// output[n-2]
+    T y2; 
+    /// Constructor (initializes state to all zeros)
+    biquad_d1()
+    {
+        reset();
+    }
+    /// direct I form with four state variables
+    inline T process(T in)
+    {
+        T out = in * a0 + x1 * a1 + x2 * a2 - y1 * b1 - y2 * b2;
+        x2 = x1;
+        y2 = y1;
+        x1 = in;
+        y1 = out;
+        return out;
+    }
+    
+    /// direct I form with zero input
+    inline T process_zeroin()
+    {
+        T out = - y1 * b1 - y2 * b2;
+        y2 = y1;
+        y1 = out;
+        return out;
+    }
+    
+    /// simplified version for lowpass case with two zeros at -1
+    inline T process_lp(T in)
+    {
+        T out = a0*(in + x1 + x1 + x2) - y1 * b1 - y2 * b2;
+        x2 = x1;
+        y2 = y1;
+        x1 = in;
+        y1 = out;
+        return out;
+    }
+    /// Sanitize (set to 0 if potentially denormal) filter state
+    inline void sanitize() 
+    {
+        dsp::sanitize(x1);
+        dsp::sanitize(y1);
+        dsp::sanitize(x2);
+        dsp::sanitize(y2);
+    }
+    /// Reset state variables
+    inline void reset()
+    {
+        dsp::zero(x1);
+        dsp::zero(y1);
+        dsp::zero(x2);
+        dsp::zero(y2);
+    }
+    inline bool empty() {
+        return (y1 == 0.f && y2 == 0.f);
+    }
+    
+};
+    
+/**
+ * Two-pole two-zero filter, for floating point values.
+ * Uses slightly faster Direct II form (combined FIR and IIR halves).
+ * However, when used with wildly varying coefficients, it may 
+ * make more zipper noise than Direct I form, so it's better to
+ * use it when filter coefficients are not changed mid-stream.
+ */
+template<class Coeff = float, class T = float>
+struct biquad_d2: public biquad_coeffs<Coeff>
+{
+    using biquad_coeffs<Coeff>::a0;
+    using biquad_coeffs<Coeff>::a1;
+    using biquad_coeffs<Coeff>::a2;
+    using biquad_coeffs<Coeff>::b1;
+    using biquad_coeffs<Coeff>::b2;
+    /// state[n-1]
+    float w1; 
+    /// state[n-2]
+    float w2; 
+    /// Constructor (initializes state to all zeros)
+    biquad_d2()
+    {
+        reset();
+    }
+    /// direct II form with two state variables
+    inline T process(T in)
+    {
+        T tmp = in - w1 * b1 - w2 * b2;
+        T out = tmp * a0 + w1 * a1 + w2 * a2;
+        w2 = w1;
+        w1 = tmp;
+        return out;
+    }
+    
+    // direct II form with two state variables, lowpass version
+    // interesting fact: this is actually slower than the general version!
+    inline T process_lp(T in)
+    {
+        T tmp = in - w1 * b1 - w2 * b2;
+        T out = (tmp  + w2 + w1* 2) * a0;
+        w2 = w1;
+        w1 = tmp;
+        return out;
+    }
+
+    /// Is the filter state completely silent? (i.e. set to 0 by sanitize function)
+    inline bool empty() {
+        return (w1 == 0.f && w2 == 0.f);
+    }
+    
+    
+    /// Sanitize (set to 0 if potentially denormal) filter state
+    inline void sanitize() 
+    {
+        dsp::sanitize(w1);
+        dsp::sanitize(w2);
+    }
+    
+    /// Reset state variables
+    inline void reset()
+    {
+        dsp::zero(w1);
+        dsp::zero(w2);
+    }
+};
+
+/**
+ * Two-pole two-zero filter, for floating point values.
+ * Uses "traditional" Direct I form (separate FIR and IIR halves).
+ * don't use this for integers because it won't work
+ */
+template<class Coeff = float, class T = float>
+struct biquad_d1_lerp: public biquad_coeffs<Coeff>
+{
+    using biquad_coeffs<Coeff>::a0;
+    using biquad_coeffs<Coeff>::a1;
+    using biquad_coeffs<Coeff>::a2;
+    using biquad_coeffs<Coeff>::b1;
+    using biquad_coeffs<Coeff>::b2;
+    Coeff a0cur, a1cur, a2cur, b1cur, b2cur;
+    Coeff a0delta, a1delta, a2delta, b1delta, b2delta;
+    /// input[n-1]
+    T x1; 
+    /// input[n-2]
+    T x2; 
+    /// output[n-1]
+    T y1; 
+    /// output[n-2]
+    T y2; 
+    /// Constructor (initializes state to all zeros)
+    biquad_d1_lerp()
+    {
+        reset();
+    }
+    #define _DO_COEFF(coeff) coeff##delta = (coeff - coeff##cur) * (frac)
+    void big_step(Coeff frac)
+    {
+        _DO_COEFF(a0);
+        _DO_COEFF(a1);
+        _DO_COEFF(a2);
+        _DO_COEFF(b1);
+        _DO_COEFF(b2);
+    }
+    #undef _DO_COEFF
+    /// direct I form with four state variables
+    inline T process(T in)
+    {
+        T out = in * a0cur + x1 * a1cur + x2 * a2cur - y1 * b1cur - y2 * b2cur;
+        x2 = x1;
+        y2 = y1;
+        x1 = in;
+        y1 = out;
+        a0cur += a0delta;
+        a1cur += a1delta;
+        a2cur += a2delta;
+        b1cur += b1delta;
+        b2cur += b2delta;
+        return out;
+    }
+    
+    /// direct I form with zero input
+    inline T process_zeroin()
+    {
+        T out = - y1 * b1 - y2 * b2;
+        y2 = y1;
+        y1 = out;
+        b1cur += b1delta;
+        b2cur += b2delta;
+        return out;
+    }
+    
+    /// simplified version for lowpass case with two zeros at -1
+    inline T process_lp(T in)
+    {
+        T out = a0*(in + x1 + x1 + x2) - y1 * b1 - y2 * b2;
+        x2 = x1;
+        y2 = y1;
+        x1 = in;
+        y1 = out;
+        return out;
+    }
+    /// Sanitize (set to 0 if potentially denormal) filter state
+    inline void sanitize() 
+    {
+        dsp::sanitize(x1);
+        dsp::sanitize(y1);
+        dsp::sanitize(x2);
+        dsp::sanitize(y2);
+        dsp::sanitize(a0cur);
+        dsp::sanitize(a1cur);
+        dsp::sanitize(a2cur);
+        dsp::sanitize(b1cur);
+        dsp::sanitize(b2cur);
+    }
+    /// Reset state variables
+    inline void reset()
+    {
+        dsp::zero(x1);
+        dsp::zero(y1);
+        dsp::zero(x2);
+        dsp::zero(y2);
+        dsp::zero(a0cur);
+        dsp::zero(a1cur);
+        dsp::zero(a2cur);
+        dsp::zero(b1cur);
+        dsp::zero(b2cur);
+    }
+    inline bool empty() {
+        return (y1 == 0.f && y2 == 0.f);
+    }
+    
+};
+    
+};
+
+#endif

plugins/ladspa_effect/calf/calf/buffer.h

@@ -0,0 +1,229 @@
+/* Calf DSP Library
+ * Buffer abstractions.
+ *
+ * Copyright (C) 2007 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA  02110-1301  USA
+ */
+#ifndef __BUFFER_H
+#define __BUFFER_H
+
+namespace dsp {
+
+/// decrease by N if >= N (useful for circular buffers)
+template<int N> inline int wrap_around(int a) {
+    return (a >= N) ? a - N : a;
+}
+
+// provide fast specializations for powers of 2
+template<> inline int wrap_around<2>(int a) { return a & 1; }
+template<> inline int wrap_around<4>(int a) { return a & 3; }
+template<> inline int wrap_around<8>(int a) { return a & 7; }
+template<> inline int wrap_around<16>(int a) { return a & 15; }
+template<> inline int wrap_around<32>(int a) { return a & 31; }
+template<> inline int wrap_around<64>(int a) { return a & 63; }
+template<> inline int wrap_around<128>(int a) { return a & 127; }
+template<> inline int wrap_around<256>(int a) { return a & 255; }
+template<> inline int wrap_around<512>(int a) { return a & 511; }
+template<> inline int wrap_around<1024>(int a) { return a & 1023; }
+template<> inline int wrap_around<2048>(int a) { return a & 2047; }
+template<> inline int wrap_around<4096>(int a) { return a & 4095; }
+template<> inline int wrap_around<8192>(int a) { return a & 8191; }
+template<> inline int wrap_around<16384>(int a) { return a & 16383; }
+template<> inline int wrap_around<32768>(int a) { return a & 32767; }
+template<> inline int wrap_around<65536>(int a) { return a & 65535; }
+
+template<class Buf, class T>
+void fill(Buf &buf, T value) {
+    T* data = buf.data();
+    int size = buf.size();
+    for (int i=0; i<size; i++)
+        *data++ = value;
+}
+
+template<class T>
+void fill(T *data, int size, T value) {
+    for (int i=0; i<size; i++)
+        *data++ = value;
+}
+
+template<class T, class U>
+void copy(T *dest, U *src, int size, T scale = 1, T add = 0) {
+    for (int i=0; i<size; i++) 
+        *dest++ = (*src++) * scale + add;
+}
+
+template<class T>
+struct sample_traits {
+    enum {
+        channels = 1,
+        bps = sizeof(T)*8
+    };
+};
+
+template<class T>
+struct sample_traits<stereo_sample<T> > {
+    enum {
+        channels = 2,
+        bps = sizeof(T)*8
+    };
+};
+
+template<int N, class T = float>
+class fixed_size_buffer {
+public:
+    typedef T data_type;
+    enum { buffer_size = N };
+    inline int size() { return N; }
+};
+
+template<int N, class T = float>
+class mem_fixed_size_buffer: public fixed_size_buffer<N, T> {
+    T *buf;
+public:
+    mem_fixed_size_buffer(T ubuf[N]) { buf = ubuf; }
+    void set_data(T buf[N]) { this->buf = buf; }
+    inline T* data() { return buf; }
+    inline const T* data() const { return buf; }
+    inline T& operator[](int pos) { return buf[pos]; }
+    inline const T& operator[](int pos) const { return buf[pos]; }
+};
+
+template<int N, class T = float>
+class auto_buffer: public fixed_size_buffer<N, T> {
+    T buf[N];
+public:    
+    T* data() const { return buf; }
+    inline T& operator[](int pos) { return buf[pos]; }
+    inline const T& operator[](int pos) const { return buf[pos]; }
+};
+
+template<class T = float>
+class dynamic_buffer {
+    T *buf;
+    int buf_size;
+    bool owns;
+public:
+    dynamic_buffer() { owns = false; }
+    dynamic_buffer(T *_buf, int _buf_size, bool _own)
+    : buf(_buf), buf_size(_buf_size), owns(_own) {
+    }
+    dynamic_buffer(int _size) {
+        buf = new T[_size];
+        buf_size = _size;
+        owns = true;
+    }
+    inline T* data() { return buf; }
+    inline const T* data() const { return buf; }
+    inline int size() { return buf_size; }
+    void resize(int new_size, bool fill_with_zeros = false) {
+        T *new_buf = new T[new_size];
+        memcpy(new_buf, buf, std::min(buf_size, new_size));
+        if (fill_with_zeros && buf_size < new_size)
+            dsp::zero(new_buf + buf_size, new_size - buf_size);
+        if (owns)
+            delete []buf;
+        buf = new_buf;
+        buf_size = new_size;
+        owns = true;
+    }
+    inline T& operator[](int pos) { return buf[pos]; }
+    inline const T& operator[](int pos) const { return buf[pos]; }
+    ~dynamic_buffer() {
+        if (owns)
+            delete []buf;
+    }    
+}; 
+
+template<class T, class U>
+void copy_buf(T &dest_buf, const U &src_buf, T scale = 1, T add = 0) {
+    typedef typename T::data_type data_type;
+    data_type *dest = dest_buf.data();
+    const data_type *src = src_buf.data();
+    int size = src.size();
+    for (int i=0; i<size; i++) 
+        *dest++ = (*src++) * scale + add;
+}
+
+template<class T>
+struct buffer_traits {
+};
+
+/// this class template defines some basic position operations for fixed_size_buffers
+template<int N, class T>
+struct buffer_traits<fixed_size_buffer<N, T> > {
+    int inc_wrap(int pos) const { 
+        return wrap_around<T::size>(pos+1);
+    }
+    
+    int pos_diff(int pos1, int pos2) const { 
+        int pos = pos1 - pos2;
+        if (pos < 0) pos += T::size;
+        return pos;
+    }
+};
+
+/// this is useless for now (and untested too)
+template<class B>
+class circular_buffer: public B {
+    typedef typename B::data_type data_type;
+    typedef class buffer_traits<B> traits;
+    B buffer;
+    int rpos, wpos;
+    circular_buffer() {
+        clear();
+    }
+    void clear() {
+        rpos = 0;
+        wpos = 0;
+    }
+    inline void put(data_type data) {
+        buffer[wpos] = data;
+        wpos = traits::inc_wrap(wpos);
+    }
+    inline bool empty() {
+        return rpos == wpos;
+    }
+    inline bool full() {
+        return rpos == traits::inc_wrap(wpos);
+    }
+    inline const data_type& get() {
+        int oldrpos = rpos;
+        rpos = traits::inc_wrap(rpos);
+        return buffer[oldrpos];
+    }
+    inline int get_rbytes() {
+        return traits::pos_diff(wpos, rpos);
+    }
+    inline int get_wbytes() {
+        if (full()) return 0;
+        return traits::pos_diff(rpos, wpos);
+    }
+};
+
+/// this is useless for now
+template<int N, class T = float>
+class mono_auto_buffer: public auto_buffer<N, T> {
+};
+
+/// this is useless for now
+template<int N, class T = float>
+class stereo_auto_buffer: public auto_buffer<N, stereo_sample<T> > {
+};
+
+};
+
+#endif

plugins/ladspa_effect/calf/calf/delay.h

@@ -0,0 +1,185 @@
+/* Calf DSP Library
+ * Reusable audio effect classes.
+ *
+ * Copyright (C) 2001-2007 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA  02110-1301  USA
+ */
+#ifndef __CALF_DELAY_H
+#define __CALF_DELAY_H
+
+#include "primitives.h"
+#include "buffer.h"
+#include "onepole.h"
+
+namespace dsp {
+
+/**
+ * Delay primitive. Can be used for most delay stuff, including
+ * variable (modulated) delays like chorus/flanger. Note that
+ * for modulated delay effects use of GetInterp is preferred,
+ * because it handles fractional positions and uses linear
+ * interpolation, which sounds better most of the time.
+ *
+ * @param N maximum length
+ * @param C number of channels read/written for each sample (1 mono, 2 stereo etc)
+ */
+template<int N, class T> 
+struct simple_delay {
+    auto_buffer<N, T> data;
+    int pos;
+
+    simple_delay() {
+        reset();
+    }
+    void reset() {
+        pos = 0;
+        for (int i=0; i<N; i++)
+            zero(data[i]);
+    }    
+    /** Write one C-channel sample from idata[0], idata[1] etc into buffer */
+    inline void put(T idata) {
+        data[pos] = idata;
+        pos = wrap_around<N>(pos+1);
+    }
+
+    /** 
+     * Read one C-channel sample into odata[0], odata[1] etc into buffer.
+     * Don't use for modulated delays (chorus/flanger etc) unless you
+     * want them to crackle and generally sound ugly
+     * @param odata pointer to write into
+     * @param delay delay relative to current writing pos
+     */
+    template<class U>
+    inline void get(U &odata, int delay) {
+        assert(delay >= 0 && delay < N);
+        int ppos = wrap_around<N>(pos + N - delay);
+        odata = data[ppos];
+    }
+    
+    /**
+     * Read and write during the same function call
+     */
+    inline T process(T idata, int delay)
+    {
+        assert(delay >= 0 && delay < N);
+        int ppos = wrap_around<N>(pos + N - delay);
+        T odata = data[ppos];
+        data[pos] = idata;
+        pos = wrap_around<N>(pos+1);
+        return odata;
+    }
+
+    /** Read one C-channel sample at fractional position.
+     * This version can be used for modulated delays, because
+     * it uses linear interpolation.
+     * @param odata value to write into
+     * @param delay delay relative to current writing pos
+     * @param udelay fractional delay (0..1)
+     */
+    template<class U>
+    inline void get_interp(U &odata, int delay, float udelay) {
+//        assert(delay >= 0 && delay < N-1);
+        int ppos = wrap_around<N>(pos + N - delay);
+        int pppos = wrap_around<N>(ppos + N - 1);
+        odata = lerp(data[ppos], data[pppos], udelay);
+    }
+    
+    /** Read one C-channel sample at fractional position.
+     * This version can be used for modulated delays, because
+     * it uses linear interpolation.
+     * @param odata value to write into
+     * @param delay delay relative to current writing pos
+     * @param udelay fractional delay (0..1)
+     */
+    inline T get_interp_1616(unsigned int delay) {
+        float udelay = (float)((delay & 0xFFFF) * (1.0 / 65536.0));
+        delay = delay >> 16;
+//        assert(delay >= 0 && delay < N-1);
+        int ppos = wrap_around<N>(pos + N - delay);
+        int pppos = wrap_around<N>(ppos + N - 1);
+        return lerp(data[ppos], data[pppos], udelay);
+    }
+    
+    /**
+     * Comb filter. Feedback delay line with given delay and feedback values
+     * @param in input signal
+     * @param delay delay length (must be <N and integer)
+     * @param fb feedback (must be <1 or it will be unstable)
+     */
+    inline T process_comb(T in, unsigned int delay, float fb)
+    {
+        T old, cur;
+        get(old, delay);
+        cur = in + fb*old;
+        sanitize(cur);
+        put(cur);
+        return old;
+    }
+    
+    /**
+     * Comb filter with linear interpolation. Feedback delay line with given delay and feedback values
+     * Note that linear interpolation introduces some weird effects in frequency response.
+     * @param in input signal
+     * @param delay fractional delay length (must be < 65536 * N)
+     * @param fb feedback (must be <1 or it will be unstable)
+     */
+    inline T process_comb_lerp16(T in, unsigned int delay, float udelay, float fb)
+    {
+        T old, cur;
+        get_interp(old, delay>>16, dsp::fract16(delay));
+        cur = in + fb*old;
+        sanitize(cur);
+        put(cur);
+        return old;
+    }
+    
+    /**
+     * Comb allpass filter. The comb filter with additional direct path, which is supposed to cancel the coloration.
+     * @param in input signal
+     * @param delay delay length (must be <N and integer)
+     * @param fb feedback (must be <1 or it will be unstable)
+     */
+    inline T process_allpass_comb(T in, unsigned int delay, float fb)
+    {
+        T old, cur;
+        get(old, delay);
+        cur = in + fb*old;
+        sanitize(cur);
+        put(cur);
+        return old - fb * cur;
+    }
+
+        /**
+     * Comb allpass filter. The comb filter with additional direct path, which is supposed to cancel the coloration.
+     * @param in input signal
+     * @param delay fractional delay length (must be < 65536 * N)
+     * @param fb feedback (must be <1 or it will be unstable)
+     */
+    inline T process_allpass_comb_lerp16(T in, unsigned int delay, float fb)
+    {
+        T old, cur;
+        get_interp(old, delay>>16, dsp::fract16(delay));
+        cur = in + fb*old;
+        sanitize(cur);
+        put(cur);
+        return old - fb * cur;
+    }
+};
+
+};
+
+#endif

plugins/ladspa_effect/calf/calf/envelope.h

@@ -0,0 +1,206 @@
+/* Calf DSP Library
+ * ADSR envelope class (and other envelopes in future)
+ *
+ * Copyright (C) 2007-2008 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA  02110-1301  USA
+ */
+#ifndef __CALF_ENVELOPE_H
+#define __CALF_ENVELOPE_H
+
+#include "primitives.h"
+
+namespace dsp {
+
+/// Rate-based ADSFR envelope class. Note that if release rate is slower than decay
+/// rate, this envelope won't use release rate until output level falls below sustain level
+/// it's different to what certain hardware synth companies did, but it prevents the very
+/// un-musical (IMHO) behaviour known from (for example) SoundFont 2.
+class adsr
+{
+public:
+    enum env_state { 
+        STOP, ///< envelope is stopped
+        ATTACK, ///< attack - rise from 0 to 1
+        DECAY, ///< decay - fall from 1 to sustain level
+        SUSTAIN, ///< sustain - remain at sustain level (unless sustain is 0 - then it gets stopped); with fade != 0 it goes towards 0% (positive fade) or 100% (negative fade)
+        RELEASE, ///< release - fall from sustain (or pre-sustain) level to 0
+        LOCKDECAY,  ///< locked decay 
+    };
+    
+    /// Current envelope stage
+    env_state state;
+    /// @note these are *rates*, not times
+    double attack, decay, sustain, release, fade;
+    /// Requested release time (not the rate!) in frames, used for recalculating the rate if sustain is changed
+    double release_time;
+    /// Current envelope (output) level
+    double value;
+    /// Release rate used for the current note (calculated from this note's sustain level, and not the current sustain level,
+    /// which may have changed after note has been released)
+    double thisrelease;
+    /// Sustain level used for the current note (used to calculate release rate if sustain changed during release stage
+    /// of the current note)
+    double thiss;
+    /// Value from the time before advance() was called last time
+    double old_value;
+    
+    adsr()
+    {
+        attack = decay = sustain = release = thisrelease = thiss = 0.f;
+        reset();
+    }
+    /// Stop (reset) the envelope
+    inline void reset()
+    {
+        old_value = value = 0.0;
+        thiss = 0.0;
+        state = STOP;
+    }
+    /// Set the envelope parameters (updates rate member variables based on values passed)
+    /// @param a attack time
+    /// @param d decay time
+    /// @param s sustain level
+    /// @param r release time
+    /// @param er Envelope (update) rate
+    /// @param f fade time (if applicable)
+    inline void set(float a, float d, float s, float r, float er, float f = 0.f)
+    {
+        attack = 1.0 / (a * er);
+        decay = (1 - s) / (d * er);
+        sustain = s;
+        release_time = r * er;
+        release = s / release_time;
+        if (fabs(f) > small_value<float>())
+            fade = 1.0 / (f * er);
+        else
+            fade = 0.0;
+        // in release:
+        // lock thiss setting (start of release for current note) and unlock thisrelease setting (current note's release rate)
+        if (state != RELEASE)
+            thiss = s;
+        else
+            thisrelease = thiss / release_time;
+    }
+    /// @retval true if envelope is in released state (forced decay, release or stopped)
+    inline bool released() const
+    {
+        return state == LOCKDECAY || state == RELEASE || state == STOP;
+    }
+    /// @retval true if envelope is stopped (has not been started or has run till its end)
+    inline bool stopped() const
+    {
+        return state == STOP;
+    }
+    /// Start the envelope
+    inline void note_on()
+    {
+        state = ATTACK;
+        thiss = sustain;
+    }
+    /// Release the envelope
+    inline void note_off()
+    {
+        // Do nothing if envelope is already stopped
+        if (state == STOP)
+            return;
+        // XXXKF what if envelope is already released? (doesn't happen in any current synth, but who knows?)
+        // Raise sustain value if it has been changed... I'm not sure if it's needed
+        thiss = std::max(sustain, value);
+        // Calculate release rate from sustain level
+        thisrelease = thiss / release_time;
+        // we're in attack or decay, and if decay is faster than release
+        if (value > sustain && decay > thisrelease) {
+            // use standard release time later (because we'll be switching at sustain point)
+            thisrelease = release;
+            state = LOCKDECAY;
+        } else {
+            // in attack/decay, but use fixed release time
+            // in case value fell below sustain, assume it didn't (for the purpose of calculating release rate only)
+            state = RELEASE;
+        }
+    }
+    /// Calculate next envelope value
+    inline void advance()
+    {
+        old_value = value;
+        // XXXKF This may use a state array instead of a switch some day (at least for phases other than attack and possibly sustain)
+        switch(state)
+        {
+        case ATTACK:
+            value += attack;
+            if (value >= 1.0) {
+                value = 1.0;
+                state = DECAY;
+            }
+            break;
+        case DECAY:
+            value -= decay;
+            if (value < sustain)
+            {
+                value = sustain;
+                state = SUSTAIN;
+            }
+            break;
+        case LOCKDECAY:
+            value -= decay;
+            if (value < sustain)
+            {
+                if (value < 0.f)
+                    value = 0.f;
+                state = RELEASE;
+                thisrelease = release;
+            }
+            break;
+        case SUSTAIN:
+            if (fade != 0.f)
+            {
+                value -= fade;
+                if (value > 1.f)
+                    value = 1.f;
+            }
+            else
+                value = sustain;
+            if (value < 0.00001f) {
+                value = 0;
+                state = STOP;
+            }
+            break;
+        case RELEASE:
+            value -= thisrelease;
+            if (value <= 0.f) {
+                value = 0.f;
+                state = STOP;
+            }
+            break;
+        case STOP:
+            value = 0.f;
+            break;
+        }
+    }
+    /// Return a value between old_value (previous step) and value (current step)
+    /// @param pos between 0 and 1
+    inline double interpolate(double pos)
+    {
+        return old_value + (value - old_value) * pos;
+    }
+};
+
+};
+
+#endif
+
+

plugins/ladspa_effect/calf/calf/fft.h

@@ -0,0 +1,113 @@
+/* Calf DSP Library
+ * FFT class
+ *
+ * Copyright (C) 2007 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __CALF_FFT_H
+#define __CALF_FFT_H
+
+#include <complex>
+
+namespace dsp {
+
+/// FFT routine copied from my old OneSignal library, modified to 
+/// match Calf's style. It's not fast at all, just a straightforward
+/// implementation.
+template<class T, int O>
+class fft
+{
+    typedef typename std::complex<T> complex;
+    int scramble[1<<O];
+    complex sines[1<<O];
+public:
+    fft()
+    {
+        int N=1<<O;
+        assert(N >= 4);
+        for (int i=0; i<N; i++)
+        {
+            int v=0;
+            for (int j=0; j<O; j++)
+                if (i&(1<<j))
+                    v+=(N>>(j+1));
+            scramble[i]=v;
+        }
+        int N90 = N >> 2;
+        T divN = 2 * M_PI / N;
+        // use symmetry
+        for (int i=0; i<N90; i++)
+        {
+            T angle = divN * i;
+            T c = cos(angle), s = sin(angle);
+            sines[i + 3 * N90] = -(sines[i + N90] = complex(-s, c));
+            sines[i + 2 * N90] = -(sines[i] = complex(c, s));
+        }
+    }
+    void calculate(complex *input, complex *output, bool inverse)
+    {
+        int N=1<<O;
+        int N1=N-1;
+        int i;
+        // Scramble the input data
+        if (inverse)
+        {
+            float mf=1.0/N;
+            for (i=0; i<N; i++)
+            {
+                complex &c=input[scramble[i]];
+                output[i]=mf*complex(c.imag(),c.real());
+            }
+        }
+        else
+            for (i=0; i<N; i++)
+                output[i]=input[scramble[i]];
+
+        // O butterfiles
+        for (i=0; i<O; i++)
+        {
+            int PO=1<<i, PNO=1<<(O-i-1);
+            int j,k;
+            for (j=0; j<PNO; j++)
+            {
+                int base=j<<(i+1);
+                for (k=0; k<PO; k++)
+                {
+                    int B1=base+k;
+                    int B2=base+k+(1<<i);
+                    complex r1=output[B1];
+                    complex r2=output[B2];
+                    output[B1]=r1+r2*sines[(B1<<(O-i-1))&N1];
+                    output[B2]=r1+r2*sines[(B2<<(O-i-1))&N1];
+                }
+            }
+        }
+        if (inverse)
+        {
+            for (i=0; i<N; i++)
+            {
+                const complex &c=output[i];
+                output[i]=complex(c.imag(),c.real());
+            }
+        }
+    }
+};
+
+};
+
+#endif

plugins/ladspa_effect/calf/calf/fixed_point.h

@@ -0,0 +1,269 @@
+/* Calf DSP Library
+ * DSP primitives.
+ *
+ * Copyright (C) 2001-2007 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+#ifndef __CALF_FIXED_POINT_H
+#define __CALF_FIXED_POINT_H
+
+namespace dsp {
+
+inline uint32_t shr(uint32_t v, int bits = 1) { return v>>bits; };
+inline int32_t shr(int32_t v, int bits = 1) { return v>>bits; };
+inline uint64_t shr(uint64_t v, int bits = 1) { return v>>bits; };
+inline int64_t shr(int64_t v, int bits = 1) { return v>>bits; };
+inline float shr(float v, int bits = 1) { return v*(1.0/(1<<bits)); };
+inline double shr(double v, int bits = 1) { return v*(1.0/(1<<bits)); };
+template<class T, int FracBits>
+inline T shr(T v, int bits = 1) {
+    v.set(v >> bits);
+    return v;
+}
+
+template<class T, int FracBits> class fixed_point {
+    T value;
+    enum { IntBits = (sizeof(T)/8) - FracBits };
+
+public:
+    /// default constructor, does not initialize the value, just like - say - float doesn't
+    inline fixed_point() {
+    }
+
+    /// copy constructor from any other fixed_point value
+    template<class U, int FracBits2> inline fixed_point(const fixed_point<U, FracBits2> &v) {
+        if (FracBits == FracBits2) value = v.get();
+        else if (FracBits > FracBits2) value = v.get() << abs(FracBits - FracBits2);
+        else value = v.get() >> abs(FracBits - FracBits2);
+    }
+
+    /* this would be way too confusing, it wouldn't be obvious if it expects a whole fixed point or an integer part
+    explicit inline fixed_point(T v) {
+        this->value = v;
+    }
+    */
+    explicit inline fixed_point(double v) {
+        value = (T)(v*one());
+    }
+    
+    /// Makes an instance from a representation value (ie. same type of value as is used for internal storage and get/set)
+    static inline fixed_point from_base(const T &v) 
+    {
+        fixed_point result;
+        result.value = v;
+        return result;
+    }
+    
+    inline static T one() {
+        return (T)(1) << FracBits;
+    }
+
+    inline void set(T value) {
+        this->value = value;
+    }
+      
+    inline T get() const {
+        return value;
+    }
+      
+    inline operator double() const {
+        return value * (1.0/one());
+    }
+    
+    inline fixed_point &operator=(double v) {
+        value = (T)(v*one());
+        return *this;
+    }
+    
+    template<class U, int FracBits2> static inline T rebase(const fixed_point<U, FracBits2> &v) {
+        if (FracBits == FracBits2) 
+            return v.get();
+        if (FracBits > FracBits2) 
+            return v.get() << abs(FracBits - FracBits2);
+        return v.get() >> abs(FracBits2 - FracBits);
+    }
+    
+    template<class U, int FracBits2> inline fixed_point &operator=(const fixed_point<U, FracBits2> &v) {
+        value = rebase<U, FracBits2>(v);
+        return *this;
+    }
+
+    template<class U, int FracBits2> inline fixed_point &operator+=(const fixed_point<U, FracBits2> &v) {
+        value += rebase<U, FracBits2>(v);
+        return *this;
+    }
+
+    template<class U, int FracBits2> inline fixed_point &operator-=(const fixed_point<U, FracBits2> &v) {
+        value -= rebase<U, FracBits2>(v);
+        return *this;
+    }
+
+    template<class U, int FracBits2> inline fixed_point operator+(const fixed_point<U, FracBits2> &v) const {
+        fixed_point fpv;
+        fpv.set(value + rebase<U, FracBits2>(v));
+        return fpv;
+    }
+
+    template<class U, int FracBits2> inline fixed_point operator-(const fixed_point<U, FracBits2> &v) const {
+        fixed_point fpv;
+        fpv.set(value - rebase<U, FracBits2>(v));
+        return fpv;
+    }
+
+    /// multiply two fixed point values, using long long int to store the temporary multiplication result
+    template<class U, int FracBits2> inline fixed_point operator*(const fixed_point<U, FracBits2> &v) const {
+        fixed_point tmp;
+        tmp.set(((int64_t)value) * v.get() >> FracBits2);
+        return tmp;
+    }
+
+    /// multiply two fixed point values, using BigType (usually 64-bit int) to store the temporary multiplication result
+    template<class U, int FracBits2, class BigType> inline fixed_point& operator*=(const fixed_point<U, FracBits2> &v) {
+        value = (T)(((BigType)value) * v.get() >> FracBits2);
+        return *this;
+    }
+
+    inline fixed_point operator+(int v) const {
+        fixed_point tmp;
+        tmp.set(value + (v << FracBits));
+        return tmp;
+    }
+
+    inline fixed_point operator-(int v) const {
+        fixed_point tmp;
+        tmp.set(value - (v << FracBits));
+        return tmp;
+    }
+
+    inline fixed_point operator*(int v) const {
+        fixed_point tmp;
+        tmp.value = value*v;
+        return tmp;
+    }
+
+    inline fixed_point& operator+=(int v) {
+        value += (v << FracBits);
+        return *this;
+    }
+
+    inline fixed_point& operator-=(int v) {
+        value -= (v << FracBits);
+        return *this;
+    }
+
+    inline fixed_point& operator*=(int v) {
+        value *= v;
+        return *this;
+    }
+
+    /// return integer part
+    inline T ipart() const {
+        return value >> FracBits;
+    }
+
+    /// return integer part as unsigned int
+    inline unsigned int uipart() const {
+        return ((unsigned)value) >> FracBits;
+    }
+
+    /// return integer part as unsigned int
+    inline unsigned int ui64part() const {
+        return ((uint64_t)value) >> FracBits;
+    }
+
+    /// return fractional part as 0..(2^FracBits-1)
+    inline T fpart() const {
+        return value & ((1 << FracBits)-1);
+    }
+
+    /// return fractional part as 0..(2^Bits-1)
+    template<int Bits>
+    inline T fpart() const {
+        int fbits = value & ((1 << FracBits)-1);
+        if (Bits == FracBits) return fbits;
+        int shift = abs(Bits-FracBits);
+        return (Bits < FracBits) ? (fbits >> shift) : (fbits << shift);
+    }
+
+    /// return fractional part as 0..1
+    inline double fpart_as_double() const {
+        return (value & ((1 << FracBits)-1)) * (1.0 / (1 << FracBits));
+    }
+
+    /// use fractional part (either whole or given number of most significant bits) for interpolating between two values
+    /// note that it uses integer arithmetic only, and isn't suitable for floating point or fixed point U!
+    /// @param UseBits can be used when there's a risk of exceeding range of U because max(fpart)*max(v1 or v2) > range of U
+    template<class U, int UseBits, class MulType> 
+    inline U lerp_by_fract_int(U v1, U v2) const {
+        int fp = fpart<UseBits>();
+        assert ( fp >=0 && fp <= (1<<UseBits));
+        // printf("diff = 
+        return v1 + shr(((MulType)(v2-v1) * fp), UseBits);
+    }
+
+    template<class U, int UseBits> 
+    inline U lerp_table_lookup_int(U data[(1<<IntBits)+1]) const {
+        unsigned int pos = uipart();
+        return lerp_by_fract_int<U, UseBits>(data[pos], data[pos+1]);
+    }
+
+    /// Untested... I've started it to get a sin/cos readout for rotaryorgan, but decided to use table-less solution instead
+    /// Do not assume it works, because it most probably doesn't
+    template<class U, int UseBits> 
+    inline U lerp_table_lookup_int_shift(U data[(1<<IntBits)+1], unsigned int shift) {
+        unsigned int pos = (uipart() + shift) & ((1 << IntBits) - 1);
+        return lerp_by_fract_int<U, UseBits>(data[pos], data[pos+1]);
+    }
+
+    template<class U> 
+    inline U lerp_table_lookup_float(U data[(1<<IntBits)+1]) const {
+        unsigned int pos = uipart();
+        return data[pos] + (data[pos+1]-data[pos]) * fpart_as_double();
+    }
+
+    template<class U> 
+    inline U lerp_table_lookup_float_mask(U data[(1<<IntBits)+1], unsigned int mask) const {
+        unsigned int pos = ui64part() & mask;
+        // printf("full = %lld pos = %d + %f\n", value, pos, fpart_as_double());
+        return data[pos] + (data[pos+1]-data[pos]) * fpart_as_double();
+    }
+
+    template<class U, int UseBits, class MulType> 
+    inline U lerp_ptr_lookup_int(U *data) const {
+        unsigned int pos = ui64part();
+        return lerp_by_fract_int<U, UseBits, MulType>(data[pos], data[pos+1]);
+    }
+
+    template<class U> 
+    inline U lerp_ptr_lookup_float(U *data) const {
+        unsigned int pos = ui64part();
+        return data[pos] + (data[pos+1]-data[pos]) * fpart_as_double();
+    }
+};
+
+template<class T, int FractBits>
+inline fixed_point<T, FractBits> operator*(int v, fixed_point<T, FractBits> v2) {
+    v2 *= v;
+    return v2;
+}
+
+/// wave position (unsigned 64-bit int including 24-bit fractional part)
+typedef fixed_point<unsigned long long int, 24> wpos;
+
+};
+
+#endif

plugins/ladspa_effect/calf/calf/giface.h

@@ -0,0 +1,561 @@
+/* Calf DSP Library
+ * Common plugin interface definitions (shared between LADSPA/LV2/DSSI/standalone).
+ *
+ * Copyright (C) 2007 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+#ifndef __CALF_GIFACE_H
+#define __CALF_GIFACE_H
+
+#include <stdint.h>
+#include <stdlib.h>
+#include <pthread.h>
+#include <exception>
+#include <string>
+#include "primitives.h"
+#include "preset.h"
+
+namespace osctl {
+    struct osc_client;
+}
+
+namespace calf_plugins {
+
+enum {
+    MAX_SAMPLE_RUN = 256
+};
+    
+/// Values ORed together for flags field in parameter_properties
+enum parameter_flags
+{
+  PF_TYPEMASK = 0x000F, ///< bit mask for type
+  PF_FLOAT = 0x0000, ///< any float value
+  PF_INT = 0x0001,   ///< integer value (still represented as float)
+  PF_BOOL = 0x0002,  ///< bool value (usually >=0.5f is treated as TRUE, which is inconsistent with LV2 etc. which treats anything >0 as TRUE)
+  PF_ENUM = 0x0003,  ///< enum value (min, min+1, ..., max, only guaranteed to work when min = 0)
+  PF_ENUM_MULTI = 0x0004, ///< SET / multiple-choice
+  PF_STRING = 0x0005, ///< see: http://lv2plug.in/docs/index.php?title=String_port
+  
+  PF_SCALEMASK = 0xF0, ///< bit mask for scale
+  PF_SCALE_DEFAULT = 0x00, ///< no scale given
+  PF_SCALE_LINEAR = 0x10, ///< linear scale
+  PF_SCALE_LOG = 0x20, ///< log scale
+  PF_SCALE_GAIN = 0x30, ///< gain = -96dB..0 or -inf dB
+  PF_SCALE_PERC = 0x40, ///< percent
+  PF_SCALE_QUAD = 0x50, ///< quadratic scale (decent for some gain/amplitude values)
+  PF_SCALE_LOG_INF = 0x60, ///< log scale + +inf (FAKE_INFINITY)
+
+  PF_CTLMASK =     0x0F00, ///< bit mask for control type
+  PF_CTL_DEFAULT = 0x0000, ///< try to figure out automatically
+  PF_CTL_KNOB =    0x0100, ///< knob
+  PF_CTL_FADER =   0x0200, ///< fader (slider)
+  PF_CTL_TOGGLE =  0x0300, ///< toggle button
+  PF_CTL_COMBO =   0x0400, ///< combo box
+  PF_CTL_RADIO =   0x0500, ///< radio button
+  PF_CTL_BUTTON =  0x0600, ///< push button
+  PF_CTL_METER  =  0x0700, ///< volume meter
+  PF_CTL_LED    =  0x0800, ///< light emitting diode
+  
+  PF_CTLOPTIONS     = 0x00F000, ///< bit mask for control (widget) options
+  PF_CTLO_HORIZ     = 0x001000, ///< horizontal version of the control (unused)
+  PF_CTLO_VERT      = 0x002000, ///< vertical version of the control (unused)
+  PF_CTLO_LABEL     = 0x004000, ///< add a text display to the control (meters only)
+  PF_CTLO_REVERSE   = 0x008000, ///< use VU_MONOCHROME_REVERSE mode (meters only)
+
+  PF_PROP_NOBOUNDS =  0x010000, ///< no epp:hasStrictBounds
+  PF_PROP_EXPENSIVE = 0x020000, ///< epp:expensive, may trigger expensive calculation
+  PF_PROP_OUTPUT_GAIN=0x050000, ///< epp:outputGain + skip epp:hasStrictBounds
+  PF_PROP_OUTPUT    = 0x080000, ///< output port
+  PF_PROP_OPTIONAL  = 0x100000, ///< connection optional
+  PF_PROP_GRAPH     = 0x200000, ///< add graph
+  PF_PROP_MSGCONTEXT= 0x400000, ///< message context
+  
+  PF_UNITMASK     = 0xFF000000,  ///< bit mask for units   \todo reduce to use only 5 bits
+  PF_UNIT_DB      = 0x01000000,  ///< decibels
+  PF_UNIT_COEF    = 0x02000000,  ///< multiply-by factor
+  PF_UNIT_HZ      = 0x03000000,  ///< Hertz
+  PF_UNIT_SEC     = 0x04000000,  ///< second
+  PF_UNIT_MSEC    = 0x05000000,  ///< millisecond
+  PF_UNIT_CENTS   = 0x06000000,  ///< cents (1/100 of a semitone, 1/1200 of an octave)
+  PF_UNIT_SEMITONES = 0x07000000,///< semitones
+  PF_UNIT_BPM     = 0x08000000,  ///< beats per minute
+  PF_UNIT_DEG     = 0x09000000,  ///< degrees
+  PF_UNIT_NOTE    = 0x0A000000,  ///< MIDI note number
+  PF_UNIT_RPM     = 0x0B000000,  ///< revolutions per minute
+};
+
+/// A fake infinity value (because real infinity may break some hosts)
+#define FAKE_INFINITY (65536.0 * 65536.0)
+/// Check for infinity (with appropriate-ish tolerance)
+#define IS_FAKE_INFINITY(value) (fabs(value-FAKE_INFINITY) < 1.0)
+
+/// Information record about plugin's menu command
+struct plugin_command_info
+{
+    const char *label;           ///< short command name / label
+    const char *name;            ///< human-readable command name
+    const char *description;     ///< description (for status line etc.)
+};
+
+/// Range, default value, flags and names for a parameter
+struct parameter_properties
+{
+    /// default value
+    float def_value;
+    /// minimum value
+    float min;
+    /// maximum value
+    float max;
+    /// number of steps (for an integer value from 0 to 100 this will be 101; for 0/90/180/270/360 this will be 5), or 0 for continuous
+    float step;
+    /// logical OR of parameter_flags
+    uint32_t flags;
+    /// for PF_ENUM: array of text values (from min to max step 1), otherwise NULL
+    const char **choices;
+    /// parameter label (for use in LV2 label field etc.)
+    const char *short_name;
+    /// parameter human-readable name
+    const char *name;
+    /// convert from [0, 1] range to [min, max] (applying scaling)
+    float from_01(double value01) const;
+    /// convert from [min, max] to [0, 1] range (applying reverse scaling)
+    double to_01(float value) const;
+    /// stringify (in sensible way)
+    std::string to_string(float value) const;
+    /// get required width (for reserving GUI space)
+    int get_char_count() const;
+    /// get increment step based on step value (if specified) and other factors
+    float get_increment() const;
+};
+
+struct cairo_iface
+{
+    virtual void set_source_rgba(float r, float g, float b, float a = 1.f) = 0;
+    virtual void set_line_width(float width) = 0;
+    virtual ~cairo_iface() {}
+};
+
+struct progress_report_iface
+{
+    virtual void report_progress(float percentage, const std::string &message) = 0;
+    virtual ~progress_report_iface() {}
+};
+
+/// 'provides live line graph values' interface
+struct line_graph_iface
+{
+    /// Obtain subindex'th graph of parameter 'index'
+    /// @param index parameter/graph number (usually tied to particular plugin control port)
+    /// @param subindex graph number (there may be multiple overlaid graphs for one parameter, eg. for monosynth 2x12dB filters)
+    /// @param data buffer for normalized output values
+    /// @param points number of points to fill
+    /// @param context cairo context to adjust (for multicolour graphs etc.)
+    /// @retval true graph data was returned; subindex+1 graph may or may not be available
+    /// @retval false graph data was not returned; subindex+1 graph does not exist either
+    virtual bool get_graph(int index, int subindex, float *data, int points, cairo_iface *context) { return false; }
+
+    /// Obtain subindex'th dot of parameter 'index'
+    /// @param index parameter/dot number (usually tied to particular plugin control port)
+    /// @param subindex dot number (there may be multiple dots graphs for one parameter)
+    virtual bool get_dot(int index, int subindex, float &x, float &y, int &size, cairo_iface *context) { return false; }
+    
+    /// Obtain subindex'th dot of parameter 'index'
+    /// @param index parameter/dot number (usually tied to particular plugin control port)
+    /// @param subindex dot number (there may be multiple dots graphs for one parameter)
+    virtual bool get_gridline(int index, int subindex, float &pos, bool &vertical, std::string &legend, cairo_iface *context) { return false; }
+    
+    /// Obtain subindex'th static graph of parameter index (static graphs are only dependent on parameter value, not plugin state)
+    /// @param index parameter/graph number (usually tied to particular plugin control port)
+    /// @param subindex graph number (there may be multiple overlaid graphs for one parameter, eg. for monosynth 2x12dB filters)
+    /// @param value parameter value to pick the graph for
+    /// @param data buffer for normalized output values
+    /// @param points number of points to fill
+    /// @param context cairo context to adjust (for multicolour graphs etc.)
+    /// @retval true graph data was returned; subindex+1 graph may or may not be available
+    /// @retval false graph data was not returned; subindex+1 graph does not exist either
+    virtual bool get_static_graph(int index, int subindex, float value, float *data, int points, cairo_iface *context) { return false; }
+    
+    /// Return which graphs need to be redrawn and which can be cached for later reuse
+    /// @param generation 0 (at start) or the last value returned by the function (corresponds to a set of input values)
+    /// @param subindex_graph First graph that has to be redrawn (because it depends on values that might have changed)
+    /// @param subindex_dot First dot that has to be redrawn
+    /// @param subindex_gridline First gridline/legend that has to be redrawn
+    /// @retval Current generation (to pass when calling the function next time); if different than passed generation value, call the function again to retrieve which graph offsets should be put into cache
+    virtual int get_changed_offsets(int generation, int &subindex_graph, int &subindex_dot, int &subindex_gridline) { subindex_graph = subindex_dot = subindex_gridline = 0; return 0; }
+    
+    /// Standard destructor to make compiler happy
+    virtual ~line_graph_iface() {}
+};
+
+enum table_column_type
+{
+    TCT_UNKNOWN, ///< guard invalid type
+    TCT_FLOAT, ///< float value (encoded as C locale string)
+    TCT_ENUM, ///< enum value (see: 'values' array in table_column_info) - encoded as string base 10 representation of integer
+    TCT_STRING, ///< string value (encoded as C-escaped string)
+    TCT_OBJECT, ///< external object, encoded as string
+    TCT_LABEL, ///< string value (encoded as C-escaped string)
+};
+
+/// parameters of 
+struct table_column_info
+{
+    const char *name; ///< column label
+    table_column_type type; ///< column data type
+    float min; ///< minimum value (for float)
+    float max; ///< maximum value (for float and enum)
+    float def_value; ///< default value (for float and enum)
+    const char **values; ///< NULL unless a TCT_ENUM, where it represents a NULL-terminated list of choices
+};
+
+/// 'has string parameters containing tabular data' interface
+struct table_edit_iface
+{
+    /// retrieve the table layout for specific parameter
+    virtual const table_column_info *get_table_columns(int param) = 0;
+
+    /// return the current number of rows
+    virtual uint32_t get_table_rows(int param) = 0;
+    
+    /// retrieve data item from the plugin
+    virtual std::string get_cell(int param, int row, int column) { return calf_utils::i2s(row)+":"+calf_utils::i2s(column); }
+
+    /// set data item to the plugin
+    virtual void set_cell(int param, int row, int column, const std::string &src, std::string &error) { error.clear(); }
+    
+    /// return a line graph interface for a specific parameter/column (unused for now)
+    virtual line_graph_iface *get_graph_iface(int param, int column) { return NULL; }
+    
+    /// return an editor name for a specific grid cell (unused for now - I don't even know how editors be implemented)
+    virtual const char *get_cell_editor(int param, int column) { return NULL; }
+    
+    virtual ~table_edit_iface() {}
+};
+
+/// 'may receive configure variables' interface
+struct send_configure_iface
+{
+    /// Called to set configure variable
+    /// @param key variable name
+    /// @param value variable content
+    virtual void send_configure(const char *key, const char *value) = 0;
+    
+    virtual ~send_configure_iface() {}
+};
+
+/// 'may receive new status values' interface
+struct send_updates_iface
+{
+    /// Called to set configure variable
+    /// @param key variable name
+    /// @param value variable content
+    virtual void send_status(const char *key, const char *value) = 0;
+    
+    virtual ~send_updates_iface() {}
+};
+
+struct plugin_command_info;
+
+/// General information about the plugin - @todo XXXKF lacks the "new" id-label-name triple
+struct ladspa_plugin_info
+{
+    /// LADSPA ID
+    uint32_t unique_id;
+    /// plugin short name (camel case)
+    const char *label;
+    /// plugin human-readable name
+    const char *name;
+    /// maker (author)
+    const char *maker;
+    /// copyright notice
+    const char *copyright;
+    /// plugin type for LRDF/LV2
+    const char *plugin_type;
+};
+
+/// An interface returning metadata about a plugin
+struct plugin_metadata_iface
+{
+    /// @return plugin long name
+    virtual const char *get_name() = 0;
+    /// @return plugin LV2 label
+    virtual const char *get_id() = 0;
+    /// @return plugin human-readable label
+    virtual const char *get_label() = 0;
+    /// @return total number of parameters
+    virtual int get_param_count() = 0;
+    /// Return custom XML
+    virtual const char *get_gui_xml() = 0;
+    /// @return number of audio inputs
+    virtual int get_input_count()=0;
+    /// @return number of audio outputs
+    virtual int get_output_count()=0;
+    /// @return true if plugin can work in hard-realtime conditions
+    virtual bool is_rt_capable()=0;
+    /// @return true if plugin has MIDI input
+    virtual bool get_midi()=0;
+    /// @return true if plugin has MIDI input
+    virtual bool requires_midi()=0;
+    /// @return port offset of first control (parameter) port (= number of audio inputs + number of audio outputs in all existing plugins as for 1 Aug 2008)
+    virtual int get_param_port_offset() = 0;
+    /// @return line_graph_iface if any
+    virtual line_graph_iface *get_line_graph_iface() = 0;
+    /// @return table_edit_iface if any
+    virtual table_edit_iface *get_table_edit_iface() = 0;
+    /// @return NULL-terminated list of menu commands
+    virtual plugin_command_info *get_commands() { return NULL; }
+    /// @return description structure for given parameter
+    virtual parameter_properties *get_param_props(int param_no) = 0;
+    /// @return retrieve names of audio ports (@note control ports are named in parameter_properties, not here)
+    virtual const char **get_port_names() = 0;
+    /// @return description structure for the plugin
+    virtual const ladspa_plugin_info &get_plugin_info() = 0;
+    /// is a given parameter a control voltage?
+    virtual bool is_cv(int param_no) = 0;
+    /// is the given parameter non-interpolated?
+    virtual bool is_noisy(int param_no) = 0;
+    /// does the plugin require message context? (or DSSI configure) may be slow
+    virtual bool requires_message_context() = 0;
+    /// does the plugin require string port extension? (or DSSI configure) may be slow
+    virtual bool requires_string_ports() = 0;
+    /// add all message context parameter numbers to the ports vector
+    virtual void get_message_context_parameters(std::vector<int> &ports) = 0;
+
+    /// Do-nothing destructor to silence compiler warning
+    virtual ~plugin_metadata_iface() {}
+};
+
+/// Interface for host-GUI-plugin interaction (should be really split in two, but ... meh)
+struct plugin_ctl_iface: public virtual plugin_metadata_iface
+{
+    /// @return value of given parameter
+    virtual float get_param_value(int param_no) = 0;
+    /// Set value of given parameter
+    virtual void set_param_value(int param_no, float value) = 0;
+    /// Load preset with given number
+    virtual bool activate_preset(int bank, int program) = 0;
+    /// @return volume level for port'th port (if supported by the implementation, currently only jack_host<Module> implements that by measuring signal level on plugin ports)
+    virtual float get_level(unsigned int port)=0;
+    /// Execute menu command with given number
+    virtual void execute(int cmd_no)=0;
+    /// Set a configure variable on a plugin
+    virtual char *configure(const char *key, const char *value) { return NULL; }
+    /// Send all configure variables set within a plugin to given destination (which may be limited to only those that plugin understands)
+    virtual void send_configures(send_configure_iface *)=0;
+    /// Restore all state (parameters and configure vars) to default values - implemented in giface.cpp
+    virtual void clear_preset();
+    /// Call a named function in a plugin - this will most likely be redesigned soon - and never used
+    /// @retval false call has failed, result contains an error message
+    virtual bool blobcall(const char *command, const std::string &request, std::string &result) { result = "Call not supported"; return false; }
+    /// Update status variables changed since last_serial
+    /// @return new last_serial
+    virtual int send_status_updates(send_updates_iface *sui, int last_serial) { return last_serial; }
+    /// Do-nothing destructor to silence compiler warning
+    virtual ~plugin_ctl_iface() {}
+};
+
+struct plugin_list_info_iface;
+
+/// Get a list of all "large" (effect/synthesizer) plugins
+extern void get_all_plugins(std::vector<plugin_metadata_iface *> &plugins);
+/// Get a list of all "small" (module) plugins
+extern void get_all_small_plugins(plugin_list_info_iface *plii);
+/// Load and strdup a text file with GUI definition
+extern const char *load_gui_xml(const std::string &plugin_id);
+
+/// Empty implementations for plugin functions. Note, that functions aren't virtual, because they're called via the particular
+/// subclass (flanger_audio_module etc) via template wrappers (ladspa_wrapper<> etc), not via base class pointer/reference
+template<class Metadata>
+class audio_module: public Metadata
+{
+public:
+    typedef Metadata metadata_type;
+
+    progress_report_iface *progress_report;
+
+    audio_module() {
+        progress_report = NULL;
+    }
+
+    /// Handle MIDI Note On
+    inline void note_on(int note, int velocity) {}
+    /// Handle MIDI Note Off
+    inline void note_off(int note, int velocity) {}
+    /// Handle MIDI Program Change
+    inline void program_change(int program) {}
+    /// Handle MIDI Control Change
+    inline void control_change(int controller, int value) {}
+    /// Handle MIDI Pitch Bend
+    /// @param value pitch bend value (-8192 to 8191, defined as in MIDI ie. 8191 = 200 ct by default)
+    inline void pitch_bend(int value) {}
+    /// Handle MIDI Channel Pressure
+    /// @param value channel pressure (0 to 127)
+    inline void channel_pressure(int value) {}
+    /// Called when params are changed (before processing)
+    inline void params_changed() {}
+    /// LADSPA-esque activate function, except it is called after ports are connected, not before
+    inline void activate() {}
+    /// LADSPA-esque deactivate function
+    inline void deactivate() {}
+    /// Set sample rate for the plugin
+    inline void set_sample_rate(uint32_t sr) { }
+    /// Execute menu command with given number
+    inline void execute(int cmd_no) {}
+    /// DSSI configure call
+    virtual char *configure(const char *key, const char *value) { return NULL; }
+    /// Send all understood configure vars (none by default)
+    inline void send_configures(send_configure_iface *sci) {}
+    /// Send all supported status vars (none by default)
+    inline int send_status_updates(send_updates_iface *sui, int last_serial) { return last_serial; }
+    /// Reset parameter values for epp:trigger type parameters (ones activated by oneshot push button instead of check box)
+    inline void params_reset() {}
+    /// Called after instantiating (after all the feature pointers are set - including interfaces like progress_report_iface)
+    inline void post_instantiate() {}
+    /// Handle 'message context' port message
+    /// @arg output_ports pointer to bit array of output port "changed" flags, note that 0 = first audio input, not first parameter (use input_count + output_count)
+    inline uint32_t message_run(const void *valid_ports, void *output_ports) { 
+        fprintf(stderr, "ERROR: message run not implemented\n");
+        return 0;
+    }
+};
+
+extern bool check_for_message_context_ports(parameter_properties *parameters, int count);
+extern bool check_for_string_ports(parameter_properties *parameters, int count);
+
+#if USE_DSSI
+
+enum line_graph_item
+{
+    LGI_END = 0,
+    LGI_GRAPH,
+    LGI_SUBGRAPH,
+    LGI_LEGEND,
+    LGI_DOT,
+    LGI_END_ITEM,
+    LGI_SET_RGBA,
+    LGI_SET_WIDTH,
+};
+
+/// A class to send status updates via OSC
+struct dssi_feedback_sender
+{
+    /// OSC client object used to send updates
+    osctl::osc_client *client;
+    /// Background thread handle
+    pthread_t bg_thread;
+    /// Quit flag (used to terminate the thread)
+    bool quit;
+    /// Indices of graphs to send
+    std::vector<int> indices;
+    /// Source for the graph data (interface to marshal)
+    calf_plugins::line_graph_iface *graph;
+    
+    dssi_feedback_sender(const char *URI, line_graph_iface *_graph, calf_plugins::parameter_properties *props, int num_params);
+    void update();
+    ~dssi_feedback_sender();
+};
+#endif
+
+/// Metadata base class template, to provide default versions of interface functions
+template<class Metadata>
+class plugin_metadata: public virtual plugin_metadata_iface
+{    
+public:
+    static const char *port_names[];
+    static parameter_properties param_props[];
+    static ladspa_plugin_info plugin_info;
+
+    // These below are stock implementations based on enums and static members in Metadata classes
+    // they may be overridden to provide more interesting functionality
+
+    const char *get_name() { return Metadata::impl_get_name(); } 
+    const char *get_id() { return Metadata::impl_get_id(); } 
+    const char *get_label() { return Metadata::impl_get_label(); } 
+    int get_input_count() { return Metadata::in_count; }
+    int get_output_count() { return Metadata::out_count; }
+    int get_param_count() { return Metadata::param_count; }
+    bool get_midi() { return Metadata::support_midi; }
+    bool requires_midi() { return Metadata::require_midi; }
+    bool is_rt_capable() { return Metadata::rt_capable; }
+    line_graph_iface *get_line_graph_iface() { return dynamic_cast<line_graph_iface *>(this); }    
+    table_edit_iface *get_table_edit_iface() { return dynamic_cast<table_edit_iface *>(this); }    
+    int get_param_port_offset()  { return Metadata::in_count + Metadata::out_count; }
+    const char *get_gui_xml() { static const char *data_ptr = calf_plugins::load_gui_xml(get_id()); return data_ptr; }
+    plugin_command_info *get_commands() { return NULL; }
+    parameter_properties *get_param_props(int param_no) { return &param_props[param_no]; }
+    const char **get_port_names() { return port_names; }
+    bool is_cv(int param_no) { return true; }
+    bool is_noisy(int param_no) { return false; }
+    const ladspa_plugin_info &get_plugin_info() { return plugin_info; }
+    bool requires_message_context() { return check_for_message_context_ports(param_props, Metadata::param_count); }
+    bool requires_string_ports() { return check_for_string_ports(param_props, Metadata::param_count); }
+    void get_message_context_parameters(std::vector<int> &ports) {
+        for (int i = 0; i < get_param_count(); ++i) {
+            if (get_param_props(i)->flags & PF_PROP_MSGCONTEXT)
+                ports.push_back(i);
+        }
+    }
+};
+
+/// A class for delegating metadata implementation to a "remote" metadata class.
+/// Used for GUI wrappers that cannot have a dependency on actual classes,
+/// and which instead take an "external" metadata object pointer, obtained
+/// through get_all_plugins.
+class plugin_metadata_proxy: public virtual plugin_metadata_iface
+{
+public:
+    plugin_metadata_iface *impl;
+public:
+    plugin_metadata_proxy(plugin_metadata_iface *_impl) { impl = _impl; }
+    const char *get_name() { return impl->get_name(); } 
+    const char *get_id() { return impl->get_id(); } 
+    const char *get_label() { return impl->get_label(); } 
+    int get_input_count() { return impl->get_input_count(); }
+    int get_output_count() { return impl->get_output_count(); }
+    int get_param_count() { return impl->get_param_count(); }
+    bool get_midi() { return impl->get_midi(); }
+    bool requires_midi() { return impl->requires_midi(); }
+    bool is_rt_capable() { return impl->is_rt_capable(); }
+    line_graph_iface *get_line_graph_iface() { return impl->get_line_graph_iface(); }    
+    table_edit_iface *get_table_edit_iface() { return impl->get_table_edit_iface(); }    
+    int get_param_port_offset()  { return impl->get_param_port_offset(); }
+    const char *get_gui_xml() { return impl->get_gui_xml(); }
+    plugin_command_info *get_commands() { return impl->get_commands(); }
+    parameter_properties *get_param_props(int param_no) { return impl->get_param_props(param_no); }
+    const char **get_port_names() { return impl->get_port_names(); }
+    bool is_cv(int param_no) { return impl->is_cv(param_no); }
+    bool is_noisy(int param_no) { return impl->is_noisy(param_no); }
+    const ladspa_plugin_info &get_plugin_info() { return impl->get_plugin_info(); }
+    bool requires_message_context() { return impl->requires_message_context(); }
+    bool requires_string_ports() { return impl->requires_string_ports(); }
+    void get_message_context_parameters(std::vector<int> &ports) { impl->get_message_context_parameters(ports); }
+};
+
+#define CALF_PORT_NAMES(name) template<> const char *::plugin_metadata<name##_metadata>::port_names[]
+#define CALF_PORT_PROPS(name) template<> parameter_properties plugin_metadata<name##_metadata>::param_props[]
+#define CALF_PLUGIN_INFO(name) template<> calf_plugins::ladspa_plugin_info plugin_metadata<name##_metadata>::plugin_info
+#define PLUGIN_NAME_ID_LABEL(name, id, label) \
+    static const char *impl_get_name() { return name; } \
+    static const char *impl_get_id() { return id; } \
+    static const char *impl_get_label() { return label; } \
+    
+
+extern const char *calf_copyright_info;
+
+};
+
+#endif

plugins/ladspa_effect/calf/calf/inertia.h

@@ -0,0 +1,256 @@
+/* Calf DSP Library
+ * Basic "inertia" (parameter smoothing) classes.
+ * Copyright (C) 2001-2007 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+#ifndef __CALF_INERTIA_H
+#define __CALF_INERTIA_H
+
+#include "primitives.h"
+
+namespace dsp {
+    
+/// Algorithm for a constant time linear ramp
+class linear_ramp
+{
+public:
+    int ramp_len;
+    float mul, delta;
+public:
+    /// Construct for given ramp length
+    linear_ramp(int _ramp_len) {
+        ramp_len = _ramp_len;
+        mul = (float)(1.0f / ramp_len);
+    }
+    /// Change ramp length
+    inline void set_length(int _ramp_len) {
+        ramp_len = _ramp_len;
+        mul = (float)(1.0f / ramp_len);
+    }
+    inline int length()
+    {
+        return ramp_len;
+    }
+    inline void start_ramp(float start, float end)
+    {
+        delta = mul * (end - start);
+    }
+    /// Return value after single step
+    inline float ramp(float value)
+    {
+        return value + delta;
+    }
+    /// Return value after many steps
+    inline float ramp_many(float value, int count)
+    {
+        return value + delta * count;
+    }
+};
+    
+/// Algorithm for a constant time linear ramp
+class exponential_ramp
+{
+public:
+    int ramp_len;
+    float root, delta;
+public:
+    exponential_ramp(int _ramp_len) {
+        ramp_len = _ramp_len;
+        root = (float)(1.0f / ramp_len);
+    }
+    inline void set_length(int _ramp_len) {
+        ramp_len = _ramp_len;
+        root = (float)(1.0f / ramp_len);
+    }
+    inline int length()
+    {
+        return ramp_len;
+    }
+    inline void start_ramp(float start, float end)
+    {
+        delta = pow(end / start, root);
+    }
+    /// Return value after single step
+    inline float ramp(float value)
+    {
+        return value * delta;
+    }
+    /// Return value after many steps
+    inline float ramp_many(float value, float count)
+    {
+        return value * pow(delta, count);
+    }
+};
+    
+/// Generic inertia using ramping algorithm specified as template argument. The basic idea
+/// is producing smooth(ish) output for discrete input, using specified algorithm to go from
+/// last output value to input value. It is not the same as classic running average lowpass
+/// filter, because ramping time is finite and pre-determined (it calls ramp algorithm's length()
+/// function to obtain the expected ramp length)
+template<class Ramp>
+class inertia
+{
+public:
+    float old_value;
+    float value;
+    unsigned int count;
+    Ramp ramp;
+
+public:
+    inertia(const Ramp &_ramp, float init_value = 0.f)
+    : ramp(_ramp)
+    {
+        value = old_value = init_value;
+        count = 0;
+    }
+    /// Set value immediately (no inertia)
+    void set_now(float _value)
+    {
+        value = old_value = _value;
+        count = 0;
+    }
+    /// Set with inertia
+    void set_inertia(float source)
+    {
+        if (source != old_value) {
+            ramp.start_ramp(value, source);
+            count = ramp.length();
+            old_value = source;
+        }
+    }
+    /// Get smoothed value of given source value
+    inline float get(float source)
+    {
+        if (source != old_value) {
+            ramp.start_ramp(value, source);
+            count = ramp.length();
+            old_value = source;
+        }
+        if (!count)
+            return old_value;
+        value = ramp.ramp(value);
+        count--;
+        if (!count) // finished ramping, set to desired value to get rid of accumulated rounding errors
+            value = old_value;
+        return value;
+    }
+    /// Get smoothed value assuming no new input
+    inline float get()
+    {
+        if (!count)
+            return old_value;
+        value = ramp.ramp(value);
+        count--;
+        if (!count) // finished ramping, set to desired value to get rid of accumulated rounding errors
+            value = old_value;
+        return value;
+    }
+    /// Do one inertia step, without returning the new value and without changing destination value
+    inline void step()
+    {
+        if (count) {
+            value = ramp.ramp(value);
+            count--;
+            if (!count) // finished ramping, set to desired value to get rid of accumulated rounding errors
+                value = old_value;
+        }
+    }
+    /// Do many inertia steps, without returning the new value and without changing destination value
+    inline void step_many(unsigned int steps)
+    {
+        if (steps < count) {
+            // Skip only a part of the current ramping period
+            value = ramp.ramp_many(value, steps);
+            count -= steps;
+            if (!count) // finished ramping, set to desired value to get rid of accumulated rounding errors
+                value = old_value;
+        }
+        else
+        {
+            // The whole ramping period has been skipped, just go to destination
+            value = old_value;
+            count = 0;
+        }
+    }
+    /// Get last smoothed value, without affecting anything
+    inline float get_last() const
+    {
+        return value;
+    }
+    /// Is it still ramping?
+    inline bool active() const
+    {
+        return count > 0;
+    }
+};
+
+class once_per_n
+{
+public:
+    unsigned int frequency;
+    unsigned int left;
+public:
+    once_per_n(unsigned int _frequency)
+    : frequency(_frequency), left(_frequency)
+    {}
+    inline void start()
+    {
+        left = frequency;
+    }
+    /// Set timer to "elapsed" state (elapsed() will return true during next call)
+    inline void signal()
+    {
+        left = 0;
+    }
+    inline unsigned int get(unsigned int desired)
+    {
+        if (desired > left) {
+            desired = left;
+            left = 0;
+            return desired;
+        }
+        left -= desired;
+        return desired;
+    }
+    inline bool elapsed()
+    {
+        if (!left) {
+            left = frequency;
+            return true;
+        }
+        return false;
+    }
+};
+
+class gain_smoothing: public inertia<linear_ramp>
+{
+public:
+    gain_smoothing()
+    : inertia<linear_ramp>(linear_ramp(64))
+    {
+    }
+    void set_sample_rate(int sr)
+    {
+        ramp = linear_ramp(sr / 441);
+    }
+    // to change param, use set_inertia(value)
+    // to read param, use get()
+};
+    
+}
+
+#endif

plugins/ladspa_effect/calf/calf/ladspa_wrap.h

@@ -0,0 +1,516 @@
+/* Calf DSP Library
+ * API wrappers for LADSPA/DSSI
+ *
+ * Copyright (C) 2007-2008 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+#ifndef __CALF_LADSPA_WRAP_H
+#define __CALF_LADSPA_WRAP_H
+
+#if USE_LADSPA
+
+#include <ladspa.h>
+#if USE_DSSI
+#include <dssi.h>
+#endif
+#include "giface.h"
+
+namespace calf_plugins {
+
+template<class Module>
+inline int calc_real_param_count()
+{
+    for (int i=0; i < Module::param_count; i++)
+    {
+        if ((Module::param_props[i].flags & PF_TYPEMASK) >= PF_STRING)
+            return i;
+    }
+    return Module::param_count;
+}
+
+/// A template implementing plugin_ctl_iface for a given plugin
+template<class Module>
+struct ladspa_instance: public Module, public plugin_ctl_iface
+{
+    bool activate_flag;
+#if USE_DSSI
+    dssi_feedback_sender *feedback_sender;
+#endif
+    
+    static int real_param_count()
+    {
+        static int _real_param_count = calc_real_param_count<Module>();
+        return _real_param_count;
+    }
+    ladspa_instance()
+    {
+        for (int i=0; i < Module::in_count; i++)
+            Module::ins[i] = NULL;
+        for (int i=0; i < Module::out_count; i++)
+            Module::outs[i] = NULL;
+        int rpc = real_param_count();
+        for (int i=0; i < rpc; i++)
+            Module::params[i] = NULL;
+        activate_flag = true;
+#if USE_DSSI
+        feedback_sender = NULL;
+#endif
+    }
+    virtual parameter_properties *get_param_props(int param_no)
+    {
+        return &Module::param_props[param_no];
+    }
+    virtual float get_param_value(int param_no)
+    {
+        // XXXKF hack
+        if (param_no >= real_param_count())
+            return 0;
+        return *Module::params[param_no];
+    }
+    virtual void set_param_value(int param_no, float value)
+    {
+        // XXXKF hack
+        if (param_no >= real_param_count())
+            return;
+        *Module::params[param_no] = value;
+    }
+    virtual int get_param_count()
+    {
+        return real_param_count();
+    }
+    virtual int get_param_port_offset() 
+    {
+        return Module::in_count + Module::out_count;
+    }
+    virtual const char *get_gui_xml() {
+        return Module::get_gui_xml();
+    }
+    virtual line_graph_iface *get_line_graph_iface()
+    {
+        return dynamic_cast<line_graph_iface *>(this);
+    }
+    virtual bool activate_preset(int bank, int program) { 
+        return false;
+    }
+    virtual const char *get_name()
+    {
+        return Module::get_name();
+    }
+    virtual const char *get_id()
+    {
+        return Module::get_id();
+    }
+    virtual const char *get_label()
+    {
+        return Module::get_label();
+    }
+    virtual char *configure(const char *key, const char *value)
+    {
+#if USE_DSSI
+        if (!strcmp(key, "OSC:FEEDBACK_URI"))
+        {
+            line_graph_iface *lgi = dynamic_cast<line_graph_iface *>(this);
+            if (!lgi)
+                return NULL;
+            if (*value)
+            {
+                if (feedback_sender) {
+                    delete feedback_sender;
+                    feedback_sender = NULL;
+                }
+                feedback_sender = new dssi_feedback_sender(value, lgi, get_param_props(0), get_param_count());
+            }
+            else
+            {
+                if (feedback_sender) {
+                    delete feedback_sender;
+                    feedback_sender = NULL;
+                }
+            }
+            return NULL;
+        }
+        else 
+        if (!strcmp(key, "OSC:UPDATE"))
+        {
+            if (feedback_sender)
+                feedback_sender->update();
+            return NULL;
+        }
+        else
+#endif
+        if (!strcmp(key, "ExecCommand"))
+        {
+            if (*value)
+            {
+                execute(atoi(value));
+            }
+            return NULL;
+        }
+        return Module::configure(key, value);
+    }
+    virtual int get_input_count() { return Module::in_count; }
+    virtual int get_output_count() { return Module::out_count; }
+    virtual bool get_midi() { return Module::support_midi; }
+    virtual float get_level(unsigned int port) { return 0.f; }
+    virtual void execute(int cmd_no) {
+        Module::execute(cmd_no);
+    }
+    virtual void send_configures(send_configure_iface *sci) { 
+        Module::send_configures(sci);
+    }
+};
+
+/// A wrapper class for plugin class object (there is only one ladspa_wrapper for many instances of the same plugin)
+template<class Module>
+struct ladspa_wrapper
+{
+    typedef ladspa_instance<Module> instance;
+    
+    /// LADSPA descriptor
+    static LADSPA_Descriptor descriptor;
+    /// LADSPA descriptor for DSSI (uses a different name for the plugin, otherwise same as descriptor)
+    static LADSPA_Descriptor descriptor_for_dssi;
+#if USE_DSSI
+    /// Extended DSSI descriptor (points to descriptor_for_dssi for things like name/label/port info etc.)
+    static DSSI_Descriptor dssi_descriptor;
+    static DSSI_Program_Descriptor dssi_default_program;
+
+    static std::vector<plugin_preset> *presets;
+    static std::vector<DSSI_Program_Descriptor> *preset_descs;
+#endif
+    
+    ladspa_wrapper() 
+    {
+        int ins = Module::in_count;
+        int outs = Module::out_count;
+        int params = ladspa_instance<Module>::real_param_count();
+        ladspa_plugin_info &plugin_info = Module::plugin_info;
+        descriptor.UniqueID = plugin_info.unique_id;
+        descriptor.Label = plugin_info.label;
+        descriptor.Name = strdup((std::string(plugin_info.name) + " LADSPA").c_str());
+        descriptor.Maker = plugin_info.maker;
+        descriptor.Copyright = plugin_info.copyright;
+        descriptor.Properties = Module::rt_capable ? LADSPA_PROPERTY_HARD_RT_CAPABLE : 0;
+        descriptor.PortCount = ins + outs + params;
+        descriptor.PortNames = new char *[descriptor.PortCount];
+        descriptor.PortDescriptors = new LADSPA_PortDescriptor[descriptor.PortCount];
+        descriptor.PortRangeHints = new LADSPA_PortRangeHint[descriptor.PortCount];
+        int i;
+        for (i = 0; i < ins + outs; i++)
+        {
+            LADSPA_PortRangeHint &prh = ((LADSPA_PortRangeHint *)descriptor.PortRangeHints)[i];
+            ((int *)descriptor.PortDescriptors)[i] = i < ins ? LADSPA_PORT_INPUT | LADSPA_PORT_AUDIO
+                                                  : i < ins + outs ? LADSPA_PORT_OUTPUT | LADSPA_PORT_AUDIO
+                                                                   : LADSPA_PORT_INPUT | LADSPA_PORT_CONTROL;
+            prh.HintDescriptor = 0;
+            ((const char **)descriptor.PortNames)[i] = Module::port_names[i];
+        }
+        for (; i < ins + outs + params; i++)
+        {
+            LADSPA_PortRangeHint &prh = ((LADSPA_PortRangeHint *)descriptor.PortRangeHints)[i];
+            parameter_properties &pp = Module::param_props[i - ins - outs];
+            ((int *)descriptor.PortDescriptors)[i] = 
+                LADSPA_PORT_CONTROL | (pp.flags & PF_PROP_OUTPUT ? LADSPA_PORT_OUTPUT : LADSPA_PORT_INPUT);
+            prh.HintDescriptor = LADSPA_HINT_BOUNDED_ABOVE | LADSPA_HINT_BOUNDED_BELOW;
+            ((const char **)descriptor.PortNames)[i] = pp.name;
+            prh.LowerBound = pp.min;
+            prh.UpperBound = pp.max;
+            switch(pp.flags & PF_TYPEMASK) {
+                case PF_BOOL: 
+                    prh.HintDescriptor |= LADSPA_HINT_TOGGLED;
+                    prh.HintDescriptor &= ~(LADSPA_HINT_BOUNDED_ABOVE | LADSPA_HINT_BOUNDED_BELOW);
+                    break;
+                case PF_INT: 
+                case PF_ENUM: 
+                    prh.HintDescriptor |= LADSPA_HINT_INTEGER;
+                    break;
+                default: {
+                    int defpt = (int)(100 * (pp.def_value - pp.min) / (pp.max - pp.min));
+                    if ((pp.flags & PF_SCALEMASK) == PF_SCALE_LOG)
+                        defpt = (int)(100 * log(pp.def_value / pp.min) / log(pp.max / pp.min));
+                    if (defpt < 12)
+                        prh.HintDescriptor |= LADSPA_HINT_DEFAULT_MINIMUM;
+                    else if (defpt < 37)
+                        prh.HintDescriptor |= LADSPA_HINT_DEFAULT_LOW;
+                    else if (defpt < 63)
+                        prh.HintDescriptor |= LADSPA_HINT_DEFAULT_MIDDLE;
+                    else if (defpt < 88)
+                        prh.HintDescriptor |= LADSPA_HINT_DEFAULT_HIGH;
+                    else
+                        prh.HintDescriptor |= LADSPA_HINT_DEFAULT_MAXIMUM;
+                }
+            }
+            if (pp.def_value == 0 || pp.def_value == 1 || pp.def_value == 100 || pp.def_value == 440 ) {
+                prh.HintDescriptor &= ~LADSPA_HINT_DEFAULT_MASK;
+                if (pp.def_value == 1)
+                    prh.HintDescriptor |= LADSPA_HINT_DEFAULT_1;
+                else if (pp.def_value == 100)
+                    prh.HintDescriptor |= LADSPA_HINT_DEFAULT_100;
+                else if (pp.def_value == 440)
+                    prh.HintDescriptor |= LADSPA_HINT_DEFAULT_440;
+                else
+                    prh.HintDescriptor |= LADSPA_HINT_DEFAULT_0;
+            }
+            switch(pp.flags & PF_SCALEMASK) {
+                case PF_SCALE_LOG:
+                    prh.HintDescriptor |= LADSPA_HINT_LOGARITHMIC;
+                    break;
+            }
+        }
+        descriptor.ImplementationData = this;
+        descriptor.instantiate = cb_instantiate;
+        descriptor.connect_port = cb_connect;
+        descriptor.activate = cb_activate;
+        descriptor.run = cb_run;
+        descriptor.run_adding = NULL;
+        descriptor.set_run_adding_gain = NULL;
+        descriptor.deactivate = cb_deactivate;
+        descriptor.cleanup = cb_cleanup;
+#if USE_DSSI
+        memcpy(&descriptor_for_dssi, &descriptor, sizeof(descriptor));
+        descriptor_for_dssi.Name = strdup((std::string(plugin_info.name) + " DSSI").c_str());
+        memset(&dssi_descriptor, 0, sizeof(dssi_descriptor));
+        dssi_descriptor.DSSI_API_Version = 1;
+        dssi_descriptor.LADSPA_Plugin = &descriptor_for_dssi;
+        dssi_descriptor.configure = cb_configure;
+        dssi_descriptor.get_program = cb_get_program;
+        dssi_descriptor.select_program = cb_select_program;
+        if (Module::support_midi)
+            dssi_descriptor.run_synth = cb_run_synth;
+        
+        presets = new std::vector<plugin_preset>;
+        preset_descs = new std::vector<DSSI_Program_Descriptor>;
+
+        preset_list plist_tmp, plist;
+        plist.load_defaults(true);
+        plist_tmp.load_defaults(false);
+        plist.presets.insert(plist.presets.end(), plist_tmp.presets.begin(), plist_tmp.presets.end());
+        
+        // XXXKF this assumes that plugin name in preset is case-insensitive equal to plugin label
+        // if I forget about this, I'll be in a deep trouble
+        dssi_default_program.Bank = 0;
+        dssi_default_program.Program = 0;
+        dssi_default_program.Name = "default";
+
+        int pos = 1;
+        for (unsigned int i = 0; i < plist.presets.size(); i++)
+        {
+            plugin_preset &pp = plist.presets[i];
+            if (strcasecmp(pp.plugin.c_str(), descriptor.Label))
+                continue;
+            DSSI_Program_Descriptor pd;
+            pd.Bank = pos >> 7;
+            pd.Program = pos++;
+            pd.Name = pp.name.c_str();
+            preset_descs->push_back(pd);
+            presets->push_back(pp);
+        }
+        // printf("presets = %p:%d name = %s\n", presets, presets->size(), descriptor.Label);
+        
+#endif
+    }
+
+    ~ladspa_wrapper()
+    {
+        delete []descriptor.PortNames;
+        delete []descriptor.PortDescriptors;
+        delete []descriptor.PortRangeHints;
+#if USE_DSSI
+        presets->clear();
+        preset_descs->clear();
+        delete presets;
+        delete preset_descs;
+#endif
+    }
+
+    /// LADSPA instantiation function (create a plugin instance)
+    static LADSPA_Handle cb_instantiate(const struct _LADSPA_Descriptor * Descriptor, unsigned long sample_rate)
+    {
+        instance *mod = new instance();
+        mod->set_sample_rate(sample_rate);
+        mod->post_instantiate();
+        return mod;
+    }
+
+#if USE_DSSI
+    /// DSSI get program descriptor function; for 0, it returns the default program (from parameter properties table), for others, it uses global or user preset
+    static const DSSI_Program_Descriptor *cb_get_program(LADSPA_Handle Instance, unsigned long index) {
+        if (index > presets->size())
+            return NULL;
+        if (index)
+            return &(*preset_descs)[index - 1];
+        return &dssi_default_program;
+    }
+    
+    /// DSSI select program function; for 0, it sets the defaults, for others, it sets global or user preset
+    static void cb_select_program(LADSPA_Handle Instance, unsigned long Bank, unsigned long Program) {
+        instance *mod = (instance *)Instance;
+        unsigned int no = (Bank << 7) + Program - 1;
+        // printf("no = %d presets = %p:%d\n", no, presets, presets->size());
+        if (no == -1U) {
+            int rpc = ladspa_instance<Module>::real_param_count();
+            for (int i =0 ; i < rpc; i++)
+                *mod->params[i] = Module::param_props[i].def_value;
+            return;
+        }
+        if (no >= presets->size())
+            return;
+        plugin_preset &p = (*presets)[no];
+        // printf("activating preset %s\n", p.name.c_str());
+        p.activate(mod);
+    }
+    
+#endif
+    
+    /// LADSPA port connection function
+    static void cb_connect(LADSPA_Handle Instance, unsigned long port, LADSPA_Data *DataLocation) {
+        unsigned long ins = Module::in_count;
+        unsigned long outs = Module::out_count;
+        unsigned long params = ladspa_instance<Module>::real_param_count();
+        instance *const mod = (instance *)Instance;
+        if (port < ins)
+            mod->ins[port] = DataLocation;
+        else if (port < ins + outs)
+            mod->outs[port - ins] = DataLocation;
+        else if (port < ins + outs + params) {
+            int i = port - ins - outs;
+            mod->params[i] = DataLocation;
+            *mod->params[i] = Module::param_props[i].def_value;
+        }
+    }
+
+    /// LADSPA activate function (note that at this moment the ports are not set)
+    static void cb_activate(LADSPA_Handle Instance) {
+        instance *const mod = (instance *)Instance;
+        mod->activate_flag = true;
+    }
+    
+    /// utility function: zero port values if mask is 0
+    static inline void zero_by_mask(Module *module, uint32_t mask, uint32_t offset, uint32_t nsamples)
+    {
+        for (int i=0; i<Module::out_count; i++) {
+            if ((mask & (1 << i)) == 0) {
+                dsp::zero(module->outs[i] + offset, nsamples);
+            }
+        }
+    }
+
+    /// LADSPA run function - does set sample rate / activate logic when it's run first time after activation
+    static void cb_run(LADSPA_Handle Instance, unsigned long SampleCount) {
+        instance *const mod = (instance *)Instance;
+        if (mod->activate_flag)
+        {
+            mod->activate();
+            mod->activate_flag = false;
+        }
+        mod->params_changed();
+        process_slice(mod, 0, SampleCount);
+    }
+    
+    /// utility function: call process, and if it returned zeros in output masks, zero out the relevant output port buffers
+    static inline void process_slice(Module *mod, uint32_t offset, uint32_t end)
+    {
+        while(offset < end)
+        {
+            uint32_t newend = std::min(offset + MAX_SAMPLE_RUN, end);
+            uint32_t out_mask = mod->process(offset, newend - offset, -1, -1);
+            zero_by_mask(mod, out_mask, offset, newend - offset);
+            offset = newend;
+        }
+    }
+
+#if USE_DSSI
+    /// DSSI "run synth" function, same as run() except it allows for event delivery
+    static void cb_run_synth(LADSPA_Handle Instance, unsigned long SampleCount, 
+            snd_seq_event_t *Events, unsigned long EventCount) {
+        instance *const mod = (instance *)Instance;
+        if (mod->activate_flag)
+        {
+            mod->activate();
+            mod->activate_flag = false;
+        }
+        mod->params_changed();
+        
+        uint32_t offset = 0;
+        for (uint32_t e = 0; e < EventCount; e++)
+        {
+            uint32_t timestamp = Events[e].time.tick;
+            if (timestamp != offset)
+                process_slice(mod, offset, timestamp);
+            process_dssi_event(mod, Events[e]);
+            offset = timestamp;
+        }
+        if (offset != SampleCount)
+            process_slice(mod, offset, SampleCount);
+    }
+    
+    /// DSSI configure function (named properties)
+    static char *cb_configure(LADSPA_Handle Instance,
+		       const char *Key,
+		       const char *Value)
+    {
+        instance *const mod = (instance *)Instance;
+        return mod->configure(Key, Value);
+    }
+    
+    /// Utility function: handle MIDI event (only handles a subset in this version)
+    static void process_dssi_event(Module *module, snd_seq_event_t &event)
+    {
+        switch(event.type) {
+            case SND_SEQ_EVENT_NOTEON:
+                module->note_on(event.data.note.note, event.data.note.velocity);
+                break;
+            case SND_SEQ_EVENT_NOTEOFF:
+                module->note_off(event.data.note.note, event.data.note.velocity);
+                break;
+            case SND_SEQ_EVENT_PGMCHANGE:
+                module->program_change(event.data.control.value);
+                break;
+            case SND_SEQ_EVENT_CONTROLLER:
+                module->control_change(event.data.control.param, event.data.control.value);
+                break;
+            case SND_SEQ_EVENT_PITCHBEND:
+                module->pitch_bend(event.data.control.value);
+                break;
+        }
+    }
+#endif
+
+    /// LADSPA deactivate function
+    static void cb_deactivate(LADSPA_Handle Instance) {
+        instance *const mod = (instance *)Instance;
+        mod->deactivate();
+    }
+
+    /// LADSPA cleanup (delete instance) function
+    static void cb_cleanup(LADSPA_Handle Instance) {
+        instance *const mod = (instance *)Instance;
+        delete mod;
+    }
+    
+    /// Get a wrapper singleton - used to prevent initialization order problems which were present in older versions
+    static ladspa_wrapper &get() { 
+        static ladspa_wrapper instance;
+        return instance;
+    }
+};
+
+};
+
+#endif
+
+#endif

plugins/ladspa_effect/calf/calf/loudness.h

@@ -0,0 +1,90 @@
+/* Calf DSP Library
+ * A-weighting filter for 
+ * Copyright (C) 2001-2007 Krzysztof Foltman
+ *
+ * Most of code in this file is based on freely
+ * available other work of other people (filter equations).
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+#ifndef __CALF_LOUDNESS_H
+#define __CALF_LOUDNESS_H
+
+#include "biquad.h"
+
+namespace dsp {
+    
+class aweighter {
+public:
+    biquad_d2<float> bq1, bq2, bq3;
+    
+    /// Produce one output sample from one input sample
+    float process(float sample)
+    {
+        return bq1.process(bq2.process(bq3.process(sample)));
+    }
+    
+    /// Set sample rate (updates filter coefficients)
+    void set(float sr)
+    {
+        // analog coeffs taken from: http://www.diracdelta.co.uk/science/source/a/w/aweighting/source.html
+        // first we need to adjust them by doing some obscene sort of reverse pre-warping (a broken one, too!)
+        float f1 = biquad_coeffs<float>::unwarpf(20.6f, sr);
+        float f2 = biquad_coeffs<float>::unwarpf(107.7f, sr);
+        float f3 = biquad_coeffs<float>::unwarpf(738.f, sr);
+        float f4 = biquad_coeffs<float>::unwarpf(12200.f, sr);
+        // then map s domain to z domain using bilinear transform
+        // note: f1 and f4 are double poles
+        bq1.set_bilinear(0, 0, 1, f1*f1, 2 * f1, 1);
+        bq2.set_bilinear(1, 0, 0, f2*f3, f2 + f3, 1);
+        bq3.set_bilinear(0, 0, 1, f4*f4, 2 * f4, 1);
+        // the coeffs above give non-normalized value, so it should be normalized to produce 0dB at 1 kHz
+        // find actual gain
+        float gain1kHz = freq_gain(1000.0, sr);
+        // divide one filter's x[n-m] coefficients by that value
+        float gc = 1.0 / gain1kHz;
+        bq1.a0 *= gc;
+        bq1.a1 *= gc;
+        bq1.a2 *= gc;
+    }
+    
+    /// Reset to zero if at risk of denormals
+    void sanitize()
+    {
+        bq1.sanitize();
+        bq2.sanitize();
+        bq3.sanitize();
+    }
+    
+    /// Reset state to zero
+    void reset()
+    {
+        bq1.reset();
+        bq2.reset();
+        bq3.reset();
+    }
+    
+    /// Gain and a given frequency
+    float freq_gain(float freq, float sr)
+    {
+        return bq1.freq_gain(freq, sr) * bq2.freq_gain(freq, sr) * bq3.freq_gain(freq, sr);
+    }
+    
+};
+
+};
+
+#endif

plugins/ladspa_effect/calf/calf/lv2helpers.h

@@ -0,0 +1,281 @@
+/* Calf DSP Library
+ * LV2-related helper classes and functions
+ *
+ * Copyright (C) 2001-2008 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+#ifndef CALF_LV2HELPERS_H
+#define CALF_LV2HELPERS_H
+
+#if USE_LV2
+
+#include <calf/lv2_event.h>
+#include <calf/lv2_uri_map.h>
+
+class uri_map_access
+{
+public:
+    /// URI map feature pointer (previously in a mixin, but polymorphic ports made it necessary for most plugins)
+    LV2_URI_Map_Feature *uri_map;
+
+    uri_map_access()
+    : uri_map(NULL)
+    {}
+
+    /// Map an URI through an URI map
+    uint32_t map_uri(const char *ns, const char *URI)
+    {
+        if (uri_map)
+            return uri_map->uri_to_id(uri_map->callback_data, ns, URI);
+        return 0;
+    }
+    /// Called on instantiation for every LV2 feature sent by a host
+    void use_feature(const char *URI, void *data) {
+        if (!strcmp(URI, LV2_URI_MAP_URI))
+        {
+            uri_map = (LV2_URI_Map_Feature *)data;
+            map_uris();
+        }
+    }
+    virtual void map_uris()
+    {
+    }
+    virtual ~uri_map_access() {}
+};
+    
+/// A mixin for adding the event feature and URI map to the small plugin
+template<class T>
+class event_mixin: public T
+{
+public:
+    /// Event feature pointer
+    LV2_Event_Feature *event_feature;
+    virtual void use_feature(const char *URI, void *data) {
+        if (!strcmp(URI, LV2_EVENT_URI))
+        {
+            event_feature = (LV2_Event_Feature *)data;
+        }
+        T::use_feature(URI, data);
+    }
+    /// Create a reference
+    inline void ref_event(LV2_Event *event) { event_feature->lv2_event_ref(event_feature->callback_data, event); }
+    /// Destroy a reference
+    inline void unref_event(LV2_Event *event) { event_feature->lv2_event_unref(event_feature->callback_data, event); }
+};
+
+/// A mixin for adding the URI map and MIDI event type retrieval to small plugins
+template<class T>
+class midi_mixin: public virtual event_mixin<T>
+{
+public:
+    /// MIDI event ID, as resolved using the URI map feature
+    uint32_t midi_event_type;
+    virtual void map_uris() {
+        midi_event_type = this->map_uri("http://lv2plug.in/ns/ext/event", "http://lv2plug.in/ns/ext/midi#MidiEvent");
+        printf("MIDI event type = %d\n", midi_event_type);
+        event_mixin<T>::map_uris();
+    }
+};
+
+/// A mixin for adding the URI map and MIDI event type retrieval to small plugins
+template<class T>
+class message_mixin: public virtual event_mixin<T>
+{
+public:
+    /// MIDI event ID, as resolved using the URI map feature
+    uint32_t message_event_type;
+    virtual void map_uris() {
+        message_event_type = this->map_uri("http://lv2plug.in/ns/ext/event", "http://lv2plug.in/ns/dev/msg#MessageEvent");
+        printf("Message event type = %d\n", message_event_type);
+        event_mixin<T>::map_uris();
+    }
+};
+
+/// LV2 event structure + payload as 0-length array for easy access
+struct lv2_event: public LV2_Event
+{
+    uint8_t data[];
+    inline lv2_event &operator=(const lv2_event &src) {
+        *(LV2_Event *)this = (const LV2_Event &)src;
+        memcpy(data, src.data, src.size);
+        return *this;
+    }
+    /// Returns a 64-bit timestamp for easy and inefficient comparison
+    inline uint64_t timestamp() const {
+        return ((uint64_t)frames << 32) | subframes;
+    }
+private:
+    /// forbid default constructor - this object cannot be constructed, only obtained via cast from LV2_Event* (or &) to lv2_event* (or &)
+    lv2_event() {}
+    /// forbid copy constructor - see default constructor
+    lv2_event(const lv2_event &) {}
+};
+
+/// A read-only iterator-like object for reading from event buffers
+class event_port_read_iterator
+{
+protected:
+    const LV2_Event_Buffer *buffer;
+    uint32_t offset;
+public:
+    /// Default constructor creating a useless iterator you can assign to
+    event_port_read_iterator()
+    : buffer(NULL)
+    , offset(0)
+    {
+    }
+    
+    /// Create an iterator based on specified buffer and index/offset values
+    event_port_read_iterator(const LV2_Event_Buffer *_buffer, uint32_t _offset = 0)
+    : buffer(_buffer)
+    , offset(0)
+    {
+    }
+
+    /// Are any data left to be read?
+    inline operator bool() const {
+        return offset < buffer->size;
+    }
+    
+    /// Read pointer
+    inline const lv2_event &operator*() const {
+        return *(const lv2_event *)(buffer->data + offset);
+    }
+    /// Pointer to member
+    inline const lv2_event *operator->() const {
+        return &**this;
+    }
+
+    /// Move to the next element
+    inline event_port_read_iterator operator++() {
+        offset += ((**this).size + 19) &~7;
+        return *this;
+    }
+
+    /// Move to the next element
+    inline event_port_read_iterator operator++(int) {
+        event_port_read_iterator old = *this;
+        offset += ((**this).size + 19) &~7;
+        return old;
+    }
+};
+
+/// A write-only iterator-like object for writing to event buffers
+class event_port_write_iterator
+{
+protected:
+    LV2_Event_Buffer *buffer;
+public:
+    /// Default constructor creating a useless iterator you can assign to
+    event_port_write_iterator()
+    : buffer(NULL)
+    {
+    }
+    
+    /// Create a write iterator based on specified buffer and index/offset values
+    event_port_write_iterator(LV2_Event_Buffer *_buffer)
+    : buffer(_buffer)
+    {
+    }
+
+    /// @return the remaining buffer space
+    inline uint32_t space_left() const {
+        return buffer->capacity - buffer->size;
+    }
+    /// @return write pointer
+    inline lv2_event &operator*() {
+        return *(lv2_event *)(buffer->data + buffer->size);
+    }
+    /// Pointer to member
+    inline lv2_event *operator->() {
+        return &**this;
+    }
+    /// Move to the next element after the current one has been written (must be called after each write)
+    inline event_port_write_iterator operator++() {
+        buffer->size += ((**this).size + 19) &~7;
+        buffer->event_count ++;
+        return *this;
+    }
+    /// Move to the next element after the current one has been written
+    inline lv2_event *operator++(int) {
+        lv2_event *ptr = &**this;
+        buffer->size += ((**this).size + 19) &~7;
+        buffer->event_count ++;
+        return ptr;
+    }
+};
+
+template<class Iter1, class Iter2>
+class event_port_merge_iterator
+{
+public:
+    Iter1 first;
+    Iter2 second;
+public:
+    event_port_merge_iterator() {}
+    event_port_merge_iterator(const Iter1 &_first, const Iter2 &_second)
+    : first(_first)
+    , second(_second)
+    {
+    }
+    /// @retval true if any of the iterators have any data left
+    inline operator bool() const {
+        return ((bool)first) || ((bool)second);
+    }
+    inline bool select_first() const
+    {
+        if (!(bool)second)
+            return true;
+        if (!(bool)first)
+            return false;
+        return first->timestamp() < second->timestamp();
+    }
+    /// Returns the earliest of (*first, *second)
+    inline const lv2_event &operator*() const {
+        if (select_first())
+        {
+            assert((bool)first);
+            return *first;
+        }
+        assert((bool)second);
+        return *second;
+    }
+    /// Pointer to member
+    inline const lv2_event *operator->() const {
+        return &**this;
+    }
+    /// Prefix increment
+    inline event_port_merge_iterator operator++() {
+        if (select_first())
+            first++;
+        else
+            second++;
+        return *this;
+    }
+    /// Postfix increment
+    inline event_port_merge_iterator operator++(int) {
+        event_port_merge_iterator ptr = *this;
+        if (select_first())
+            first++;
+        else
+            second++;
+        return ptr;
+    }
+};
+
+#endif
+#endif

plugins/ladspa_effect/calf/calf/lv2wrap.h

@@ -0,0 +1,349 @@
+/* Calf DSP Library
+ * LV2 wrapper templates
+ *
+ * Copyright (C) 2001-2008 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+#ifndef CALF_LV2WRAP_H
+#define CALF_LV2WRAP_H
+
+#if USE_LV2
+
+#include <string>
+#include <vector>
+#include <lv2.h>
+#include <calf/giface.h>
+#include <calf/lv2-midiport.h>
+#include <calf/lv2_contexts.h>
+#include <calf/lv2_event.h>
+#include <calf/lv2_progress.h>
+#include <calf/lv2_string_port.h>
+#include <calf/lv2_uri_map.h>
+
+namespace calf_plugins {
+
+template<class Module>
+struct lv2_instance: public plugin_ctl_iface, public progress_report_iface, public Module
+{
+    bool set_srate;
+    int srate_to_set;
+    LV2_MIDI *midi_data;
+    LV2_Event_Buffer *event_data;
+    LV2_URI_Map_Feature *uri_map;
+    LV2_Event_Feature *event_feature;
+    uint32_t midi_event_type;
+    std::vector<int> message_params;
+    LV2_Progress *progress_report_feature;
+    lv2_instance()
+    {
+        for (int i=0; i < Module::in_count; i++)
+            Module::ins[i] = NULL;
+        for (int i=0; i < Module::out_count; i++)
+            Module::outs[i] = NULL;
+        for (int i=0; i < Module::param_count; i++)
+            Module::params[i] = NULL;
+        uri_map = NULL;
+        midi_data = NULL;
+        event_data = NULL;
+        midi_event_type = 0xFFFFFFFF;
+        set_srate = true;
+        srate_to_set = 44100;
+        get_message_context_parameters(message_params);
+        progress_report_feature = NULL;
+        // printf("message params %d\n", (int)message_params.size());
+    }
+    /// This, and not Module::post_instantiate, is actually called by lv2_wrapper class
+    void post_instantiate()
+    {
+        if (progress_report_feature)
+            Module::progress_report = this;
+        Module::post_instantiate();
+    }
+    virtual parameter_properties *get_param_props(int param_no)
+    {
+        return &Module::param_props[param_no];
+    }
+    virtual float get_param_value(int param_no)
+    {
+        return *Module::params[param_no];
+    }
+    virtual void set_param_value(int param_no, float value)
+    {
+        *Module::params[param_no] = value;
+    }
+    virtual int get_param_count()
+    {
+        return Module::param_count;
+    }
+    virtual int get_param_port_offset() 
+    {
+        return Module::in_count + Module::out_count;
+    }
+    virtual const char *get_gui_xml() {
+        return Module::get_gui_xml();
+    }
+    virtual line_graph_iface *get_line_graph_iface()
+    {
+        return dynamic_cast<line_graph_iface *>(this);
+    }
+    virtual bool activate_preset(int bank, int program) { 
+        return false;
+    }
+    virtual const char *get_name()
+    {
+        return Module::get_name();
+    }
+    virtual const char *get_id()
+    {
+        return Module::get_id();
+    }
+    virtual const char *get_label()
+    {
+        return Module::get_label();
+    }
+    virtual int get_input_count() { return Module::in_count; }
+    virtual int get_output_count() { return Module::out_count; }
+    virtual bool get_midi() { return Module::support_midi; }
+    virtual float get_level(unsigned int port) { return 0.f; }
+    virtual void execute(int cmd_no) {
+        Module::execute(cmd_no);
+    }
+    virtual void report_progress(float percentage, const std::string &message) {
+        if (progress_report_feature)
+            (*progress_report_feature->progress)(progress_report_feature->context, percentage, !message.empty() ? message.c_str() : NULL);
+    }
+    void send_configures(send_configure_iface *sci) { 
+        Module::send_configures(sci);
+    }
+    uint32_t impl_message_run(const void *valid_inputs, void *output_ports) {
+        for (unsigned int i = 0; i < message_params.size(); i++)
+        {
+            int pn = message_params[i];
+            parameter_properties &pp = *get_param_props(pn);
+            if ((pp.flags & PF_TYPEMASK) == PF_STRING
+                && (((LV2_String_Data *)Module::params[pn])->flags & LV2_STRING_DATA_CHANGED_FLAG)) {
+                printf("Calling configure on %s\n", pp.short_name);
+                configure(pp.short_name, ((LV2_String_Data *)Module::params[pn])->data);
+            }
+        }
+        return Module::message_run(valid_inputs, output_ports);
+    }
+    char *configure(const char *key, const char *value) { 
+        // disambiguation - the plugin_ctl_iface version is just a stub, so don't use it
+        return Module::configure(key, value);
+    }
+#if 0
+    // the default implementation should be fine
+    virtual void clear_preset() {
+        // This is never called in practice, at least for now
+        // However, it will change when presets are implemented
+        for (int i=0; i < Module::param_count; i++)
+            *Module::params[i] = Module::param_props[i].def_value;
+        /*
+        const char **p = Module::get_default_configure_vars();
+        if (p)
+        {
+            for(; p[0]; p += 2)
+                configure(p[0], p[1]);
+        }
+        */
+    }
+#endif
+};
+
+struct LV2_Calf_Descriptor {
+    plugin_ctl_iface *(*get_pci)(LV2_Handle Instance);
+};
+
+template<class Module>
+struct lv2_wrapper
+{
+    typedef lv2_instance<Module> instance;
+    static LV2_Descriptor descriptor;
+    static LV2_Calf_Descriptor calf_descriptor;
+    static LV2MessageContext message_context;
+    std::string uri;
+    
+    lv2_wrapper()
+    {
+        ladspa_plugin_info &info = Module::plugin_info;
+        uri = "http://calf.sourceforge.net/plugins/" + std::string(info.label);
+        descriptor.URI = uri.c_str();
+        descriptor.instantiate = cb_instantiate;
+        descriptor.connect_port = cb_connect;
+        descriptor.activate = cb_activate;
+        descriptor.run = cb_run;
+        descriptor.deactivate = cb_deactivate;
+        descriptor.cleanup = cb_cleanup;
+        descriptor.extension_data = cb_ext_data;
+        calf_descriptor.get_pci = cb_get_pci;
+        message_context.message_connect_port = cb_connect;
+        message_context.message_run = cb_message_run;
+    }
+
+    static void cb_connect(LV2_Handle Instance, uint32_t port, void *DataLocation) {
+        unsigned long ins = Module::in_count;
+        unsigned long outs = Module::out_count;
+        unsigned long params = Module::param_count;
+        instance *const mod = (instance *)Instance;
+        if (port < ins)
+            mod->ins[port] = (float *)DataLocation;
+        else if (port < ins + outs)
+            mod->outs[port - ins] = (float *)DataLocation;
+        else if (port < ins + outs + params) {
+            int i = port - ins - outs;
+            mod->params[i] = (float *)DataLocation;
+        }
+        else if (Module::support_midi && port == ins + outs + params) {
+            mod->event_data = (LV2_Event_Buffer *)DataLocation;
+        }
+    }
+
+    static void cb_activate(LV2_Handle Instance) {
+        instance *const mod = (instance *)Instance;
+        mod->set_srate = true;
+    }
+    
+    static void cb_deactivate(LV2_Handle Instance) {
+        instance *const mod = (instance *)Instance;
+        mod->deactivate();
+    }
+
+    static LV2_Handle cb_instantiate(const LV2_Descriptor * Descriptor, double sample_rate, const char *bundle_path, const LV2_Feature *const *features)
+    {
+        instance *mod = new instance();
+        // XXXKF some people use fractional sample rates; we respect them ;-)
+        mod->srate_to_set = (uint32_t)sample_rate;
+        mod->set_srate = true;
+        while(*features)
+        {
+            if (!strcmp((*features)->URI, LV2_URI_MAP_URI))
+            {
+                mod->uri_map = (LV2_URI_Map_Feature *)((*features)->data);
+                mod->midi_event_type = mod->uri_map->uri_to_id(
+                    mod->uri_map->callback_data, 
+                    "http://lv2plug.in/ns/ext/event",
+                    "http://lv2plug.in/ns/ext/midi#MidiEvent");
+            }
+            else if (!strcmp((*features)->URI, LV2_EVENT_URI))
+            {
+                mod->event_feature = (LV2_Event_Feature *)((*features)->data);
+            }
+            else if (!strcmp((*features)->URI, LV2_PROGRESS_URI))
+            {
+                mod->progress_report_feature = (LV2_Progress *)((*features)->data);
+            }
+            features++;
+        }
+        mod->post_instantiate();
+        return mod;
+    }
+    static inline void zero_by_mask(Module *module, uint32_t mask, uint32_t offset, uint32_t nsamples)
+    {
+        for (int i=0; i<Module::out_count; i++) {
+            if ((mask & (1 << i)) == 0) {
+                dsp::zero(module->outs[i] + offset, nsamples);
+            }
+        }
+    }
+    static plugin_ctl_iface *cb_get_pci(LV2_Handle Instance)
+    {
+        return static_cast<plugin_ctl_iface *>(Instance);
+    }
+
+    static inline void process_slice(Module *mod, uint32_t offset, uint32_t end)
+    {
+        while(offset < end)
+        {
+            uint32_t newend = std::min(offset + MAX_SAMPLE_RUN, end);
+            uint32_t out_mask = mod->process(offset, newend - offset, -1, -1);
+            zero_by_mask(mod, out_mask, offset, newend - offset);
+            offset = newend;
+        }
+    }
+
+    static uint32_t cb_message_run(LV2_Handle Instance, const void *valid_inputs, void *outputs_written) {
+        instance *mod = (instance *)Instance;
+        return mod->impl_message_run(valid_inputs, outputs_written);
+    }
+    static void cb_run(LV2_Handle Instance, uint32_t SampleCount) {
+        instance *const mod = (instance *)Instance;
+        if (mod->set_srate) {
+            mod->set_sample_rate(mod->srate_to_set);
+            mod->activate();
+            mod->set_srate = false;
+        }
+        mod->params_changed();
+        uint32_t offset = 0;
+        if (mod->event_data)
+        {
+            // printf("Event data: count %d\n", mod->event_data->event_count);
+            struct LV2_Midi_Event: public LV2_Event {
+                unsigned char data[1];
+            };
+            unsigned char *data = (unsigned char *)(mod->event_data->data);
+            for (uint32_t i = 0; i < mod->event_data->event_count; i++) {
+                LV2_Midi_Event *item = (LV2_Midi_Event *)data;
+                uint32_t ts = item->frames;
+                // printf("Event: timestamp %d subframes %d type %d vs %d\n", item->frames, item->subframes, item->type, mod->midi_event_type);
+                if (ts > offset)
+                {
+                    process_slice(mod, offset, ts);
+                    offset = ts;
+                }
+                if (item->type == mod->midi_event_type) 
+                {
+                    // printf("Midi message %x %x %x %x %d\n", item->data[0], item->data[1], item->data[2], item->data[3], item->size);
+                    switch(item->data[0] >> 4)
+                    {
+                    case 8: mod->note_off(item->data[1], item->data[2]); break;
+                    case 9: mod->note_on(item->data[1], item->data[2]); break;
+                    case 11: mod->control_change(item->data[1], item->data[2]); break;
+                    case 12: mod->program_change(item->data[1]); break;
+                    case 14: mod->pitch_bend(item->data[1] + 128 * item->data[2] - 8192); break;
+                    }
+                }
+                else
+                if (item->type == 0 && mod->event_feature)
+                    mod->event_feature->lv2_event_unref(mod->event_feature->callback_data, item);
+                // printf("timestamp %f item size %d first byte %x\n", item->timestamp, item->size, item->data[0]);
+                data += ((sizeof(LV2_Event) + item->size + 7))&~7;
+            }
+        }
+        process_slice(mod, offset, SampleCount);
+    }
+    static void cb_cleanup(LV2_Handle Instance) {
+        instance *const mod = (instance *)Instance;
+        delete mod;
+    }
+    static const void *cb_ext_data(const char *URI) {
+        if (!strcmp(URI, "http://foltman.com/ns/calf-plugin-instance"))
+            return &calf_descriptor;
+        if (!strcmp(URI, LV2_CONTEXT_MESSAGE))
+            return &message_context;
+        return NULL;
+    }
+    static lv2_wrapper &get() { 
+        static lv2_wrapper instance;
+        return instance;
+    }
+};
+
+};
+
+#endif
+#endif

plugins/ladspa_effect/calf/calf/metadata.h

@@ -0,0 +1,312 @@
+/* Calf DSP Library
+ * Audio module (plugin) metadata - header file
+ *
+ * Copyright (C) 2007-2008 Krzysztof Foltman
+ * Copyright (C) 2008 Thor Harald Johansen <thj@thj.no>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __CALF_METADATA_H
+#define __CALF_METADATA_H
+
+#include "giface.h"
+
+namespace calf_plugins {
+
+struct flanger_metadata: public plugin_metadata<flanger_metadata>
+{
+public:
+    enum { par_delay, par_depth, par_rate, par_fb, par_stereo, par_reset, par_amount, par_dryamount, param_count };
+    enum { in_count = 2, out_count = 2, support_midi = false, require_midi = false, rt_capable = true };
+    PLUGIN_NAME_ID_LABEL("flanger", "flanger", "Flanger")
+};
+
+struct phaser_metadata: public plugin_metadata<phaser_metadata>
+{
+    enum { par_freq, par_depth, par_rate, par_fb, par_stages, par_stereo, par_reset, par_amount, par_dryamount, param_count };
+    enum { in_count = 2, out_count = 2, support_midi = false, require_midi = false, rt_capable = true };
+    PLUGIN_NAME_ID_LABEL("phaser", "phaser", "Phaser")
+};
+
+struct filter_metadata: public plugin_metadata<filter_metadata>
+{
+    enum { par_cutoff, par_resonance, par_mode, par_inertia, param_count };
+    enum { in_count = 2, out_count = 2, rt_capable = true, require_midi = false, support_midi = false };
+    PLUGIN_NAME_ID_LABEL("filter", "filter", "Filter")
+    /// do not export mode and inertia as CVs, as those are settings and not parameters
+    bool is_cv(int param_no) { return param_no != par_mode && param_no != par_inertia; }
+};
+
+/// Filterclavier - metadata
+struct filterclavier_metadata: public plugin_metadata<filterclavier_metadata>
+{
+    enum { par_transpose, par_detune, par_max_resonance, par_mode, par_inertia,  param_count };
+    enum { in_count = 2, out_count = 2, rt_capable = true, require_midi = true, support_midi = true };
+    PLUGIN_NAME_ID_LABEL("filterclavier", "filterclavier", "Filterclavier")
+    /// do not export mode and inertia as CVs, as those are settings and not parameters
+    bool is_cv(int param_no) { return param_no != par_mode && param_no != par_inertia; }
+};
+
+struct reverb_metadata: public plugin_metadata<reverb_metadata>
+{
+    enum { par_decay, par_hfdamp, par_roomsize, par_diffusion, par_amount, par_dry, par_predelay, par_basscut, par_treblecut, param_count };
+    enum { in_count = 2, out_count = 2, support_midi = false, require_midi = false, rt_capable = true };
+    PLUGIN_NAME_ID_LABEL("reverb", "reverb", "Reverb")
+};
+
+struct vintage_delay_metadata: public plugin_metadata<vintage_delay_metadata>
+{
+    enum { par_bpm, par_divide, par_time_l, par_time_r, par_feedback, par_amount, par_mixmode, par_medium, par_dryamount, param_count };
+    enum { in_count = 2, out_count = 2, rt_capable = true, support_midi = false, require_midi = false };
+    PLUGIN_NAME_ID_LABEL("vintage_delay", "vintagedelay", "Vintage Delay")
+};
+
+struct rotary_speaker_metadata: public plugin_metadata<rotary_speaker_metadata>
+{
+public:
+    enum { par_speed, par_spacing, par_shift, par_moddepth, par_treblespeed, par_bassspeed, par_micdistance, par_reflection, param_count };
+    enum { in_count = 2, out_count = 2, support_midi = true, require_midi = false, rt_capable = true };
+    PLUGIN_NAME_ID_LABEL("rotary_speaker", "rotaryspeaker", "Rotary Speaker")
+};
+
+/// A multitap stereo chorus thing - metadata
+struct multichorus_metadata: public plugin_metadata<multichorus_metadata>
+{
+public:    
+    enum { par_delay, par_depth, par_rate, par_stereo, par_voices, par_vphase, par_amount, par_dryamount, par_freq, par_freq2, par_q, par_overlap, param_count };
+    enum { in_count = 2, out_count = 2, rt_capable = true, support_midi = false, require_midi = false };
+    PLUGIN_NAME_ID_LABEL("multichorus", "multichorus", "Multi Chorus")
+};
+
+/// Monosynth - metadata
+struct monosynth_metadata: public plugin_metadata<monosynth_metadata>
+{
+    enum { wave_saw, wave_sqr, wave_pulse, wave_sine, wave_triangle, wave_varistep, wave_skewsaw, wave_skewsqr, wave_test1, wave_test2, wave_test3, wave_test4, wave_test5, wave_test6, wave_test7, wave_test8, wave_count };
+    enum { flt_lp12, flt_lp24, flt_2lp12, flt_hp12, flt_lpbr, flt_hpbr, flt_bp6, flt_2bp6 };
+    enum { par_wave1, par_wave2, par_pw1, par_pw2, par_detune, par_osc2xpose, par_oscmode, par_oscmix, par_filtertype, par_cutoff, par_resonance, par_cutoffsep, par_envmod, par_envtores, par_envtoamp, par_attack, par_decay, par_sustain, par_fade, par_release, 
+        par_keyfollow, par_legato, par_portamento, par_vel2filter, par_vel2amp, par_master, par_pwhlrange, 
+        par_lforate, par_lfodelay, par_lfofilter, par_lfopitch, par_lfopw, par_mwhl_lfo, par_scaledetune,
+        param_count };
+    enum { in_count = 0, out_count = 2, support_midi = true, require_midi = true, rt_capable = true };
+    enum { step_size = 64, step_shift = 6 };
+    enum {
+        modsrc_none,
+        modsrc_velocity,
+        modsrc_pressure,
+        modsrc_modwheel,
+        modsrc_env1,
+        modsrc_lfo1,
+        modsrc_count,
+    };
+    enum {
+        moddest_none,
+        moddest_attenuation,
+        moddest_oscmix,
+        moddest_cutoff,
+        moddest_resonance,
+        moddest_o1detune,
+        moddest_o2detune,
+        moddest_o1pw,
+        moddest_o2pw,
+        moddest_count,
+    };
+    PLUGIN_NAME_ID_LABEL("monosynth", "monosynth", "Monosynth")
+};
+    
+/// Thor's compressor - metadata
+struct compressor_metadata: public plugin_metadata<compressor_metadata>
+{
+    enum { in_count = 2, out_count = 2, support_midi = false, require_midi = false, rt_capable = true };
+    enum { param_threshold, param_ratio, param_attack, param_release, param_makeup, param_knee, param_detection, param_stereo_link, param_aweighting, param_compression, param_peak, param_clip, param_bypass, // param_freq, param_bw, 
+        param_count };
+    PLUGIN_NAME_ID_LABEL("compressor", "compressor", "Compressor")
+};
+
+/// Organ - enums for parameter IDs etc. (this mess is caused by organ split between plugin and generic class - which was
+/// a bad design decision and should be sorted out some day) XXXKF @todo
+struct organ_enums
+{
+    enum { 
+        par_drawbar1, par_drawbar2, par_drawbar3, par_drawbar4, par_drawbar5, par_drawbar6, par_drawbar7, par_drawbar8, par_drawbar9, 
+        par_frequency1, par_frequency2, par_frequency3, par_frequency4, par_frequency5, par_frequency6, par_frequency7, par_frequency8, par_frequency9, 
+        par_waveform1, par_waveform2, par_waveform3, par_waveform4, par_waveform5, par_waveform6, par_waveform7, par_waveform8, par_waveform9, 
+        par_detune1, par_detune2, par_detune3, par_detune4, par_detune5, par_detune6, par_detune7, par_detune8, par_detune9, 
+        par_phase1, par_phase2, par_phase3, par_phase4, par_phase5, par_phase6, par_phase7, par_phase8, par_phase9, 
+        par_pan1, par_pan2, par_pan3, par_pan4, par_pan5, par_pan6, par_pan7, par_pan8, par_pan9, 
+        par_routing1, par_routing2, par_routing3, par_routing4, par_routing5, par_routing6, par_routing7, par_routing8, par_routing9, 
+        par_foldover,
+        par_percdecay, par_perclevel, par_percwave, par_percharm, par_percvel2amp,
+        par_percfmdecay, par_percfmdepth, par_percfmwave, par_percfmharm, par_percvel2fm,
+        par_perctrigger, par_percstereo,
+        par_filterchain,
+        par_filter1type,
+        par_master, 
+        par_f1cutoff, par_f1res, par_f1env1, par_f1env2, par_f1env3, par_f1keyf,
+        par_f2cutoff, par_f2res, par_f2env1, par_f2env2, par_f2env3, par_f2keyf,
+        par_eg1attack, par_eg1decay, par_eg1sustain, par_eg1release, par_eg1velscl, par_eg1ampctl, 
+        par_eg2attack, par_eg2decay, par_eg2sustain, par_eg2release, par_eg2velscl, par_eg2ampctl, 
+        par_eg3attack, par_eg3decay, par_eg3sustain, par_eg3release, par_eg3velscl, par_eg3ampctl, 
+        par_lforate, par_lfoamt, par_lfowet, par_lfophase, par_lfomode,
+        par_transpose, par_detune,
+        par_polyphony,
+        par_quadenv,
+        par_pwhlrange,
+        par_bassfreq,
+        par_bassgain,
+        par_treblefreq,
+        par_treblegain,
+        par_var_mapcurve,
+        param_count
+    };
+    enum {
+        var_count = 1
+    };
+    enum organ_waveform { 
+        wave_sine, 
+        wave_sinepl1, wave_sinepl2, wave_sinepl3,
+        wave_ssaw, wave_ssqr, wave_spls, wave_saw, wave_sqr, wave_pulse, wave_sinepl05, wave_sqr05, wave_halfsin, wave_clvg, wave_bell, wave_bell2,
+        wave_w1, wave_w2, wave_w3, wave_w4, wave_w5, wave_w6, wave_w7, wave_w8, wave_w9,
+        wave_dsaw, wave_dsqr, wave_dpls,
+        wave_count_small,
+        wave_strings = wave_count_small,
+        wave_strings2,
+        wave_sinepad,
+        wave_bellpad,
+        wave_space,
+        wave_choir,
+        wave_choir2,
+        wave_choir3,
+        wave_count,
+        wave_count_big = wave_count - wave_count_small
+    };
+    enum {
+        ampctl_none,
+        ampctl_direct,
+        ampctl_f1,
+        ampctl_f2,
+        ampctl_all,
+        ampctl_count
+    };
+    enum { 
+        lfomode_off = 0,
+        lfomode_direct,
+        lfomode_filter1,
+        lfomode_filter2,
+        lfomode_voice,
+        lfomode_global,
+        lfomode_count
+    };
+    enum {
+        perctrig_first = 0,
+        perctrig_each,
+        perctrig_eachplus,
+        perctrig_polyphonic,
+        perctrig_count
+    };
+};
+
+/// Organ - metadata
+struct organ_metadata: public organ_enums, public plugin_metadata<organ_metadata>
+{
+    enum { in_count = 0, out_count = 2, support_midi = true, require_midi = true, rt_capable = true };
+    PLUGIN_NAME_ID_LABEL("organ", "organ", "Organ")
+    plugin_command_info *get_commands();
+    const char **get_default_configure_vars();
+};
+
+/// FluidSynth - metadata
+struct fluidsynth_metadata: public plugin_metadata<fluidsynth_metadata>
+{
+    enum { par_master, par_soundfont, par_interpolation, par_reverb, par_chorus, param_count };
+    enum { in_count = 0, out_count = 2, support_midi = true, require_midi = true, rt_capable = false };
+    PLUGIN_NAME_ID_LABEL("fluidsynth", "fluidsynth", "Fluidsynth")
+    const char **get_default_configure_vars();
+};
+    
+/// Wavetable - metadata
+struct wavetable_metadata: public plugin_metadata<wavetable_metadata>
+{
+    enum {
+        wt_fmshiny,
+        wt_fmshiny2,
+        wt_rezo,
+        wt_metal,
+        wt_bell,
+        wt_blah,
+        wt_pluck,
+        wt_stretch,
+        wt_stretch2,
+        wt_hardsync,
+        wt_hardsync2,
+        wt_softsync,
+        wt_bell2,
+        wt_bell3,
+        wt_tine,
+        wt_tine2,
+        wt_clav,
+        wt_clav2,
+        wt_gtr,
+        wt_gtr2,
+        wt_gtr3,
+        wt_gtr4,
+        wt_gtr5,
+        wt_reed,
+        wt_reed2,
+        wt_silver,
+        wt_brass,
+        wt_multi,
+        wt_multi2,
+        wt_count
+    };
+    enum {
+        modsrc_none,
+        modsrc_velocity,
+        modsrc_pressure,
+        modsrc_modwheel,
+        modsrc_env1,
+        modsrc_env2,
+        modsrc_env3,
+        modsrc_count,
+    };
+    enum {
+        moddest_none,
+        moddest_attenuation,
+        moddest_oscmix,
+        moddest_cutoff,
+        moddest_resonance,
+        moddest_o1shift,
+        moddest_o2shift,
+        moddest_o1detune,
+        moddest_o2detune,
+        moddest_count,
+    };
+    enum { 
+        par_o1wave, par_o1offset, par_o1transpose, par_o1detune, par_o1level,
+        par_o2wave, par_o2offset, par_o2transpose, par_o2detune, par_o2level,
+        par_eg1attack, par_eg1decay, par_eg1sustain, par_eg1fade, par_eg1release, par_eg1velscl,
+        par_eg2attack, par_eg2decay, par_eg2sustain, par_eg2fade, par_eg2release, par_eg2velscl,
+        par_eg3attack, par_eg3decay, par_eg3sustain, par_eg3fade, par_eg3release, par_eg3velscl,
+        par_pwhlrange, 
+        param_count };
+    enum { in_count = 0, out_count = 2, support_midi = true, require_midi = true, rt_capable = true };
+    enum { step_size = 64 };
+    PLUGIN_NAME_ID_LABEL("wavetable", "wavetable", "Wavetable")
+};
+    
+};
+
+#endif

plugins/ladspa_effect/calf/calf/modmatrix.h

@@ -0,0 +1,112 @@
+/* Calf DSP Library
+ * Modulation matrix boilerplate code.
+ *
+ * Copyright (C) 2009 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+#ifndef __CALF_MODMATRIX_H
+#define __CALF_MODMATRIX_H
+ 
+#include "giface.h"
+
+namespace dsp {
+
+/// Mapping modes
+enum mapping_mode {
+    map_positive, ///< 0..100%
+    map_bipolar, ///< -100%..100%
+    map_negative, ///< -100%..0%
+    map_squared, ///< x^2
+    map_squared_bipolar, ///< x^2 scaled to -100%..100%
+    map_antisquared, ///< 1-(1-x)^2 scaled to 0..100%
+    map_antisquared_bipolar, ///< 1-(1-x)^2 scaled to -100..100%
+    map_parabola, ///< inverted parabola (peaks at 0.5, then decreases to 0)
+    map_type_count
+};
+
+/// Single entry in modulation matrix
+struct modulation_entry
+{
+    /// Mapped source
+    int src1;
+    /// Source mapping mode
+    mapping_mode mapping;
+    /// Unmapped modulating source
+    int src2;
+    /// Modulation amount
+    float amount;
+    /// Modulation destination
+    int dest;
+    
+    modulation_entry() {
+        reset();
+    }
+    
+    /// Reset the row to default
+    void reset() {
+        src1 = 0;
+        src2 = 0;
+        mapping = map_positive;
+        amount = 0.f;
+        dest = 0;
+    }
+};
+
+};
+
+namespace calf_plugins {
+
+class mod_matrix: public table_edit_iface
+{
+protected:
+    /// Polynomials for different scaling modes (1, x, x^2)
+    static const float scaling_coeffs[dsp::map_type_count][3];
+    /// Column descriptions for table widget
+    table_column_info table_columns[6];
+    
+    dsp::modulation_entry *matrix;
+    unsigned int matrix_rows;
+    const char **mod_src_names, **mod_dest_names;
+
+    mod_matrix(dsp::modulation_entry *_matrix, unsigned int _rows, const char **_src_names, const char **_dest_names);
+public:
+    virtual const table_column_info *get_table_columns(int param);
+    virtual uint32_t get_table_rows(int param);
+    virtual std::string get_cell(int param, int row, int column);
+    virtual void set_cell(int param, int row, int column, const std::string &src, std::string &error);
+
+    /// Process modulation matrix, calculate outputs from inputs
+    inline void calculate_modmatrix(float *moddest, int moddest_count, float *modsrc)
+    {
+        for (int i = 0; i < moddest_count; i++)
+            moddest[i] = 0;
+        for (unsigned int i = 0; i < matrix_rows; i++)
+        {
+            dsp::modulation_entry &slot = matrix[i];
+            if (slot.dest) {
+                float value = modsrc[slot.src1];
+                const float *c = scaling_coeffs[slot.mapping];
+                value = c[0] + c[1] * value + c[2] * value * value;
+                moddest[slot.dest] += value * modsrc[slot.src2] * slot.amount;
+            }
+        }
+    }
+};
+    
+};
+
+#endif

plugins/ladspa_effect/calf/calf/modulelist.h

@@ -0,0 +1,95 @@
+#ifdef PER_MODULE_ITEM
+    PER_MODULE_ITEM(filter, false, "filter")
+    PER_MODULE_ITEM(filterclavier, false, "filterclavier")
+    PER_MODULE_ITEM(flanger, false, "flanger")
+    PER_MODULE_ITEM(reverb, false, "reverb")
+    PER_MODULE_ITEM(monosynth, true, "monosynth")
+    PER_MODULE_ITEM(vintage_delay, false, "vintagedelay")
+    PER_MODULE_ITEM(organ, true, "organ")
+    PER_MODULE_ITEM(rotary_speaker, false, "rotaryspeaker")
+    PER_MODULE_ITEM(phaser, false, "phaser")
+    PER_MODULE_ITEM(multichorus, false, "multichorus")
+    PER_MODULE_ITEM(compressor, false, "compressor")
+#ifdef ENABLE_EXPERIMENTAL
+    PER_MODULE_ITEM(fluidsynth, true, "fluidsynth")
+    PER_MODULE_ITEM(wavetable, true, "wavetable")
+#endif
+#undef PER_MODULE_ITEM
+#endif
+#ifdef PER_SMALL_MODULE_ITEM
+#ifdef ENABLE_EXPERIMENTAL
+    PER_SMALL_MODULE_ITEM(lp_filter, "lowpass12")
+    PER_SMALL_MODULE_ITEM(hp_filter, "highpass12")
+    PER_SMALL_MODULE_ITEM(bp_filter, "bandpass6")
+    PER_SMALL_MODULE_ITEM(br_filter, "notch6")
+    PER_SMALL_MODULE_ITEM(onepole_lp_filter, "lowpass6")
+    PER_SMALL_MODULE_ITEM(onepole_hp_filter, "highpass6")
+    PER_SMALL_MODULE_ITEM(onepole_ap_filter, "allpass")
+    PER_SMALL_MODULE_ITEM(min, "min")
+    PER_SMALL_MODULE_ITEM(max, "max")
+    PER_SMALL_MODULE_ITEM(minus, "minus")
+    PER_SMALL_MODULE_ITEM(mul, "mul")
+    PER_SMALL_MODULE_ITEM(neg, "neg")
+    PER_SMALL_MODULE_ITEM(min_c, "min_c")
+    PER_SMALL_MODULE_ITEM(max_c, "max_c")
+    PER_SMALL_MODULE_ITEM(minus_c, "minus_c")
+    PER_SMALL_MODULE_ITEM(mul_c, "mul_c")
+    PER_SMALL_MODULE_ITEM(neg_c, "neg_c")
+    PER_SMALL_MODULE_ITEM(level2edge_c, "level2edge_c")
+    PER_SMALL_MODULE_ITEM(map_lin2exp, "lin2exp")
+    PER_SMALL_MODULE_ITEM(square_osc, "square_osc")
+    PER_SMALL_MODULE_ITEM(saw_osc, "saw_osc")
+    PER_SMALL_MODULE_ITEM(square_lfo, "square_lfo")
+    PER_SMALL_MODULE_ITEM(saw_lfo, "saw_lfo")
+    PER_SMALL_MODULE_ITEM(pulse_lfo, "pulse_lfo")
+    PER_SMALL_MODULE_ITEM(print_a, "print_a")
+    PER_SMALL_MODULE_ITEM(print_c, "print_c")
+    PER_SMALL_MODULE_ITEM(print_e, "print_e")
+    PER_SMALL_MODULE_ITEM(print_em, "print_em")
+    PER_SMALL_MODULE_ITEM(copy_em, "copy_em")
+    PER_SMALL_MODULE_ITEM(notefilter_m, "notefilter_m")
+    PER_SMALL_MODULE_ITEM(ccfilter_m, "ccfilter_m")
+    PER_SMALL_MODULE_ITEM(pcfilter_m, "pcfilter_m")
+    PER_SMALL_MODULE_ITEM(pressurefilter_m, "pressurefilter_m")
+    PER_SMALL_MODULE_ITEM(pitchbendfilter_m, "pitchbendfilter_m")
+    PER_SMALL_MODULE_ITEM(systemfilter_m, "systemfilter_m")
+    PER_SMALL_MODULE_ITEM(channelfilter_m, "channelfilter_m")
+    PER_SMALL_MODULE_ITEM(keyfilter_m, "keyfilter_m")
+    PER_SMALL_MODULE_ITEM(setchannel_m, "setchannel_m")
+    PER_SMALL_MODULE_ITEM(key_less_than_m, "key_less_than_m")
+    PER_SMALL_MODULE_ITEM(channel_less_than_m, "channel_less_than_m")
+    PER_SMALL_MODULE_ITEM(transpose_m, "transpose_m")
+    PER_SMALL_MODULE_ITEM(eventmerge_e, "eventmerge_e")
+    PER_SMALL_MODULE_ITEM(quadpower_a, "quadpower_a")
+    PER_SMALL_MODULE_ITEM(quadpower_c, "quadpower_c")
+    PER_SMALL_MODULE_ITEM(crossfader2_a, "crossfader2_a")
+    PER_SMALL_MODULE_ITEM(crossfader2_c, "crossfader2_c")
+    PER_SMALL_MODULE_ITEM(linear_inertia_c, "linear_inertia_c")
+    PER_SMALL_MODULE_ITEM(exp_inertia_c, "exp_inertia_c")
+    PER_SMALL_MODULE_ITEM(sample_hold_edge_c, "sample_hold_edge_c")
+    PER_SMALL_MODULE_ITEM(sample_hold_level_c, "sample_hold_level_c")
+    PER_SMALL_MODULE_ITEM(bit_and_c, "bit_and_c")
+    PER_SMALL_MODULE_ITEM(bit_or_c, "bit_or_c")
+    PER_SMALL_MODULE_ITEM(bit_xor_c, "bit_xor_c")
+    PER_SMALL_MODULE_ITEM(logical_and_c, "logical_and_c")
+    PER_SMALL_MODULE_ITEM(logical_or_c, "logical_or_c")
+    PER_SMALL_MODULE_ITEM(logical_xor_c, "logical_xor_c")
+    PER_SMALL_MODULE_ITEM(logical_not_c, "logical_not_c")
+    PER_SMALL_MODULE_ITEM(flipflop_c, "flipflop_c")
+    PER_SMALL_MODULE_ITEM(schmitt_c, "schmitt_c")
+    PER_SMALL_MODULE_ITEM(between_c, "between_c")
+    PER_SMALL_MODULE_ITEM(less_c, "less_c")
+    PER_SMALL_MODULE_ITEM(clip_c, "clip_c")
+    PER_SMALL_MODULE_ITEM(trigger_a2c, "trigger_a2c")
+    PER_SMALL_MODULE_ITEM(timer_c, "timer_c")
+    PER_SMALL_MODULE_ITEM(prio_mux_c, "prio_mux_c")
+    PER_SMALL_MODULE_ITEM(prio_enc8_c, "prio_enc8_c")
+    PER_SMALL_MODULE_ITEM(ifthenelse_c, "ifthenelse_c")
+    PER_SMALL_MODULE_ITEM(counter_c, "counter_c")
+    PER_SMALL_MODULE_ITEM(mux4_c, "mux4_c")
+    PER_SMALL_MODULE_ITEM(mux8_c, "mux8_c")
+    PER_SMALL_MODULE_ITEM(mux16_c, "mux16_c")
+    PER_SMALL_MODULE_ITEM(msgread_e, "msgread_e")
+#endif
+#undef PER_SMALL_MODULE_ITEM
+#endif

plugins/ladspa_effect/calf/calf/modules_dev.h

@@ -0,0 +1,119 @@
+/* Calf DSP Library
+ * Prototype audio modules
+ *
+ * Copyright (C) 2008 Thor Harald Johansen <thj@thj.no>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+#ifndef __CALF_MODULES_DEV_H
+#define __CALF_MODULES_DEV_H
+
+#include <calf/metadata.h>
+#include <calf/modules.h>
+
+#if ENABLE_EXPERIMENTAL
+#include <fluidsynth.h>
+#endif
+
+namespace calf_plugins {
+
+#if ENABLE_EXPERIMENTAL
+    
+/// Tiny wrapper for fluidsynth
+class fluidsynth_audio_module: public audio_module<fluidsynth_metadata>
+{
+public:
+    float *ins[in_count]; 
+    float *outs[out_count];
+    float *params[param_count];
+
+protected:
+    /// Current sample rate
+    uint32_t srate;
+    /// FluidSynth Settings object
+    fluid_settings_t *settings;
+    /// FluidSynth Synth object
+    fluid_synth_t *synth;
+    /// Soundfont filename
+    std::string soundfont;
+    /// Soundfont filename (as received from Fluidsynth)
+    std::string soundfont_name;
+    /// TAB-separated preset list (preset+128*bank TAB preset name LF)
+    std::string soundfont_preset_list;
+    /// FluidSynth assigned SoundFont ID
+    int sfid;
+    /// Map of preset+128*bank to preset name
+    std::map<uint32_t, std::string> sf_preset_names;
+    /// Last selected preset+128*bank
+    uint32_t last_selected_preset;
+    /// Serial number of status data
+    int status_serial;
+    /// Preset number to set on next process() call
+    volatile int set_preset;
+
+    /// Update last_selected_preset based on synth object state
+    void update_preset_num();
+    /// Create a fluidsynth object and load the current soundfont
+    fluid_synth_t *create_synth(int &new_sfid);
+public:
+    /// Constructor to initialize handles to NULL
+    fluidsynth_audio_module();
+
+    void post_instantiate();
+    void set_sample_rate(uint32_t sr) { srate = sr; }
+    /// Handle MIDI Note On message (by sending it to fluidsynth)
+    void note_on(int note, int vel);
+    /// Handle MIDI Note Off message (by sending it to fluidsynth)
+    void note_off(int note, int vel);
+    /// Handle pitch bend message.
+    inline void pitch_bend(int value)
+    {
+        fluid_synth_pitch_bend(synth, 0, value + 0x2000);
+    }
+    /// Handle control change messages.
+    void control_change(int controller, int value);
+    /// Handle program change messages.
+    void program_change(int program);
+
+    /// Update variables from control ports.
+    void params_changed() {
+    }
+    void activate();
+    void deactivate();
+    /// No CV inputs for now
+    bool is_cv(int param_no) { return false; }
+    /// Practically all the stuff here is noisy... for now
+    bool is_noisy(int param_no) { return true; }
+    /// Main processing function
+    uint32_t process(uint32_t offset, uint32_t nsamples, uint32_t inputs_mask, uint32_t outputs_mask);
+    /// DSSI-style configure function for handling string port data
+    char *configure(const char *key, const char *value);
+    void send_configures(send_configure_iface *sci);
+    int send_status_updates(send_updates_iface *sui, int last_serial);
+    uint32_t message_run(const void *valid_inputs, void *output_ports) { 
+        // silence a default printf (which is kind of a warning about unhandled message_run)
+        return 0;
+    }
+    ~fluidsynth_audio_module();
+};
+
+
+    
+#endif
+    
+};
+
+#endif

plugins/ladspa_effect/calf/calf/modules_small.h

@@ -0,0 +1,202 @@
+/* Calf DSP Library
+ * "Small" audio modules for modular synthesis
+ *
+ * Copyright (C) 2001-2007 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+#ifndef __CALF_MODULES_SMALL_H
+#define __CALF_MODULES_SMALL_H
+
+#if USE_LV2
+
+#include <lv2.h>
+#include "plugininfo.h"
+#include "lv2_polymorphic_port.h"
+#include "lv2helpers.h"
+
+namespace calf_plugins {
+
+/// Empty implementations for plugin functions. Note, that some functions aren't virtual, because they're called via the particular
+/// subclass via template wrappers (ladspa_small_wrapper<> etc), not via base class pointer/reference. On the other hand,
+/// other functions are virtual when overhead is acceptable (instantiation time functions etc.)
+class null_small_audio_module: public uri_map_access
+{
+public:
+    uint32_t srate;
+    double odsr;
+    uint32_t poly_type_control, poly_type_audio;
+    /// for polymorphic ports: "is audio" flags for first 32 ports (should be sufficient for most plugins)
+    uint32_t poly_port_types;
+
+    null_small_audio_module()
+    : srate((uint32_t)-1)
+    , odsr(0.)
+    , poly_type_control(0)
+    , poly_type_audio(0)
+    , poly_port_types(0)
+    {
+    }
+
+    /// Called when host changes type of the polymorphic port
+    inline void set_port_type(uint32_t port, uint32_t type, void *type_data) {
+        if (port >= 32)
+            return;
+        uint32_t port_mask = 1 << port;
+        if (type == poly_type_control)
+            poly_port_types &= ~port_mask;
+        else if (type == poly_type_audio)
+            poly_port_types |= port_mask;
+        on_port_types_changed();
+    }
+    
+    /// Returns 1 for audio ports and 0 for control ports
+    inline unsigned int port_is_audio(unsigned int port) {
+        return (poly_port_types >> port) & 1;
+    }
+        
+    /// Returns (unsigned)-1 for audio ports and 0 for control ports
+    inline unsigned int port_audio_mask(unsigned int port) {
+        return 0 - ((poly_port_types >> port) & 1);
+    }
+        
+    /// Returns (unsigned)-1 for audio ports and 0 for control ports
+    static inline unsigned int port_audio_mask(unsigned int port, uint32_t poly_port_types) {
+        return 0 - ((poly_port_types >> port) & 1);
+    }
+        
+    virtual void on_port_types_changed() {}
+    inline void set_bundle_path(const char *path) {}
+    /// Called to map all the necessary URIs
+    virtual void map_uris()
+    {
+        poly_type_control = map_uri(LV2_POLYMORPHIC_PORT_URI, "http://lv2plug.in/ns/lv2core#ControlPort");
+        poly_type_audio = map_uri(LV2_POLYMORPHIC_PORT_URI, "http://lv2plug.in/ns/lv2core#AudioPort");
+    }
+    /// Called on instantiation with the list of LV2 features called
+    virtual void use_features(const LV2_Feature *const *features) {
+        while(*features)
+        {
+            use_feature((*features)->URI, (*features)->data);
+            features++;
+        }
+    }
+    /// LADSPA-esque activate function, except it is called after ports are connected, not before
+    inline void activate() {}
+    /// LADSPA-esque deactivate function
+    inline void deactivate() {}
+    /// Set sample rate for the plugin
+    inline void set_sample_rate(uint32_t sr) { srate = sr; odsr = 1.0 / sr; }
+    static inline const void *ext_data(const char *URI) { return NULL; }
+};
+
+/// Templatized version useful when the number of inputs and outputs is small
+template<unsigned int Inputs, unsigned int Outputs>
+class small_audio_module_base: public null_small_audio_module
+{
+public:
+    enum { in_count = Inputs, out_count = Outputs };
+    /// Input pointers
+    float *ins[in_count]; 
+    /// Output pointers
+    float *outs[out_count];
+};
+
+template<class Module>
+struct lv2_small_wrapper
+{
+    typedef Module instance;
+    static LV2_Descriptor descriptor;
+    std::string uri;
+    static uint32_t poly_port_types;
+    
+    lv2_small_wrapper(const char *id)
+    {
+        uri = "http://calf.sourceforge.net/small_plugins/" + std::string(id);
+        descriptor.URI = uri.c_str();
+        descriptor.instantiate = cb_instantiate;
+        descriptor.connect_port = cb_connect;
+        descriptor.activate = cb_activate;
+        descriptor.run = cb_run;
+        descriptor.deactivate = cb_deactivate;
+        descriptor.cleanup = cb_cleanup;
+        descriptor.extension_data = cb_ext_data;
+        
+        plugin_port_type_grabber ptg(poly_port_types);
+        Module::plugin_info(&ptg);
+    }
+
+    static void cb_connect(LV2_Handle Instance, uint32_t port, void *DataLocation) {
+        unsigned long ins = Module::in_count;
+        unsigned long outs = Module::out_count;
+        instance *const mod = (instance *)Instance;
+        if (port < ins)
+            mod->ins[port] = (float *)DataLocation;
+        else if (port < ins + outs)
+            mod->outs[port - ins] = (float *)DataLocation;
+    }
+
+    static void cb_activate(LV2_Handle Instance) {
+        // Note the changed semantics (now more LV2-like)
+        instance *const mod = (instance *)Instance;
+        mod->activate();
+    }
+    
+    static void cb_deactivate(LV2_Handle Instance) {
+        instance *const mod = (instance *)Instance;
+        mod->deactivate();
+    }
+    
+    static uint32_t cb_set_type(LV2_Handle Instance, uint32_t port, uint32_t type, void *type_data)
+    {
+        instance *const mod = (instance *)Instance;
+        mod->set_port_type(port, type, type_data);
+        return 0;
+    }
+
+    static LV2_Handle cb_instantiate(const LV2_Descriptor * Descriptor, double sample_rate, const char *bundle_path, const LV2_Feature *const *features)
+    {
+        instance *mod = new instance();
+        mod->poly_port_types = poly_port_types;
+        // XXXKF some people use fractional sample rates; we respect them ;-)
+        mod->set_bundle_path(bundle_path);
+        mod->use_features(features);
+        mod->set_sample_rate((uint32_t)sample_rate);
+        return mod;
+    }
+    
+    static void cb_run(LV2_Handle Instance, uint32_t SampleCount) {
+        instance *const mod = (instance *)Instance;
+        mod->process(SampleCount);
+    }
+    
+    static void cb_cleanup(LV2_Handle Instance) {
+        instance *const mod = (instance *)Instance;
+        delete mod;
+    }
+    
+    static const void *cb_ext_data(const char *URI) {
+        return Module::ext_data(URI);
+    }
+};
+
+extern const LV2_Descriptor *lv2_small_descriptor(uint32_t index);
+
+};
+
+#endif
+
+#endif

plugins/ladspa_effect/calf/calf/modules_synths.h

@@ -0,0 +1,264 @@
+/* Calf DSP Library
+ * Audio modules - synthesizers
+ *
+ * Copyright (C) 2001-2007 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+#ifndef __CALF_MODULES_SYNTHS_H
+#define __CALF_MODULES_SYNTHS_H
+
+#include <assert.h>
+#include "biquad.h"
+#include "onepole.h"
+#include "audio_fx.h"
+#include "inertia.h"
+#include "osc.h"
+#include "synth.h"
+#include "envelope.h"
+#include "organ.h"
+#include "modmatrix.h"
+
+namespace calf_plugins {
+
+#define MONOSYNTH_WAVE_BITS 12
+    
+/// Monosynth-in-making. Parameters may change at any point, so don't make songs with it!
+/// It lacks inertia for parameters, even for those that really need it.
+class monosynth_audio_module: public audio_module<monosynth_metadata>, public line_graph_iface, public mod_matrix
+{
+public:
+    enum { mod_matrix_slots = 10 };
+    float *ins[in_count]; 
+    float *outs[out_count];
+    float *params[param_count];
+    uint32_t srate, crate;
+    static dsp::waveform_family<MONOSYNTH_WAVE_BITS> *waves;
+    dsp::waveform_oscillator<MONOSYNTH_WAVE_BITS> osc1, osc2;
+    dsp::triangle_lfo lfo;
+    bool running, stopping, gate, force_fadeout;
+    int last_key;
+    
+    float buffer[step_size], buffer2[step_size];
+    uint32_t output_pos;
+    dsp::onepole<float> phaseshifter;
+    dsp::biquad_d1_lerp<float> filter;
+    dsp::biquad_d1_lerp<float> filter2;
+    /// Waveform number - OSC1
+    int wave1;
+    /// Waveform number - OSC2
+    int wave2;
+    /// Last used waveform number - OSC1
+    int prev_wave1;
+    /// Last used waveform number - OSC2
+    int prev_wave2;
+    int filter_type, last_filter_type;
+    float freq, start_freq, target_freq, cutoff, decay_factor, fgain, fgain_delta, separation;
+    float detune, xpose, xfade, ampctl, fltctl, queue_vel;
+    float odcr, porta_time, lfo_bend, lfo_clock, last_lfov, modwheel_value;
+    /// Last value of phase shift for pulse width emulation for OSC1
+    int32_t last_pwshift1;
+    /// Last value of phase shift for pulse width emulation for OSC2
+    int32_t last_pwshift2;
+    int queue_note_on, stop_count, modwheel_value_int;
+    int legato;
+    dsp::adsr envelope;
+    dsp::keystack stack;
+    dsp::gain_smoothing master;
+    /// Smoothed cutoff value
+    dsp::inertia<dsp::exponential_ramp> inertia_cutoff;
+    /// Smoothed pitch bend value
+    dsp::inertia<dsp::exponential_ramp> inertia_pitchbend;
+    /// Smoothed channel pressure value
+    dsp::inertia<dsp::linear_ramp> inertia_pressure;
+    /// Rows of the modulation matrix
+    dsp::modulation_entry mod_matrix_data[mod_matrix_slots];
+    /// Currently used velocity
+    float velocity;
+    /// Last value of oscillator mix ratio
+    float last_xfade;
+    /// Current calculated mod matrix outputs
+    float moddest[moddest_count];
+     
+    monosynth_audio_module();    
+    static void precalculate_waves(progress_report_iface *reporter);
+    void set_sample_rate(uint32_t sr);
+    void delayed_note_on();
+    /// Handle MIDI Note On message (does not immediately trigger a note, as it must start on
+    /// boundary of step_size samples).
+    void note_on(int note, int vel);
+    /// Handle MIDI Note Off message
+    void note_off(int note, int vel);
+    /// Handle MIDI Channel Pressure
+    void channel_pressure(int value);
+    /// Handle pitch bend message.
+    inline void pitch_bend(int value)
+    {
+        inertia_pitchbend.set_inertia(pow(2.0, (value * *params[par_pwhlrange]) / (1200.0 * 8192.0)));
+    }
+    /// Update oscillator frequency based on base frequency, detune amount, pitch bend scaling factor and sample rate.
+    inline void set_frequency()
+    {
+        float detune_scaled = (detune - 1); // * log(freq / 440);
+        if (*params[par_scaledetune] > 0)
+            detune_scaled *= pow(20.0 / freq, *params[par_scaledetune]);
+        float p1 = 1, p2 = 1;
+        if (moddest[moddest_o1detune] != 0)
+            p1 = pow(2.0, moddest[moddest_o1detune] * (1.0 / 1200.0));
+        if (moddest[moddest_o2detune] != 0)
+            p2 = pow(2.0, moddest[moddest_o2detune] * (1.0 / 1200.0));
+        osc1.set_freq(freq * (1 - detune_scaled) * p1 * inertia_pitchbend.get_last() * lfo_bend, srate);
+        osc2.set_freq(freq * (1 + detune_scaled) * p2 * inertia_pitchbend.get_last() * lfo_bend * xpose, srate);
+    }
+    /// Handle control change messages.
+    void control_change(int controller, int value);
+    /// Update variables from control ports.
+    void params_changed() {
+        float sf = 0.001f;
+        envelope.set(*params[par_attack] * sf, *params[par_decay] * sf, std::min(0.999f, *params[par_sustain]), *params[par_release] * sf, srate / step_size, *params[par_fade] * sf);
+        filter_type = dsp::fastf2i_drm(*params[par_filtertype]);
+        decay_factor = odcr * 1000.0 / *params[par_decay];
+        separation = pow(2.0, *params[par_cutoffsep] / 1200.0);
+        wave1 = dsp::clip(dsp::fastf2i_drm(*params[par_wave1]), 0, (int)wave_count - 1);
+        wave2 = dsp::clip(dsp::fastf2i_drm(*params[par_wave2]), 0, (int)wave_count - 1);
+        detune = pow(2.0, *params[par_detune] / 1200.0);
+        xpose = pow(2.0, *params[par_osc2xpose] / 12.0);
+        xfade = *params[par_oscmix];
+        legato = dsp::fastf2i_drm(*params[par_legato]);
+        master.set_inertia(*params[par_master]);
+        set_frequency();
+        if (wave1 != prev_wave1 || wave2 != prev_wave2)
+            lookup_waveforms();
+    }
+    void activate();
+    void deactivate();
+    void post_instantiate()
+    {
+        precalculate_waves(progress_report);
+    }
+    /// Set waveform addresses for oscillators
+    void lookup_waveforms();
+    /// Run oscillators
+    void calculate_buffer_oscs(float lfo);
+    /// Run two filters in series to produce mono output samples.
+    void calculate_buffer_ser();
+    /// Run one filter to produce mono output samples.
+    void calculate_buffer_single();
+    /// Run two filters (one per channel) to produce stereo output samples.
+    void calculate_buffer_stereo();
+    /// Retrieve filter graph (which is 'live' so it cannot be generated by get_static_graph), or fall back to get_static_graph.
+    bool get_graph(int index, int subindex, float *data, int points, cairo_iface *context);
+    /// @retval true if the filter 1 is to be used for the left channel and filter 2 for the right channel
+    /// @retval false if filters are to be connected in series and sent (mono) to both channels    
+    inline bool is_stereo_filter() const
+    {
+        return filter_type == flt_2lp12 || filter_type == flt_2bp6;
+    }
+    /// No CV inputs for now
+    bool is_cv(int param_no) { return false; }
+    /// Practically all the stuff here is noisy
+    bool is_noisy(int param_no) { return param_no != par_cutoff; }
+    /// Calculate control signals and produce step_size samples of output.
+    void calculate_step();
+    /// Main processing function
+    uint32_t process(uint32_t offset, uint32_t nsamples, uint32_t inputs_mask, uint32_t outputs_mask);
+};
+
+struct organ_audio_module: public audio_module<organ_metadata>, public dsp::drawbar_organ, public line_graph_iface
+{
+public:
+    using drawbar_organ::note_on;
+    using drawbar_organ::note_off;
+    using drawbar_organ::control_change;
+    enum { param_count = drawbar_organ::param_count};
+    float *ins[in_count]; 
+    float *outs[out_count];
+    float *params[param_count];
+    dsp::organ_parameters par_values;
+    uint32_t srate;
+    bool panic_flag;
+    /// Value for configure variable map_curve
+    std::string var_map_curve;
+
+    organ_audio_module()
+    : drawbar_organ(&par_values)
+    {
+        var_map_curve = "2\n0 1\n1 1\n"; // XXXKF hacky bugfix
+    }
+    
+    void post_instantiate()
+    {
+        dsp::organ_voice_base::precalculate_waves(progress_report);
+    }
+
+    void set_sample_rate(uint32_t sr) {
+        srate = sr;
+    }
+    void params_changed() {
+        for (int i = 0; i < param_count - var_count; i++)
+            ((float *)&par_values)[i] = *params[i];
+
+        unsigned int old_poly = polyphony_limit;
+        polyphony_limit = dsp::clip(dsp::fastf2i_drm(*params[par_polyphony]), 1, 32);
+        if (polyphony_limit < old_poly)
+            trim_voices();
+        
+        update_params();
+    }
+    inline void pitch_bend(int amt)
+    {
+        drawbar_organ::pitch_bend(amt);
+    }
+    void activate() {
+        setup(srate);
+        panic_flag = false;
+    }
+    void deactivate();
+    uint32_t process(uint32_t offset, uint32_t nsamples, uint32_t inputs_mask, uint32_t outputs_mask) {
+        float *o[2] = { outs[0] + offset, outs[1] + offset };
+        if (panic_flag)
+        {
+            control_change(120, 0); // stop all sounds
+            control_change(121, 0); // reset all controllers
+            panic_flag = false;
+        }
+        render_separate(o, nsamples);
+        return 3;
+    }
+    /// No CV inputs for now
+    bool is_cv(int param_no) { return false; }
+    /// Practically all the stuff here is noisy
+    bool is_noisy(int param_no) { return true; }
+    void execute(int cmd_no);
+    bool get_graph(int index, int subindex, float *data, int points, cairo_iface *context);
+    
+    char *configure(const char *key, const char *value);
+    void send_configures(send_configure_iface *);
+    uint32_t message_run(const void *valid_inputs, void *output_ports) { 
+        // silence a default printf (which is kind of a warning about unhandled message_run)
+        return 0;
+    }
+};
+
+};
+
+#if ENABLE_EXPERIMENTAL
+    
+#include "wavetable.h"
+
+#endif
+
+#endif

plugins/ladspa_effect/calf/calf/multichorus.h

@@ -0,0 +1,213 @@
+/* Calf DSP Library
+ * Multitap chorus class.
+ *
+ * Copyright (C) 2001-2007 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+#ifndef __CALF_MULTICHORUS_H
+#define __CALF_MULTICHORUS_H
+
+#include "audio_fx.h"
+
+namespace dsp {
+
+typedef fixed_point<unsigned int, 20> chorus_phase;
+
+template<class T, uint32_t Voices>
+class sine_multi_lfo
+{
+protected:
+    sine_table<int, 4096, 65535> sine;
+    
+public:
+    /// Current LFO phase
+    chorus_phase phase;
+    /// LFO phase increment
+    chorus_phase dphase;
+    /// LFO phase per-voice increment
+    chorus_phase vphase;
+    /// Current number of voices
+    uint32_t voices;
+    /// Current scale (output multiplier)
+    T scale;
+    /// Per-voice offset unit (the value that says how much the voices are offset with respect to each other in non-100% 'overlap' mode), scaled so that full range = 131072
+    int32_t voice_offset;
+    /// LFO Range scaling for non-100% overlap
+    uint32_t voice_depth;
+public:
+    sine_multi_lfo()
+    {
+        phase = dphase = vphase = 0.0;
+        voice_offset = 0;
+        voice_depth = 1U << 31;
+        
+        set_voices(Voices);
+    }
+    inline uint32_t get_voices() const
+    {
+        return voices;
+    }
+    inline void set_voices(uint32_t value)
+    {
+        voices = value;
+        // use sqrt, because some phases will cancel each other - so 1 / N is usually too low
+        scale = sqrt(1.0 / voices);
+    }
+    inline void set_overlap(float overlap)
+    {
+        // If we scale the delay amount so that full range of a single LFO is 0..1, all the overlapped LFOs will cover 0..range
+        // How it's calculated:
+        // 1. First voice is assumed to always cover the range of 0..1
+        // 2. Each remaining voice contributes an interval of a width = 1 - overlap, starting from the end of the interval of the previous voice
+        // Coverage = non-overlapped part of the LFO range in the 1st voice
+        float range = 1.f + (1.f - overlap) * (voices - 1);
+        float scaling = 1.f / range;
+        voice_offset = (int)(131072 * (1 - overlap) / range);
+        voice_depth = (unsigned int)((1U << 30) * 1.0 * scaling);
+    }
+    /// Get LFO value for given voice, returns a values in range of [-65536, 65535] (or close)
+    inline int get_value(uint32_t voice) {
+        // find this voice's phase (= phase + voice * 360 degrees / number of voices)
+        chorus_phase voice_phase = phase + vphase * (int)voice;
+        // find table offset
+        unsigned int ipart = voice_phase.ipart();
+        // interpolate (use 14 bits of precision - because the table itself uses 17 bits and the result of multiplication must fit in int32_t)
+        // note, the result is still -65535 .. 65535, it's just interpolated
+        // it is never reaching -65536 - but that's acceptable
+        int intval = voice_phase.lerp_by_fract_int<int, 14, int>(sine.data[ipart], sine.data[ipart+1]);
+        // apply the voice offset/depth (rescale from -65535..65535 to appropriate voice's "band")
+        return -65535 + voice * voice_offset + ((voice_depth >> (30-13)) * (65536 + intval) >> 13);
+    }
+    inline void step() {
+        phase += dphase;
+    }
+    inline T get_scale() const {
+        return scale;
+    }
+    void reset() {
+        phase = 0.f;
+    }
+};
+
+/**
+ * Multi-tap chorus without feedback.
+ * Perhaps MaxDelay should be a bit longer!
+ */
+template<class T, class MultiLfo, class Postprocessor, int MaxDelay=4096>
+class multichorus: public chorus_base
+{
+protected:
+    simple_delay<MaxDelay,T> delay;
+public:    
+    MultiLfo lfo;
+    Postprocessor post;
+public:
+    multichorus() {
+        rate = 0.63f;
+        dry = 0.5f;
+        wet = 0.5f;
+        min_delay = 0.005f;
+        mod_depth = 0.0025f;
+        setup(44100);
+    }
+    void reset() {
+        delay.reset();
+        lfo.reset();
+    }
+    void set_rate(float rate) {
+        chorus_base::set_rate(rate);
+        lfo.dphase = dphase;
+    }
+    virtual void setup(int sample_rate) {
+        modulation_effect::setup(sample_rate);
+        delay.reset();
+        lfo.reset();
+        set_min_delay(get_min_delay());
+        set_mod_depth(get_mod_depth());
+    }
+    template<class OutIter, class InIter>
+    void process(OutIter buf_out, InIter buf_in, int nsamples) {
+        int mds = min_delay_samples + mod_depth_samples * 1024 + 2*65536;
+        int mdepth = mod_depth_samples;
+        // 1 sample peak-to-peak = mod_depth_samples of 32 (this scaling stuff is tricky and may - but shouldn't - be wrong)
+        // with 192 kHz sample rate, 1 ms = 192 samples, and the maximum 20 ms = 3840 samples (so, 4096 will be used)
+        // 3840 samples of mod depth = mdepth of 122880 (which multiplied by 65536 doesn't fit in int32_t)
+        // so, it will be right-shifted by 2, which gives it a safe range of 30720
+        // NB: calculation of mod_depth_samples (and multiply-by-32) is in chorus_base::set_mod_depth
+        mdepth = mdepth >> 2;
+        T scale = lfo.get_scale();
+        for (int i=0; i<nsamples; i++) {
+            phase += dphase;
+            
+            float in = *buf_in++;
+            
+            delay.put(in);
+            unsigned int nvoices = lfo.get_voices();
+            T out = 0.f;
+            // add up values from all voices, each voice tell its LFO phase and the buffer value is picked at that location
+            for (unsigned int v = 0; v < nvoices; v++)
+            {
+                int lfo_output = lfo.get_value(v);
+                // 3 = log2(32 >> 2) + 1 because the LFO value is in range of [-65535, 65535] (17 bits)
+                int dv = mds + (mdepth * lfo_output >> (3 + 1));
+                int ifv = dv >> 16;
+                T fd; // signal from delay's output
+                delay.get_interp(fd, ifv, (dv & 0xFFFF)*(1.0/65536.0));
+                out += fd;
+            }
+            // apply the post filter
+            out = post.process(out);
+            T sdry = in * gs_dry.get();
+            T swet = out * gs_wet.get() * scale;
+            *buf_out++ = sdry + swet;
+            lfo.step();
+        }
+        post.sanitize();
+    }
+    float freq_gain(float freq, float sr)
+    {
+        typedef std::complex<double> cfloat;
+        freq *= 2.0 * M_PI / sr;
+        cfloat z = 1.0 / exp(cfloat(0.0, freq)); // z^-1        
+        cfloat h = 0.0;
+        int mds = min_delay_samples + mod_depth_samples * 1024 + 2*65536;
+        int mdepth = mod_depth_samples;
+        mdepth = mdepth >> 2;
+        T scale = lfo.get_scale();
+        unsigned int nvoices = lfo.get_voices();
+        for (unsigned int v = 0; v < nvoices; v++)
+        {
+            int lfo_output = lfo.get_value(v);
+            // 3 = log2(32 >> 2) + 1 because the LFO value is in range of [-65535, 65535] (17 bits)
+            int dv = mds + (mdepth * lfo_output >> (3 + 1));
+            int fldp = dv >> 16;
+            cfloat zn = std::pow(z, fldp); // z^-N
+            h += zn + (zn * z - zn) * cfloat(dv / 65536.0 - fldp);
+        }
+        // apply the post filter
+        h *= post.h_z(z);
+        // mix with dry signal
+        float v = std::abs(cfloat(gs_dry.get_last()) + cfloat(scale * gs_wet.get_last()) * h);
+        return v;
+    }
+};
+
+};
+
+#endif
+
+

plugins/ladspa_effect/calf/calf/onepole.h

@@ -0,0 +1,192 @@
+/* Calf DSP Library
+ * Basic one-pole one-zero filters based on bilinear transform.
+ * Copyright (C) 2001-2007 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+#ifndef __CALF_ONEPOLE_H
+#define __CALF_ONEPOLE_H
+
+#include "primitives.h"
+
+namespace dsp {
+
+/**
+ * one-pole filter, for floating point values
+ * coefficient calculation is based on bilinear transform, and the code itself is based on my very old OneSignal lib
+ * lp and hp are *somewhat* tested, allpass is not tested at all
+ * don't use this for integers because it won't work
+ */
+template<class T = float, class Coeff = float>
+class onepole
+{
+public:
+    typedef std::complex<double> cfloat;
+
+    T x1, y1;
+    Coeff a0, a1, b1;
+
+    onepole()
+    {
+        reset();
+    }
+    
+    /// Set coefficients for a lowpass filter
+    void set_lp(float fc, float sr)
+    {
+        //   x   x
+        //  x+1 x-1
+        Coeff x = tan (M_PI * fc / (2 * sr));
+        Coeff q = 1/(1+x);
+	a0 = a1 = x*q;
+	b1 = (x-1)*q;
+    }
+    
+    /// Set coefficients for an allpass filter
+    void set_ap(float fc, float sr)
+    {
+        // x-1  x+1
+        // x+1  x-1
+	Coeff x = tan (M_PI * fc / (2 * sr));
+	Coeff q = 1/(1+x);
+	b1 = a0 = (x-1)*q;
+        a1 = 1;
+    }
+    
+    /// Set coefficients for an allpass filter, using omega instead of fc and sr
+    /// omega = (PI / 2) * fc / sr
+    void set_ap_w(float w)
+    {
+        // x-1  x+1
+        // x+1  x-1
+	Coeff x = tan (w);
+	Coeff q = 1/(1+x);
+	b1 = a0 = (x-1)*q;
+        a1 = 1;
+    }
+    
+    /// Set coefficients for a highpass filter
+    void set_hp(float fc, float sr)
+    {
+        //   x   -x
+        //  x+1  x-1
+	Coeff x = tan (M_PI * fc / (2 * sr));
+	Coeff q = 1/(1+x);
+	a0 = q;
+        a1 = -a0;
+	b1 = (x-1)*q;
+    }
+    
+    /// Process one sample
+    inline T process(T in)
+    {
+        T out = in * a0 + x1 * a1 - y1 * b1;
+        x1 = in;
+        y1 = out;
+        return out;
+    }
+    
+    /// Process one sample, assuming it's a lowpass filter (optimized special case)
+    inline T process_lp(T in)
+    {
+        T out = (in + x1) * a0 - y1 * b1;
+        x1 = in;
+        y1 = out;
+        return out;
+    }
+
+    /// Process one sample, assuming it's a highpass filter (optimized special case)
+    inline T process_hp(T in)
+    {
+        T out = (in - x1) * a0 - y1 * b1;
+        x1 = in;
+        y1 = out;
+        return out;
+    }
+    
+    /// Process one sample, assuming it's an allpass filter (optimized special case)
+    inline T process_ap(T in)
+    {
+        T out = (in - y1) * a0 + x1;
+        x1 = in;
+        y1 = out;
+        return out;
+    }
+    
+    /// Process one sample using external state variables
+    inline T process_ap(T in, float &x1, float &y1)
+    {
+        T out = (in - y1) * a0 + x1;
+        x1 = in;
+        y1 = out;
+        return out;
+    }
+    
+    /// Process one sample using external state variables, including filter coeff
+    inline T process_ap(T in, float &x1, float &y1, float a0)
+    {
+        T out = (in - y1) * a0 + x1;
+        x1 = in;
+        y1 = out;
+        return out;
+    }
+    
+    inline bool empty() {
+        return y1 == 0;
+    }
+    
+    inline void sanitize() 
+    {
+        dsp::sanitize(x1);
+        dsp::sanitize(y1);
+    }
+    
+    inline void reset()
+    {
+        dsp::zero(x1);
+        dsp::zero(y1);
+    }
+    
+    template<class U>
+    inline void copy_coeffs(const onepole<U> &src)
+    {
+        a0 = src.a0;
+        a1 = src.a1;
+        b1 = src.b1;
+    }
+    
+    /// Return the filter's gain at frequency freq
+    /// @param freq   Frequency to look up
+    /// @param sr     Filter sample rate (used to convert frequency to angular frequency)
+    float freq_gain(float freq, float sr)
+    {
+        freq *= 2.0 * M_PI / sr;
+        cfloat z = 1.0 / exp(cfloat(0.0, freq));
+        
+        return std::abs(h_z(z));
+    }
+    
+    /// Return H(z) the filter's gain at frequency freq
+    /// @param z   Z variable (e^jw)
+    cfloat h_z(const cfloat &z)
+    {
+        return (cfloat(a0) + double(a1) * z) / (cfloat(1.0) + double(b1) * z);
+    }
+};
+
+};
+
+#endif

plugins/ladspa_effect/calf/calf/organ.h

@@ -0,0 +1,374 @@
+/* Calf DSP Library
+ * Drawbar organ emulator. 
+ *
+ * Copyright (C) 2007 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __CALF_ORGAN_H
+#define __CALF_ORGAN_H
+
+#include "synth.h"
+#include "envelope.h"
+#include "metadata.h"
+
+#define ORGAN_KEYTRACK_POINTS 4
+
+namespace dsp
+{
+
+struct organ_parameters {
+    enum { FilterCount = 2, EnvCount = 3 };
+    struct organ_filter_parameters
+    {
+        float cutoff;
+        float resonance;
+        float envmod[organ_parameters::EnvCount];
+        float keyf;
+    };
+
+    struct organ_env_parameters
+    {
+        float attack, decay, sustain, release, velscale, ampctl;
+    };
+        
+    //////////////////////////////////////////////////////////////////////////
+    // these parameters are binary-copied from control ports (order is important!)
+    
+    float drawbars[9];
+    float harmonics[9];
+    float waveforms[9];
+    float detune[9];
+    float phase[9];
+    float pan[9];
+    float routing[9];
+    float foldover;
+    float percussion_time;
+    float percussion_level;
+    float percussion_wave;
+    float percussion_harmonic;
+    float percussion_vel2amp;
+    float percussion_fm_time;
+    float percussion_fm_depth;
+    float percussion_fm_wave;
+    float percussion_fm_harmonic;
+    float percussion_vel2fm;
+    float percussion_trigger;
+    float percussion_stereo;
+    float filter_chain;
+    float filter1_type;
+    float master;
+
+    organ_filter_parameters filters[organ_parameters::FilterCount];
+    organ_env_parameters envs[organ_parameters::EnvCount];
+    float lfo_rate;
+    float lfo_amt;
+    float lfo_wet;
+    float lfo_phase;
+    float lfo_mode;
+    
+    float global_transpose;
+    float global_detune;
+    
+    float polyphony;
+    
+    float quad_env;
+    
+    float pitch_bend_range;
+    
+    float bass_freq;
+    float bass_gain;
+    float treble_freq;
+    float treble_gain;
+    
+    float dummy_mapcurve;
+    
+    //////////////////////////////////////////////////////////////////////////
+    // these parameters are calculated
+    
+    double perc_decay_const, perc_fm_decay_const;
+    float multiplier[9];
+    int phaseshift[9];
+    float cutoff;
+    unsigned int foldvalue;
+    float pitch_bend;
+
+    float percussion_keytrack[ORGAN_KEYTRACK_POINTS][2];
+    
+    organ_parameters() : pitch_bend(1.0f) {}
+
+    inline int get_percussion_wave() { return dsp::fastf2i_drm(percussion_wave); }
+    inline int get_percussion_fm_wave() { return dsp::fastf2i_drm(percussion_fm_wave); }
+};
+
+#define ORGAN_WAVE_BITS 12
+#define ORGAN_WAVE_SIZE 4096
+#define ORGAN_BIG_WAVE_BITS 17
+#define ORGAN_BIG_WAVE_SIZE 131072
+/// 2^ORGAN_BIG_WAVE_SHIFT = how many (quasi)periods per sample
+#define ORGAN_BIG_WAVE_SHIFT 5
+
+class organ_voice_base: public calf_plugins::organ_enums
+{
+public:
+    typedef waveform_family<ORGAN_WAVE_BITS> small_wave_family;
+    typedef waveform_family<ORGAN_BIG_WAVE_BITS> big_wave_family;
+public:
+    organ_parameters *parameters;
+protected:
+    static small_wave_family (*waves)[wave_count_small];
+    static big_wave_family (*big_waves)[wave_count_big];
+
+    // dsp::sine_table<float, ORGAN_WAVE_SIZE, 1> sine_wave;
+    int note;
+    dsp::decay amp;
+    /// percussion FM carrier amplitude envelope
+    dsp::decay pamp;
+    /// percussion FM modulator amplitude envelope
+    dsp::decay fm_amp;
+    dsp::fixed_point<int64_t, 20> pphase, dpphase;
+    dsp::fixed_point<int64_t, 20> modphase, moddphase;
+    float fm_keytrack;
+    int &sample_rate_ref;
+    bool &released_ref;
+    /// pamp per-sample (linear) step during release stage (calculated on release so that it will take 30ms for it to go from "current value at release point" to 0)
+    float rel_age_const;
+
+    organ_voice_base(organ_parameters *_parameters, int &_sample_rate_ref, bool &_released_ref);
+    
+    inline float wave(float *data, dsp::fixed_point<int, 20> ph) {
+        return ph.lerp_table_lookup_float(data);
+    }
+    inline float big_wave(float *data, dsp::fixed_point<int64_t, 20> &ph) {
+        // wrap to fit within the wave
+        return ph.lerp_table_lookup_float_mask(data, ORGAN_BIG_WAVE_SIZE - 1);
+    }
+public:
+    static inline small_wave_family &get_wave(int wave) {
+        return (*waves)[wave];
+    }
+    static inline big_wave_family &get_big_wave(int wave) {
+        return (*big_waves)[wave];
+    }
+    static void precalculate_waves(calf_plugins::progress_report_iface *reporter);
+    void update_pitch()
+    {
+        float phase = dsp::midi_note_to_phase(note, 100 * parameters->global_transpose + parameters->global_detune, sample_rate_ref);
+        dpphase.set((long int) (phase * parameters->percussion_harmonic * parameters->pitch_bend));
+        moddphase.set((long int) (phase * parameters->percussion_fm_harmonic * parameters->pitch_bend));
+    }
+    // this doesn't really have a voice interface
+    void render_percussion_to(float (*buf)[2], int nsamples);
+    void perc_note_on(int note, int vel);
+    void perc_note_off(int note, int vel);
+    void perc_reset()
+    {
+        pphase = 0;
+        modphase = 0;
+        dpphase = 0;
+        moddphase = 0;
+        note = -1;
+    }
+};
+
+class organ_vibrato
+{
+protected:
+    enum { VibratoSize = 6 };
+    float vibrato_x1[VibratoSize][2], vibrato_y1[VibratoSize][2];
+    float lfo_phase;
+    dsp::onepole<float> vibrato[2];
+public:
+    void reset();
+    void process(organ_parameters *parameters, float (*data)[2], unsigned int len, float sample_rate);
+};
+
+class organ_voice: public dsp::voice, public organ_voice_base {
+protected:    
+    enum { Channels = 2, BlockSize = 64, EnvCount = organ_parameters::EnvCount, FilterCount = organ_parameters::FilterCount };
+    union {
+        float output_buffer[BlockSize][Channels];
+        float aux_buffers[3][BlockSize][Channels];
+    };
+    dsp::fixed_point<int64_t, 52> phase, dphase;
+    dsp::biquad_d1<float> filterL[2], filterR[2];
+    adsr envs[EnvCount];
+    dsp::inertia<dsp::linear_ramp> expression;
+    organ_vibrato vibrato;
+    float velocity;
+    bool perc_released;
+    /// The envelopes have ended and the voice is in final fadeout stage
+    bool finishing;
+    dsp::inertia<dsp::exponential_ramp> inertia_pitchbend;
+
+public:
+    organ_voice()
+    : organ_voice_base(NULL, sample_rate, perc_released)
+    , expression(dsp::linear_ramp(16))
+    , inertia_pitchbend(dsp::exponential_ramp(1))
+    {
+        inertia_pitchbend.set_now(1);
+    }
+
+    void reset() {
+        inertia_pitchbend.ramp.set_length(sample_rate / (BlockSize * 30)); // 1/30s    
+        vibrato.reset();
+        phase = 0;
+        for (int i = 0; i < FilterCount; i++)
+        {
+            filterL[i].reset();
+            filterR[i].reset();
+        }
+    }
+
+    void note_on(int note, int vel) {
+        stolen = false;
+        finishing = false;
+        perc_released = false;
+        released = false;
+        reset();
+        this->note = note;
+        const float sf = 0.001f;
+        for (int i = 0; i < EnvCount; i++)
+        {
+            organ_parameters::organ_env_parameters &p = parameters->envs[i];
+            envs[i].set(sf * p.attack, sf * p.decay, p.sustain, sf * p.release, sample_rate / BlockSize);
+            envs[i].note_on();
+        }
+        update_pitch();
+        velocity = vel * 1.0 / 127.0;
+        amp.set(1.0f);
+        perc_note_on(note, vel);
+    }
+
+    void note_off(int /* vel */) {
+        // reset age to 0 (because decay will turn from exponential to linear, necessary because of error cumulation prevention)
+        perc_released = true;
+        if (pamp.get_active())
+        {
+            pamp.reinit();
+        }
+        rel_age_const = pamp.get() * ((1.0/44100.0)/0.03);
+        for (int i = 0; i < EnvCount; i++)
+            envs[i].note_off();
+    }
+
+    virtual float get_priority() { return stolen ? 20000 : (perc_released ? 1 : (sostenuto ? 200 : 100)); }
+    
+    virtual void steal() {
+        perc_released = true;
+        finishing = true;
+        stolen = true;
+    }
+
+    void render_block();
+    
+    virtual int get_current_note() {
+        return note;
+    }
+    virtual bool get_active() {
+        // printf("note %d getactive %d use_percussion %d pamp active %d\n", note, amp.get_active(), use_percussion(), pamp.get_active());
+        return (note != -1) && (amp.get_active() || (use_percussion() && pamp.get_active()));
+    }
+    void update_pitch();
+    inline bool use_percussion()
+    {
+        return dsp::fastf2i_drm(parameters->percussion_trigger) == perctrig_polyphonic && parameters->percussion_level > 0;
+    }
+};
+
+/// Not a true voice, just something with similar-ish interface.
+class percussion_voice: public organ_voice_base {
+public:
+    int sample_rate;
+    bool released;
+
+    percussion_voice(organ_parameters *_parameters)
+    : organ_voice_base(_parameters, sample_rate, released)
+    , released(false)
+    {
+    }
+    
+    bool get_active() {
+        return (note != -1) && pamp.get_active();
+    }
+    bool get_noticable() {
+        return (note != -1) && (pamp.get() > 0.2 * parameters->percussion_level);
+    }
+    void setup(int sr) {
+        sample_rate = sr;
+    }
+};
+
+struct drawbar_organ: public dsp::basic_synth, public calf_plugins::organ_enums {
+    organ_parameters *parameters;
+    percussion_voice percussion;
+    organ_vibrato global_vibrato;
+    two_band_eq eq_l, eq_r;
+    
+     drawbar_organ(organ_parameters *_parameters)
+    : parameters(_parameters)
+    , percussion(_parameters) {
+    }
+    void render_separate(float *output[], int nsamples);
+    dsp::voice *alloc_voice() {
+        block_voice<organ_voice> *v = new block_voice<organ_voice>();
+        v->parameters = parameters;
+        return v;
+    }
+    virtual void percussion_note_on(int note, int vel) {
+        percussion.perc_note_on(note, vel);
+    }
+    virtual void params_changed() = 0;
+    virtual void setup(int sr) {
+        basic_synth::setup(sr);
+        percussion.setup(sr);
+        parameters->cutoff = 0;
+        params_changed();
+        global_vibrato.reset();
+    }
+    void update_params();
+    void control_change(int controller, int value)
+    {
+#if 0
+        if (controller == 11)
+        {
+            parameters->cutoff = value / 64.0 - 1;
+        }
+#endif
+        dsp::basic_synth::control_change(controller, value);
+    }
+    void pitch_bend(int amt);
+    virtual bool check_percussion() { 
+        switch(dsp::fastf2i_drm(parameters->percussion_trigger))
+        {        
+            case organ_voice_base::perctrig_first:
+                return active_voices.empty();
+            case organ_voice_base::perctrig_each: 
+            default:
+                return true;
+            case organ_voice_base::perctrig_eachplus:
+                return !percussion.get_noticable();
+            case organ_voice_base::perctrig_polyphonic:
+                return false;
+        }
+    }
+};
+
+};
+
+#endif

plugins/ladspa_effect/calf/calf/osc.h

@@ -0,0 +1,306 @@
+/* Calf DSP Library
+ * Oscillators
+ *
+ * Copyright (C) 2001-2007 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __CALF_OSC_H
+#define __CALF_OSC_H
+
+#include "fft.h"
+#include <map>
+
+namespace dsp
+{
+
+/** Very simple, non-bandlimited saw oscillator. Should not be used for anything
+ *  else than testing/prototyping. Unless get() function is replaced with something
+ * with "proper" oscillator code, as the frequency setting function is fine.
+ */
+struct simple_oscillator
+{
+    /// Phase (from 0 to 0xFFFFFFFF)
+    uint32_t phase;
+    /// Per-sample phase delta (phase increment), equal to 2^32*freq/sr.
+    uint32_t phasedelta;
+    /// Reset oscillator phase to zero.
+    void reset()
+    {
+        phase = 0;
+    }
+    /// Set phase delta based on oscillator frequency and sample rate.
+    void set_freq(float freq, float sr)
+    {
+        phasedelta = (int)(freq * 65536.0 * 256.0 * 16.0 / sr) << 4;
+    }
+    /// Set phase delta based on oscillator frequency and inverse of sample rate.
+    void set_freq_odsr(float freq, double odsr)
+    {
+        phasedelta = (int)(freq * 65536.0 * 256.0 * 16.0 * odsr) << 4;
+    }
+    inline float get()
+    {
+        float value = (phase >> 16 ) / 65535.0 - 0.5;
+        phase += phasedelta;
+        return value;
+    }
+};
+
+/**
+ * FFT-based bandlimiting helper class. Allows conversion between time and frequency domains and generating brickwall filtered
+ * versions of a waveform given a pre-computed spectrum.
+ * Waveform size must be a power of two, and template argument SIZE_BITS is log2 of waveform size.
+ */
+template<int SIZE_BITS>
+struct bandlimiter
+{
+    enum { SIZE = 1 << SIZE_BITS };
+    static dsp::fft<float, SIZE_BITS> &get_fft()
+    {
+        static dsp::fft<float, SIZE_BITS> fft;
+        return fft;
+    }
+    
+    std::complex<float> spectrum[SIZE];
+    
+    /// Import time domain waveform and calculate spectrum from it
+    void compute_spectrum(float input[SIZE])
+    {
+        dsp::fft<float, SIZE_BITS> &fft = get_fft();
+        std::complex<float> *data = new std::complex<float>[SIZE];
+        for (int i = 0; i < SIZE; i++)
+            data[i] = input[i];
+        fft.calculate(data, spectrum, false);
+        delete []data;
+    }
+    
+    /// Generate the waveform from the contained spectrum.
+    void compute_waveform(float output[SIZE])
+    {
+        dsp::fft<float, SIZE_BITS> &fft = get_fft();
+        std::complex<float> *data = new std::complex<float>[SIZE];
+        fft.calculate(spectrum, data, true);
+        for (int i = 0; i < SIZE; i++)
+            output[i] = data[i].real();
+        delete []data;
+    }
+    
+    /// remove DC offset of the spectrum (it usually does more harm than good!)
+    void remove_dc()
+    {
+        spectrum[0] = 0.f;
+    }
+    
+    /// Very basic bandlimiting (brickwall filter)
+    /// might need to be improved much in future!
+    void make_waveform(float output[SIZE], int cutoff, bool foldover = false)
+    {
+        dsp::fft<float, SIZE_BITS> &fft = get_fft();
+        std::vector<std::complex<float> > new_spec, iffted;
+        new_spec.resize(SIZE);
+        iffted.resize(SIZE);
+        // Copy original harmonics up to cutoff point
+        new_spec[0] = spectrum[0];
+        for (int i = 1; i < cutoff; i++)
+            new_spec[i] = spectrum[i], 
+            new_spec[SIZE - i] = spectrum[SIZE - i];
+        // Fill the rest with zeros, optionally folding over harmonics over the
+        // cutoff point into the lower octaves while halving the amplitude.
+        // (I think it is almost nice for bell type waveforms when the original
+        // waveform has few widely spread harmonics)
+        if (foldover)
+        {
+            std::complex<float> fatt(0.5);
+            cutoff /= 2;
+            if (cutoff < 2)
+                cutoff = 2;
+            for (int i = SIZE / 2; i >= cutoff; i--)
+            {
+                new_spec[i / 2] += new_spec[i] * fatt;
+                new_spec[SIZE - i / 2] += new_spec[SIZE - i] * fatt;
+                new_spec[i] = 0.f,
+                new_spec[SIZE - i] = 0.f;
+            }
+        }
+        else
+        {
+            if (cutoff < 1)
+                cutoff = 1;
+            for (int i = cutoff; i < SIZE / 2; i++)
+                new_spec[i] = 0.f,
+                new_spec[SIZE - i] = 0.f;
+        }
+        // convert back to time domain (IFFT) and extract only real part
+        fft.calculate(new_spec.data(), iffted.data(), true);
+        for (int i = 0; i < SIZE; i++)
+            output[i] = iffted[i].real();
+    }
+};
+
+/// Set of bandlimited wavetables
+template<int SIZE_BITS>
+struct waveform_family: public std::map<uint32_t, float *>
+{
+    enum { SIZE = 1 << SIZE_BITS };
+    using std::map<uint32_t, float *>::iterator;
+    using std::map<uint32_t, float *>::end;
+    using std::map<uint32_t, float *>::lower_bound;
+    float original[SIZE];
+    
+    /// Fill the family using specified bandlimiter and original waveform. Optionally apply foldover. 
+    /// Does not produce harmonics over specified limit (limit = (SIZE / 2) / min_number_of_harmonics)
+    void make(bandlimiter<SIZE_BITS> &bl, float input[SIZE], bool foldover = false, uint32_t limit = SIZE / 2)
+    {
+        memcpy(original, input, sizeof(original));
+        bl.compute_spectrum(input);
+        make_from_spectrum(bl, foldover);
+    }
+    
+    /// Fill the family using specified bandlimiter and spectrum contained within. Optionally apply foldover. 
+    /// Does not produce harmonics over specified limit (limit = (SIZE / 2) / min_number_of_harmonics)
+    void make_from_spectrum(bandlimiter<SIZE_BITS> &bl, bool foldover = false, uint32_t limit = SIZE / 2)
+    {
+        bl.remove_dc();
+        
+        uint32_t base = 1 << (32 - SIZE_BITS);
+        uint32_t cutoff = SIZE / 2, top = SIZE / 2;
+        float vmax = 0;
+        for (unsigned int i = 0; i < cutoff; i++)
+            vmax = std::max(vmax, abs(bl.spectrum[i]));
+        float vthres = vmax / 1024.0;  // -60dB
+        float cumul = 0.f;
+        while(cutoff > (SIZE / limit)) {
+            if (!foldover)
+            {
+                // skip harmonics too quiet to be heard, but measure their loudness cumulatively,
+                // because even if a single harmonic is too quiet, a whole bunch of them may add up 
+                // to considerable amount of space
+                cumul = 0.f;
+                while(cutoff > 1 && cumul + abs(bl.spectrum[cutoff - 1]) < vthres)
+                {
+                    cumul += abs(bl.spectrum[cutoff - 1]);
+                    cutoff--;
+                }
+            }
+            float *wf = new float[SIZE+1];
+            bl.make_waveform(wf, cutoff, foldover);
+            wf[SIZE] = wf[0];
+            (*this)[base * (top / cutoff)] = wf;
+            cutoff = (int)(0.75 * cutoff);
+        }
+    }
+    
+    /// Retrieve waveform pointer suitable for specified phase_delta
+    inline float *get_level(uint32_t phase_delta)
+    {
+        iterator i = upper_bound(phase_delta);
+        if (i == end())
+            return NULL;
+        // printf("Level = %08x\n", i->first);
+        return i->second;
+    }
+    /// Destructor, deletes the waveforms and removes them from the map.
+    ~waveform_family()
+    {
+        for (iterator i = begin(); i != end(); i++)
+            delete []i->second;
+        clear();
+    }
+};
+
+#if 0
+// cubic interpolation
+static inline float cerp(float pm1, float p0, float p1, float p2, float t)
+{
+    return (-t*(t-1)*(t-2) * pm1 + 3*(t+1)*(t-1)*(t-2) * p0 - 3*(t+1)*t*(t-2) * p1 + (t+1)*t*(t-1) * p2) * (1.0 / 6.0);
+}
+#endif
+/**
+ * Simple table-based lerping oscillator. Uses waveform of size 2^SIZE_BITS.
+ * Combine with waveform_family if bandlimited waveforms are needed. Because
+ * of linear interpolation, it's usually a good idea to use large tables
+ * (2048-4096 points), otherwise aliasing may be produced.
+ */
+template<int SIZE_BITS>
+struct waveform_oscillator: public simple_oscillator
+{
+    enum { SIZE = 1 << SIZE_BITS, MASK = SIZE - 1, SCALE = 1 << (32 - SIZE_BITS) };
+    float *waveform;
+    inline float get()
+    {
+        uint32_t wpos = phase >> (32 - SIZE_BITS);
+        float value = dsp::lerp(waveform[wpos], waveform[(wpos + 1) & MASK], (phase & (SCALE - 1)) * (1.0f / SCALE));
+        phase += phasedelta;
+        return value;
+    }
+    /// Add/substract two phase-shifted values
+    inline float get_phaseshifted(uint32_t shift, float mix)
+    {
+        uint32_t wpos = phase >> (32 - SIZE_BITS);
+        float value1 = dsp::lerp(waveform[wpos], waveform[(wpos + 1) & MASK], (phase & (SCALE - 1)) * (1.0f / SCALE));
+        wpos = (phase + shift) >> (32 - SIZE_BITS);
+        float value2 = dsp::lerp(waveform[wpos], waveform[(wpos + 1) & MASK], ((phase + shift) & (SCALE - 1)) * (1.0f / SCALE));
+        float value = value1 + mix * value2;
+        phase += phasedelta;
+        return value;
+    }
+};
+
+/**
+ * Simple triangle LFO without any smoothing or anything of this sort. 
+ */
+struct triangle_lfo: public simple_oscillator
+{
+    inline float get()
+    {
+        uint32_t phase2 = phase;
+        // start at 90 degrees point of the "/\" wave (-1 to +1)
+        phase2 += 1<<30; 
+        // if in second half, invert the wave (so it falls back into 0..0x7FFFFFFF)
+        phase2 ^= ((int32_t)phase2)>>31;
+        
+        float value = (phase2 >> 6) / 16777216.0 - 1.0;
+        phase += phasedelta;
+        return value;
+    }
+};
+
+/// Simple stupid inline function to normalize a waveform (by removing DC offset and ensuring max absolute value of 1).
+static inline void normalize_waveform(float *table, unsigned int size)
+{
+    float dc = 0;
+    for (unsigned int i = 0; i < size; i++)
+        dc += table[i];
+    dc /= size;
+    for (unsigned int i = 0; i < size; i++)
+        table[i] -= dc;
+    float thismax = 0;
+    for (unsigned int i = 0; i < size; i++)
+        thismax = std::max(thismax, fabsf(table[i]));
+    if (thismax < 0.000001f)
+        return;
+    double divv = 1.0 / thismax;
+    for (unsigned int i = 0; i < size; i++)
+        table[i] *= divv;
+}
+
+
+
+};
+
+#endif

plugins/ladspa_effect/calf/calf/osctl.h

@@ -0,0 +1,502 @@
+/* Calf DSP Library
+ * Open Sound Control primitives
+ *
+ * Copyright (C) 2007 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __CALF_OSCTL_H
+#define __CALF_OSCTL_H
+
+#include <string>
+#include <vector>
+#include <string.h>
+#include <errno.h>
+#include <iostream>
+#include <stdint.h>
+
+namespace osctl
+{
+    
+enum osc_type
+{
+    osc_i32 = 'i',
+    osc_f32 = 'f',
+    osc_string = 's',
+    osc_blob = 'b',
+    
+    // unsupported
+    osc_i64 = 'h',
+    osc_ts = 't',
+    osc_f64 = 'd',
+    osc_string_alt = 'S',
+    osc_char = 'c',
+    osc_rgba = 'r',
+    osc_midi = 'm',
+    osc_true = 'T',
+    osc_false = 'F',
+    osc_nil = 'N',
+    osc_inf = 'I',
+    osc_start_array = '[',
+    osc_end_array = ']'
+};
+
+extern const char *osc_type_name(osc_type type);
+
+struct osc_exception: public std::exception
+{
+    virtual const char *what() const throw() { return "OSC parsing error"; }
+};
+
+struct osc_read_exception: public std::exception
+{
+    virtual const char *what() const throw() { return "OSC buffer underflow"; }
+};
+
+struct osc_write_exception: public std::exception
+{
+    virtual const char *what() const throw() { return "OSC buffer overflow"; }
+};
+
+struct null_buffer
+{
+    static bool read(uint8_t *dest, uint32_t bytes)
+    {
+        return false;
+    }
+    static bool write(uint8_t *dest, uint32_t bytes)
+    {
+        return true;
+    }
+    static void clear()
+    {
+    }
+};
+
+struct raw_buffer
+{
+    uint8_t *ptr;
+    uint32_t pos, count, size;
+    
+    raw_buffer()
+    {
+        ptr = NULL;
+        pos = count = size = 0;
+    }
+    raw_buffer(uint8_t *_ptr, uint32_t _count, uint32_t _size)
+    {
+        set(_ptr, _count, _size);
+    }
+    inline void set(uint8_t *_ptr, uint32_t _count, uint32_t _size)
+    {
+        ptr = _ptr;
+        pos = 0;
+        count = _count;
+        size = _size;
+    }
+    bool read(uint8_t *dest, uint32_t bytes)
+    {
+        if (pos + bytes > count)
+            return false;
+        memcpy(dest, ptr + pos, bytes);
+        pos += bytes;
+        return true;
+    }
+    bool write(const uint8_t *src, uint32_t bytes)
+    {
+        if (count + bytes > size)
+            return false;
+        memcpy(ptr + count, src, bytes);
+        count += bytes;
+        return true;
+    }
+    int read_left()
+    {
+        return count - pos;
+    }
+    int write_left()
+    {
+        return size - count;
+    }
+    inline int write_misalignment()
+    {
+        return 4 - (count & 3);
+    }
+    void clear()
+    {
+        pos = 0;
+        count = 0;
+    }
+    int tell()
+    {
+        return pos;
+    }
+    void seek(int _pos)
+    {
+        pos = _pos;
+    }
+};
+
+struct string_buffer
+{
+    std::string data;
+    uint32_t pos, size;
+    
+    string_buffer()
+    {
+        pos = 0;
+        size = 1048576;
+    }
+    string_buffer(std::string _data, int _size = 1048576)
+    {
+        data = _data;
+        pos = 0;
+        size = _size;
+    }
+    bool read(uint8_t *dest, uint32_t bytes)
+    {
+        if (pos + bytes > data.length())
+            return false;
+        memcpy(dest, &data[pos], bytes);
+        pos += bytes;
+        return true;
+    }
+    bool write(const uint8_t *src, uint32_t bytes)
+    {
+        if (data.length() + bytes > size)
+            return false;
+        uint32_t wpos = data.length();
+        data.resize(wpos + bytes);
+        memcpy(&data[wpos], src, bytes);
+        return true;
+    }
+    inline int read_left()
+    {
+        return data.length() - pos;
+    }
+    inline int write_left()
+    {
+        return size - data.length();
+    }
+    inline int write_misalignment()
+    {
+        return 4 - (data.length() & 3);
+    }
+    void clear()
+    {
+        data.clear();
+        pos = 0;
+    }
+    int tell()
+    {
+        return pos;
+    }
+    void seek(int _pos)
+    {
+        pos = _pos;
+    }
+};
+
+template<class Buffer, class TypeBuffer = null_buffer, bool Throw = true>
+struct osc_stream
+{
+    Buffer &buffer;
+    TypeBuffer *type_buffer;
+    bool error;
+    
+    osc_stream(Buffer &_buffer) : buffer(_buffer), type_buffer(NULL), error(false) {}
+    osc_stream(Buffer &_buffer, TypeBuffer &_type_buffer) : buffer(_buffer), type_buffer(&_type_buffer), error(false) {}
+    inline void pad()
+    {
+        uint32_t zero = 0;
+        write(&zero, buffer.write_misalignment());
+    }
+    inline void read(void *dest, uint32_t bytes)
+    {
+        if (!buffer.read((uint8_t *)dest, bytes))
+        {
+#if 0
+            if (Throw)
+                throw osc_read_exception();
+            else
+#endif
+            {
+                error = true;
+                memset(dest, 0, bytes);
+            }
+        }
+    }
+    inline void write(const void *src, uint32_t bytes)
+    {
+        if (!buffer.write((const uint8_t *)src, bytes))
+        {
+#if 0
+            if (Throw)
+                throw osc_write_exception();
+            else
+#endif
+                error = true;
+        }
+    }
+    inline void clear()
+    {
+        buffer.clear();
+        if (type_buffer)
+            type_buffer->clear();
+    }
+    inline void write_type(char ch)
+    {
+        if (type_buffer)
+            type_buffer->write((uint8_t *)&ch, 1);
+    }
+};
+
+typedef osc_stream<string_buffer> osc_strstream;
+typedef osc_stream<string_buffer, string_buffer> osc_typed_strstream;
+
+struct osc_inline_strstream: public string_buffer, public osc_strstream
+{
+    osc_inline_strstream()
+    : string_buffer(), osc_strstream(static_cast<string_buffer &>(*this))
+    {
+    }
+};
+
+struct osc_str_typed_buffer_pair
+{
+    string_buffer buf_data, buf_types;
+};
+
+struct osc_inline_typed_strstream: public osc_str_typed_buffer_pair, public osc_typed_strstream
+{
+    osc_inline_typed_strstream()
+    : osc_str_typed_buffer_pair(), osc_typed_strstream(buf_data, buf_types)
+    {
+    }
+};
+
+template<class Buffer, class TypeBuffer>
+inline osc_stream<Buffer, TypeBuffer> &
+operator <<(osc_stream<Buffer, TypeBuffer> &s, uint32_t val)
+{
+    return s;
+}
+
+template<class Buffer, class TypeBuffer>
+inline osc_stream<Buffer, TypeBuffer> &
+operator >>(osc_stream<Buffer, TypeBuffer> &s, uint32_t &val)
+{
+    return s;
+}
+
+template<class Buffer, class TypeBuffer>
+inline osc_stream<Buffer, TypeBuffer> &
+operator >>(osc_stream<Buffer, TypeBuffer> &s, int32_t &val)
+{
+    return s;
+}
+
+template<class Buffer, class TypeBuffer>
+inline osc_stream<Buffer, TypeBuffer> &
+operator <<(osc_stream<Buffer, TypeBuffer> &s, float val)
+{
+    return s;
+}
+
+template<class Buffer, class TypeBuffer>
+inline osc_stream<Buffer, TypeBuffer> &
+operator >>(osc_stream<Buffer, TypeBuffer> &s, float &val)
+{
+    return s;
+}
+
+template<class Buffer, class TypeBuffer>
+inline osc_stream<Buffer, TypeBuffer> &
+operator <<(osc_stream<Buffer, TypeBuffer> &s, const std::string &str)
+{
+    s.write(&str[0], str.length());
+    s.pad();
+    s.write_type(osc_string);
+    return s;
+}
+
+template<class Buffer, class TypeBuffer>
+inline osc_stream<Buffer, TypeBuffer> &
+operator >>(osc_stream<Buffer, TypeBuffer> &s, std::string &str)
+{
+    // inefficient...
+    char five[5];
+    five[4] = '\0';
+    str.resize(0);
+    while(1)
+    {
+        s.read(five, 4);
+        if (five[0] == '\0')
+            break;
+        str += five;
+        if (!five[1] || !five[2] || !five[3])
+            break;
+    }
+    return s;
+}
+
+template<class Buffer, class TypeBuffer, class DestBuffer>
+inline osc_stream<Buffer, TypeBuffer> &
+read_buffer_from_osc_stream(osc_stream<Buffer, TypeBuffer> &s, DestBuffer &buf)
+{
+    return s;
+}
+
+template<class Buffer, class TypeBuffer, class SrcBuffer>
+inline osc_stream<Buffer, TypeBuffer> &
+write_buffer_to_osc_stream(osc_stream<Buffer, TypeBuffer> &s, SrcBuffer &buf)
+{
+    return s;
+}
+
+template<class Buffer, class TypeBuffer>
+inline osc_stream<Buffer, TypeBuffer> &
+operator >>(osc_stream<Buffer, TypeBuffer> &s, raw_buffer &str)
+{
+    return read_buffer_from_osc_stream(s, str);
+}
+
+template<class Buffer, class TypeBuffer>
+inline osc_stream<Buffer, TypeBuffer> &
+operator >>(osc_stream<Buffer, TypeBuffer> &s, string_buffer &str)
+{
+    return read_buffer_from_osc_stream(s, str);
+}
+
+template<class Buffer, class TypeBuffer>
+inline osc_stream<Buffer, TypeBuffer> &
+operator <<(osc_stream<Buffer, TypeBuffer> &s, raw_buffer &str)
+{
+    return write_buffer_to_osc_stream(s, str);
+}
+
+template<class Buffer, class TypeBuffer>
+inline osc_stream<Buffer, TypeBuffer> &
+operator <<(osc_stream<Buffer, TypeBuffer> &s, string_buffer &str)
+{
+    return write_buffer_to_osc_stream(s, str);
+}
+
+// XXXKF: I don't support reading binary blobs yet
+#if 0
+struct osc_net_bad_address: public std::exception
+{
+    std::string addr, error_msg;
+    osc_net_bad_address(const char *_addr)
+    {
+        addr = _addr;
+        error_msg = "Incorrect OSC URI: " + addr;
+    }
+    virtual const char *what() const throw() { return error_msg.c_str(); }
+    virtual ~osc_net_bad_address() throw () {}
+};
+
+struct osc_net_exception: public std::exception
+{
+    int net_errno;
+    std::string command, error_msg;
+    osc_net_exception(const char *cmd, int _errno = errno)
+    {
+        command = cmd;
+        net_errno = _errno;
+        error_msg = "OSC error in "+command+": "+strerror(_errno);
+    }
+    virtual const char *what() const throw() { return error_msg.c_str(); }
+    virtual ~osc_net_exception() throw () {}
+};
+    
+struct osc_net_dns_exception: public std::exception
+{
+    int net_errno;
+    std::string command, error_msg;
+    virtual const char *what() const throw() { return error_msg.c_str(); }
+    virtual ~osc_net_dns_exception() throw () {}
+};
+#endif
+    
+template<class OscStream>
+struct osc_message_sink
+{
+    virtual void receive_osc_message(std::string address, std::string type_tag, OscStream &buffer)=0;
+    virtual ~osc_message_sink() {}
+};
+
+template<class OscStream, class DumpStream>
+struct osc_message_dump: public osc_message_sink<OscStream>
+{
+    DumpStream &stream;
+    osc_message_dump(DumpStream &_stream) : stream(_stream) {}
+        
+    virtual void receive_osc_message(std::string address, std::string type_tag, OscStream &buffer)
+    {
+        int pos = buffer.buffer.tell();
+        stream << "address: " << address << ", type tag: " << type_tag << std::endl;
+        for (unsigned int i = 0; i < type_tag.size(); i++)
+        {
+            stream << "Argument " << i << " is ";
+            switch(type_tag[i])
+            {
+                case 'i': 
+                {
+                    uint32_t val;
+                    buffer >> val;
+                    stream << val;
+                    break;
+                }
+                case 'f': 
+                {
+                    float val;
+                    buffer >> val;
+                    stream << val;
+                    break;
+                }
+                case 's': 
+                {
+                    std::string val;
+                    buffer >> val;
+                    stream << val;
+                    break;
+                }
+                case 'b': 
+                {
+                    osctl::string_buffer val;
+                    buffer >> val;
+                    stream << "blob (" << val.data.length() << " bytes)";
+                    break;
+                }
+                default:
+                {
+                    stream << "unknown - cannot parse more arguments" << std::endl;
+                    i = type_tag.size();
+                    break;
+                }
+            }
+            stream << std::endl;
+        }
+        stream << std::flush;
+        buffer.buffer.seek(pos);
+    }
+};
+
+};
+
+#endif

plugins/ladspa_effect/calf/calf/plugininfo.h

@@ -0,0 +1,105 @@
+/* Calf DSP Library
+ * Plugin introspection interface
+ *
+ * Copyright (C) 2008 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __CALF_PLUGININFO_H
+#define __CALF_PLUGININFO_H
+
+#include <string>
+
+namespace calf_plugins {
+
+/// A sink to send information about an audio port
+struct plain_port_info_iface
+{
+    /// Called if it's an input port
+    virtual plain_port_info_iface& input() { return *this; }
+    /// Called if it's an output port
+    virtual plain_port_info_iface& output() { return *this; }
+    virtual plain_port_info_iface& lv2_ttl(const std::string &text) { return *this; }
+    virtual ~plain_port_info_iface() {}
+};
+
+/// A sink to send information about a control port (very incomplete, missing stuff: units, integer, boolean, toggled, notAutomatic, notGUI...)
+struct control_port_info_iface
+{
+    /// Called if it's an input port
+    virtual control_port_info_iface& input() { return *this; }
+    /// Called if it's an output port
+    virtual control_port_info_iface& output() { return *this; }
+    /// Called to mark the port as using linear range [from, to]
+    virtual control_port_info_iface& lin_range(double from, double to) { return *this; }
+    /// Called to mark the port as using log range [from, to]
+    virtual control_port_info_iface& log_range(double from, double to) { return *this; }
+    virtual control_port_info_iface& toggle() { return *this; }
+    virtual control_port_info_iface& trigger() { return *this; }
+    virtual control_port_info_iface& integer() { return *this; }
+    virtual control_port_info_iface& lv2_ttl(const std::string &text) { return *this; }
+    virtual control_port_info_iface& polymorphic() { return lv2_ttl("a poly:PolymorphicPort ;"); }
+    virtual control_port_info_iface& poly_audio() { return lv2_ttl("poly:supportsType lv2:AudioPort ;"); }
+    virtual ~control_port_info_iface() {}
+};
+
+/// A sink to send information about a plugin
+struct plugin_info_iface
+{
+    /// Set plugin names (ID, name and label)
+    virtual void names(const std::string &name, const std::string &label, const std::string &category, const std::string &microname = std::string()) {}
+    /// Add an audio port (returns a sink which accepts further description)
+    virtual plain_port_info_iface &audio_port(const std::string &id, const std::string &name, const std::string &microname = std::string("N/A"))=0;
+    /// Add an event port (returns a sink which accepts further description)
+    virtual plain_port_info_iface &event_port(const std::string &id, const std::string &name, const std::string &microname = std::string("N/A"))=0;
+    /// Add a control port (returns a sink which accepts further description)
+    virtual control_port_info_iface &control_port(const std::string &id, const std::string &name, double def_value, const std::string &microname = "N/A")=0;
+    /// Add arbitrary TTL clauses
+    virtual void lv2_ttl(const std::string &text) {}
+    /// Add small plugin GUI
+    virtual void has_gui() { lv2_ttl("uiext:ui <http://calf.sourceforge.net/small_plugins/gui/gtk2-gui> ;"); }
+    /// Called after plugin has reported all the information
+    virtual void finalize() {}
+    virtual ~plugin_info_iface() {}
+};
+
+struct plugin_port_type_grabber: public plugin_info_iface, public control_port_info_iface
+{
+    uint32_t &target;
+    uint32_t index;
+    
+    plain_port_info_iface pp;
+    control_port_info_iface cp;
+    
+    plugin_port_type_grabber(uint32_t &_target) : target(_target), index(0) { target = 0; }
+        
+    virtual plain_port_info_iface &audio_port(const std::string &id, const std::string &name, const std::string &microname = std::string("N/A")) { target |= (1 << index); index++; return pp; }
+    virtual plain_port_info_iface &event_port(const std::string &id, const std::string &name, const std::string &microname = std::string("N/A")) { index++; return pp; }
+    virtual control_port_info_iface &control_port(const std::string &id, const std::string &name, double def_value, const std::string &microname = "N/A") { index++; return cp; }
+};
+
+/// A sink to send information about plugins
+struct plugin_list_info_iface
+{
+    /// Add an empty plugin object and return the sink to be filled with information
+    virtual plugin_info_iface &plugin(const std::string &id) = 0;
+    virtual ~plugin_list_info_iface() {}
+};
+
+};
+
+#endif

plugins/ladspa_effect/calf/calf/preset.h

@@ -0,0 +1,142 @@
+/* Calf DSP Library
+ * Preset management
+ *
+ * Copyright (C) 2007 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+#ifndef __CALF_PRESET_H
+#define __CALF_PRESET_H
+
+#include <vector>
+#include <string>
+#include <map>
+#include <sstream>
+#include <ostream>
+#include <string.h>
+#include "utils.h"
+
+namespace calf_plugins {
+
+class plugin_ctl_iface;
+    
+/// Contents of single preset
+struct plugin_preset
+{
+    /// Bank the preset belongs to (not used yet)
+    int bank;
+    /// Program number of the preset (not used yet)
+    int program;
+    /// Name of the preset
+    std::string name;
+    /// Name of the plugin the preset is for
+    std::string plugin;
+    /// Names of parameters in values array (for each item in param_names there should be a counterpart in values)
+    std::vector<std::string> param_names;
+    /// Values of parameters
+    std::vector<float> values;
+    /// DSSI configure-style variables
+    std::map<std::string, std::string> variables;
+
+    plugin_preset() : bank(0), program(0) {}
+    /// Export preset as XML
+    std::string to_xml();   
+    /// "Upload" preset content to the plugin
+    void activate(plugin_ctl_iface *plugin);
+    /// "Download" preset content from the plugin
+    void get_from(plugin_ctl_iface *plugin);
+        
+    std::string get_safe_name();
+};
+
+/// Exception thrown by preset system
+struct preset_exception
+{
+    std::string message, param, fulltext;
+    int error;
+    preset_exception(const std::string &_message, const std::string &_param, int _error)
+    : message(_message), param(_param), error(_error)
+    {
+    }
+    const char *what() {
+        if (error)
+            fulltext = message + " " + param + " (" + strerror(error) + ")";
+        else
+            fulltext = message + " " + param;
+        return fulltext.c_str();
+    }
+    ~preset_exception()
+    {
+    }
+};
+
+/// A vector of presets
+typedef std::vector<plugin_preset> preset_vector;
+
+/// A single list of presets (usually there are two - @see get_builtin_presets(), get_user_presets() )
+struct preset_list
+{
+    /// Parser states
+    enum parser_state
+    {
+        START, ///< Beginning of parsing process (before root element)
+        LIST, ///< Inside root element
+        PRESET, ///< Inside preset definition
+        VALUE, ///< Inside (empty) param tag
+        VAR, ///< Inside (non-empty) var tag
+    } state;
+
+    /// Contained presets (usually for all plugins)
+    preset_vector presets;
+    /// Temporary preset used during parsing process
+    plugin_preset parser_preset;
+    /// Preset number counters for DSSI (currently broken)
+    std::map<std::string, int> last_preset_ids;
+    /// The key used in current <var name="key"> tag (for state == VAR)
+    std::string current_key;
+
+    /// Return the name of the built-in or user-defined preset file
+    static std::string get_preset_filename(bool builtin);
+    /// Load default preset list (built-in or user-defined)
+    bool load_defaults(bool builtin);
+    void parse(const std::string &data);
+    /// Load preset list from XML file
+    void load(const char *filename);
+    /// Save preset list as XML file
+    void save(const char *filename);
+    /// Append or replace a preset (replaces a preset with the same plugin and preset name)
+    void add(const plugin_preset &sp);
+    /// Get a sublist of presets for a given plugin (those with plugin_preset::plugin == plugin)
+    void get_for_plugin(preset_vector &vec, const char *plugin);
+    
+protected:
+    /// Internal function: start element handler for expat
+    static void xml_start_element_handler(void *user_data, const char *name, const char *attrs[]);
+    /// Internal function: end element handler for expat
+    static void xml_end_element_handler(void *user_data, const char *name);
+    /// Internal function: character data (tag text content) handler for expat
+    static void xml_character_data_handler(void *user_data, const char *data, int len);
+};
+
+/// Return the current list of built-in (factory) presets (these are loaded from system-wide file)
+extern preset_list &get_builtin_presets();
+
+/// Return the current list of user-defined presets (these are loaded from ~/.calfpresets)
+extern preset_list &get_user_presets();
+
+};
+
+#endif

plugins/ladspa_effect/calf/calf/primitives.h

@@ -0,0 +1,516 @@
+/* Calf DSP Library
+ * DSP primitives.
+ *
+ * Copyright (C) 2001-2007 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+#ifndef __CALF_PRIMITIVES_H
+#define __CALF_PRIMITIVES_H
+
+#include <stack>
+#include <map>
+#include <cmath>
+#include <cstdlib>
+#include <stdint.h>
+#include <stdio.h>
+
+namespace dsp {
+
+/// Set a float to zero
+inline void zero(float &v) {
+    v = 0;
+};
+
+/// Set a double to zero
+inline void zero(double &v) {
+    v = 0;
+};
+
+/// Set 64-bit unsigned integer value to zero
+inline void zero(uint64_t &v) { v = 0; };
+/// Set 32-bit unsigned integer value to zero
+inline void zero(uint32_t &v) { v = 0; };
+/// Set 16-bit unsigned integer value to zero
+inline void zero(uint16_t &v) { v = 0; };
+/// Set 8-bit unsigned integer value to zero
+inline void zero(uint8_t &v) { v = 0; };
+/// Set 64-bit signed integer value to zero
+inline void zero(int64_t &v) { v = 0; };
+/// Set 32-bit signed integer value to zero
+inline void zero(int32_t &v) { v = 0; };
+/// Set 16-bit signed integer value to zero
+inline void zero(int16_t &v) { v = 0; };
+/// Set 8-bit signed integer value to zero
+inline void zero(int8_t &v) { v = 0; };
+
+/// Set array (buffer or anything similar) to vector of zeroes
+template<class T>
+void zero(T *data, unsigned int size) {
+    T value;
+    dsp::zero(value);
+    for (unsigned int i=0; i<size; i++)
+        *data++ = value;
+}
+
+template<class T = float>struct stereo_sample {
+    T left;
+    T right;
+    /// default constructor - preserves T's semantics (ie. no implicit initialization to 0)
+    inline stereo_sample() {
+    }
+    inline stereo_sample(T _left, T _right) {
+        left = _left;
+        right = _right;
+    }
+    inline stereo_sample(T _both) {
+        left = right = _both;
+    }
+    template<typename U>
+    inline stereo_sample(const stereo_sample<U> &value) {
+        left = value.left;
+        right = value.right;
+    }
+    inline stereo_sample& operator=(const T &value) {
+        left = right = value;
+        return *this;
+    }
+    template<typename U>
+    inline stereo_sample& operator=(const stereo_sample<U> &value) {
+        left = value.left;
+        right = value.right;
+        return *this;
+    }
+/*
+    inline operator T() const {
+        return (left+right)/2;
+    }
+*/
+    inline stereo_sample& operator*=(const T &multiplier) {
+        left *= multiplier;
+        right *= multiplier;
+        return *this;
+    }
+    inline stereo_sample& operator+=(const stereo_sample<T> &value) {
+        left += value.left;
+        right += value.right;
+        return *this;
+    }
+    inline stereo_sample& operator-=(const stereo_sample<T> &value) {
+        left -= value.left;
+        right -= value.right;
+        return *this;
+    }
+    template<typename U> inline stereo_sample<U> operator*(const U &value) const {
+        return stereo_sample<U>(left*value, right*value);
+    }
+    /*inline stereo_sample<float> operator*(float value) const {
+        return stereo_sample<float>(left*value, right*value);
+    }
+    inline stereo_sample<double> operator*(double value) const {
+        return stereo_sample<double>(left*value, right*value);
+    }*/
+    inline stereo_sample<T> operator+(const stereo_sample<T> &value) {
+        return stereo_sample(left+value.left, right+value.right);
+    }
+    inline stereo_sample<T> operator-(const stereo_sample<T> &value) {
+        return stereo_sample(left-value.left, right-value.right);
+    }
+    inline stereo_sample<T> operator+(const T &value) {
+        return stereo_sample(left+value, right+value);
+    }
+    inline stereo_sample<T> operator-(const T &value) {
+        return stereo_sample(left-value, right-value);
+    }
+    inline stereo_sample<float> operator+(float value) {
+        return stereo_sample<float>(left+value, right+value);
+    }
+    inline stereo_sample<float> operator-(float value) {
+        return stereo_sample<float>(left-value, right-value);
+    }
+    inline stereo_sample<double> operator+(double value) {
+        return stereo_sample<double>(left+value, right+value);
+    }
+    inline stereo_sample<double> operator-(double value) {
+        return stereo_sample<double>(left-value, right-value);
+    }
+};
+
+/// Multiply constant by stereo_value
+template<class T>
+inline stereo_sample<T> operator*(const T &value, const stereo_sample<T> &value2) {
+    return stereo_sample<T>(value2.left*value, value2.right*value);
+}
+
+/// Add constant to stereo_value
+template<class T>
+inline stereo_sample<T> operator+(const T &value, const stereo_sample<T> &value2) {
+    return stereo_sample<T>(value2.left+value, value2.right+value);
+}
+
+/// Subtract stereo_value from constant (yields stereo_value of course)
+template<class T>
+inline stereo_sample<T> operator-(const T &value, const stereo_sample<T> &value2) {
+    return stereo_sample<T>(value-value2.left, value-value2.right);
+}
+
+/// Shift value right by 'bits' bits (multiply by 2^-bits)
+template<typename T>
+inline stereo_sample<T> shr(stereo_sample<T> v, int bits = 1) {
+    v.left = shr(v.left, bits);
+    v.right = shr(v.right, bits);
+    return v;
+}
+
+/// Set a stereo_sample<T> value to zero
+template<typename T> 
+inline void zero(stereo_sample<T> &v) {
+    dsp::zero(v.left);
+    dsp::zero(v.right);
+}
+
+/// 'Small value' for integer and other types
+template<typename T>
+inline T small_value() {
+    return 0;
+}
+
+/// 'Small value' for floats (2^-24) - used for primitive underrun prevention. The value is pretty much arbitrary (allowing for 24-bit signals normalized to 1.0).
+template<>
+inline float small_value<float>() {
+    return (1.0/16777216.0); // allows for 2^-24, should be enough for 24-bit DACs at least :)
+}
+
+/// 'Small value' for doubles (2^-24) - used for primitive underrun prevention. The value is pretty much arbitrary.
+template<>
+inline double small_value<double>() {
+    return (1.0/16777216.0);
+}
+
+/// Convert a single value to single value = do nothing :) (but it's a generic with specialisation for stereo_sample)
+template<typename T> 
+inline float mono(T v) {
+    return v;
+}
+
+/// Convert a stereo_sample to single value by averaging two channels
+template<typename T> 
+inline T mono(stereo_sample<T> v) {
+    return shr(v.left+v.right);
+}
+
+/// Clip a value to [min, max]
+template<typename T> 
+inline T clip(T value, T min, T max) {
+    if (value < min) return min;
+    if (value > max) return max;
+    return value;
+}
+
+/// Clip a double to [-1.0, +1.0]
+inline double clip11(double value) {
+    double a = fabs(value);
+    if (a<=1) return value;
+    return (a<0) ? -1.0 : 1.0;
+}
+
+/// Clip a float to [-1.0f, +1.0f]
+inline float clip11(float value) {
+    float a = fabsf(value);
+    if (a<=1) return value;
+    return (a<0) ? -1.0f : 1.0f;
+}
+
+/// Clip a double to [0.0, +1.0]
+inline double clip01(double value) {
+    double a = fabs(value-0.5);
+    if (a<=0.5) return value;
+    return (a<0) ? -0.0 : 1.0;
+}
+
+/// Clip a float to [0.0f, +1.0f]
+inline float clip01(float value) {
+    float a = fabsf(value-0.5f);
+    if (a<=0.5f) return value;
+    return (a<0) ? -0.0f : 1.0f;
+}
+
+// Linear interpolation (mix-way between v1 and v2).
+template<typename T, typename U>
+inline T lerp(T v1, T v2, U mix) {
+    return v1+(v2-v1)*mix;
+}
+
+// Linear interpolation for stereo values (mix-way between v1 and v2).
+template<typename T>
+inline stereo_sample<T> lerp(stereo_sample<T> &v1, stereo_sample<T> &v2, float mix) {
+    return stereo_sample<T>(v1.left+(v2.left-v1.left)*mix, v1.right+(v2.right-v1.right)*mix);
+}
+
+/**
+ * decay-only envelope (linear or exponential); deactivates itself when it goes below a set point (epsilon)
+ */
+class decay
+{
+    double value, initial;
+    unsigned int age, mask;
+    bool active;
+public:
+    decay() {
+        active = false;
+        mask = 127;
+        initial = value = 0.0;
+    }
+    inline bool get_active() {
+        return active;
+    }
+    inline double get() {
+        return active ? value : 0.0;
+    }
+    inline void set(double v) {
+        initial = value = v;
+        active = true;
+        age = 0;
+    }
+    /// reinitialise envelope (must be called if shape changes from linear to exponential or vice versa in the middle of envelope)
+    inline void reinit()
+    {
+        initial = value;
+        age = 1;
+    }
+    inline void add(double v) {
+        if (active)
+            value += v;
+        else
+            value = v;
+        initial = value;
+        age = 0;
+        active = true;
+    }
+    static inline double calc_exp_constant(double times, double cycles)
+    {
+        if (cycles < 1.0)
+            cycles = 1.0;
+        return pow(times, 1.0 / cycles);
+    }
+    inline void age_exp(double constant, double epsilon) {
+        if (active) {
+            if (!(age & mask))
+                value = initial * pow(constant, (double)age);
+            else
+                value *= constant;
+            if (value < epsilon)
+                active = false;
+            age++;
+        }
+    }
+    inline void age_lin(double constant, double epsilon) {
+        if (active) {
+            if (!(age & mask))
+                value = initial - constant * age;
+            else
+                value -= constant;
+            if (value < epsilon)
+                active = false;
+            age++;
+        }
+    }
+    inline void deactivate() {
+        active = false;
+        value = 0;
+    }
+};
+
+class scheduler;
+
+class task {
+public:
+    virtual void execute(scheduler *s)=0;
+    virtual void dispose() { delete this; }
+    virtual ~task() {}
+};
+
+/// this scheduler is based on std::multimap, so it isn't very fast, I guess
+/// maybe some day it should be rewritten to use heapsort or something
+/// work in progress, don't use!
+class scheduler {
+    std::multimap<unsigned int, task *> timeline;
+    unsigned int time, next_task;
+    bool eob;
+    class end_buf_task: public task {
+    public:
+        scheduler *p;
+        end_buf_task(scheduler *_p) : p(_p) {}
+        virtual void execute(scheduler *s) { p->eob = true; }
+        virtual void dispose() { }
+    } eobt;
+public:
+
+    scheduler()
+    : time(0)
+    , next_task((unsigned)-1)
+    , eob(true)
+    , eobt (this)
+    {
+        time = 0;
+        next_task = (unsigned)-1;
+        eob = false;
+    }
+    inline bool is_next_tick() {
+        if (time < next_task)
+            return true;
+        do_tasks();
+    }
+    inline void next_tick() {
+        time++;
+    }
+    void set(int pos, task *t) {
+        timeline.insert(std::pair<unsigned int, task *>(time+pos, t));
+        next_task = timeline.begin()->first;
+    }
+    void do_tasks() {
+        std::multimap<unsigned int, task *>::iterator i = timeline.begin();
+        while(i != timeline.end() && i->first == time) {
+            i->second->execute(this);
+            i->second->dispose();
+            timeline.erase(i);
+        }
+    }
+    bool is_eob() {
+        return eob;
+    }
+    void set_buffer_size(int count) {
+        set(count, &eobt);
+    }
+};
+
+/**
+ * Force "small enough" float value to zero
+ */
+inline void sanitize(float &value)
+{
+    if (std::abs(value) < small_value<float>())
+        value = 0.f;
+}
+
+/**
+ * Force "small enough" double value to zero
+ */
+inline void sanitize(double &value)
+{
+    if (std::abs(value) < small_value<double>())
+        value = 0.f;
+}
+
+/**
+ * Force "small enough" stereo value to zero
+ */
+template<class T>
+inline void sanitize(stereo_sample<T> &value)
+{
+    sanitize(value.left);
+    sanitize(value.right);
+}
+
+inline float fract16(unsigned int value)
+{
+    return (value & 0xFFFF) * (1.0 / 65536.0);
+}
+
+/**
+ * typical precalculated sine table
+ */
+template<class T, int N, int Multiplier>
+class sine_table
+{
+public:
+    static bool initialized;
+    static T data[N+1];
+    sine_table() {
+        if (initialized)
+            return;
+        initialized = true;
+        for (int i=0; i<N+1; i++)
+            data[i] = (T)(Multiplier*sin(i*2*M_PI*(1.0/N)));
+    }
+};
+
+template<class T, int N, int Multiplier>
+bool sine_table<T,N,Multiplier>::initialized = false;
+
+template<class T, int N, int Multiplier>
+T sine_table<T,N,Multiplier>::data[N+1];
+
+/// fast float to int conversion using default rounding mode
+inline int fastf2i_drm(float f)
+{
+#ifdef __X86__
+    volatile int v;
+    __asm ( "flds %1; fistpl %0" : "=m"(v) : "m"(f));
+    return v;
+#else
+    return (int)nearbyintf(f);
+#endif
+}
+
+/// Convert MIDI note to frequency in Hz.
+inline float note_to_hz(double note, double detune_cents = 0.0)
+{
+    return 440 * pow(2.0, (note - 69 + detune_cents/100.0) / 12.0);
+}
+
+/// Hermite interpolation between two points and slopes in normalized range (written after Wikipedia article)
+/// @arg t normalized x coordinate (0-1 over the interval in question)
+/// @arg p0 first point
+/// @arg p1 second point
+/// @arg m0 first slope (multiply by interval width when using over non-1-wide interval)
+/// @arg m1 second slope (multiply by interval width when using over non-1-wide interval)
+inline float normalized_hermite(float t, float p0, float p1, float m0, float m1)
+{
+    float t2 = t*t;
+    float t3 = t2*t;
+    return (2*t3 - 3*t2 + 1) * p0 + (t3 - 2*t2 + t) * m0 + (-2*t3 + 3*t2) * p1 + (t3-t2) * m1;
+}
+
+/// Hermite interpolation between two points and slopes 
+/// @arg x point within interval (x0 <= x <= x1)
+/// @arg x0 interval start
+/// @arg x1 interval end
+/// @arg p0 value at x0
+/// @arg p1 value at x1
+/// @arg m0 slope (steepness, tangent) at x0
+/// @arg m1 slope at x1
+inline float hermite_interpolation(float x, float x0, float x1, float p0, float p1, float m0, float m1)
+{
+    float width = x1 - x0;
+    float t = (x - x0) / width;
+    m0 *= width;
+    m1 *= width;
+    float t2 = t*t;
+    float t3 = t2*t;
+    
+    float ct0 = p0;
+    float ct1 = m0;
+    float ct2 = -3 * p0 - 2 * m0 + 3 * p1 - m1;
+    float ct3 = 2 * p0 + m0  - 2 * p1 + m1;
+    
+    return ct3 * t3 + ct2 * t2 + ct1 * t + ct0;
+    //return (2*t3 - 3*t2 + 1) * p0 + (t3 - 2*t2 + t) * m0 + (-2*t3 + 3*t2) * p1 + (t3-t2) * m1;
+}
+
+};
+
+#endif

plugins/ladspa_effect/calf/calf/synth.h

@@ -0,0 +1,228 @@
+/* Calf DSP Library
+ * Framework for synthesizer-like plugins. This is based
+ * on my earlier work on Drawbar electric organ emulator.
+ *
+ * Copyright (C) 2007 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+#ifndef __CALF_SYNTH_H
+#define __CALF_SYNTH_H
+
+#include <list>
+#include <stack>
+#include <bitset>
+#include "primitives.h"
+#include "audio_fx.h"
+
+namespace dsp {
+
+/**
+ * A kind of set with fast non-ordered iteration, used for storing lists of pressed keys.
+ */
+class keystack {
+private:
+    int dcount;
+    uint8_t active[128];
+    uint8_t states[128];
+public:
+    keystack() {
+        memset(states, 0xFF, sizeof(states));
+        dcount = 0;
+    }
+    void clear() {
+        for (int i=0; i<dcount; i++)
+            states[active[i]] = 0xFF;
+        dcount = 0;
+    }
+    bool push(int key) {
+        assert(key >= 0 && key <= 127);
+        if (states[key] != 0xFF) {
+            return true;
+        }
+        states[key] = dcount;
+        active[dcount++] = key;
+        return false;
+    }
+    bool pop(int key) {
+        if (states[key] == 0xFF)
+            return false;
+        int pos = states[key];
+        if (pos != dcount-1) {
+            // reuse the popped item's stack position for stack top
+            int last = active[dcount-1];
+            active[pos] = last;
+            // mark that position's new place on stack
+            states[last] = pos;
+        }
+        states[key] = 0xFF;
+        dcount--;
+        return true;
+    }    
+    inline bool has(int key) {
+        return states[key] != 0xFF;
+    }
+    inline int count() {
+        return dcount;
+    }
+    inline bool empty() {
+        return (dcount == 0);
+    }
+    inline int nth(int n) {
+        return active[n];
+    }
+};
+
+/**
+ * Convert MIDI note number to normalized UINT phase (where 1<<32 is full cycle).
+ * @param MIDI note number
+ * @param cents detune in cents (1/100 of a semitone)
+ * @param sr sample rate
+ */
+inline unsigned int midi_note_to_phase(int note, double cents, int sr) {
+    double incphase = 440*pow(2.0, (note-69)/12.0 + cents/1200.0)/sr;
+    if (incphase >= 1.0) incphase = fmod(incphase, 1.0);
+    incphase *= 65536.0*65536.0;
+    return (unsigned int)incphase;
+}
+
+// Base class for all voice objects
+class voice {
+public:
+    int sample_rate;
+    bool released, sostenuto, stolen;
+
+    voice() : sample_rate(-1), released(false), sostenuto(false), stolen(false) {}
+
+    /// reset voice to default state (used when a voice is to be reused)
+    virtual void setup(int sr) { sample_rate = sr; }
+    /// reset voice to default state (used when a voice is to be reused)
+    virtual void reset()=0;
+    /// a note was pressed
+    virtual void note_on(int note, int vel)=0;
+    /// a note was released
+    virtual void note_off(int vel)=0;
+    /// check if voice can be removed from active voice list
+    virtual bool get_active()=0;
+    /// render voice data to buffer
+    virtual void render_to(float (*buf)[2], int nsamples)=0;
+    /// very fast note off
+    virtual void steal()=0;
+    /// return the note used by this voice
+    virtual int get_current_note()=0;
+    virtual float get_priority() { return stolen ? 20000 : (released ? 1 : (sostenuto ? 200 : 100)); }
+    /// empty virtual destructor
+    virtual ~voice() {}
+};
+
+/// An "optimized" voice class using fixed-size processing units
+/// and fixed number of channels. The drawback is that voice
+/// control is not sample-accurate, and no modulation input
+/// is possible, but it should be good enough for most cases
+/// (like Calf Organ).
+template<class Base>
+class block_voice: public Base {
+public:
+    // derived from Base
+    // enum { Channels = 2 };
+    using Base::Channels;
+    // enum { BlockSize = 16 };
+    using Base::BlockSize;
+    // float output_buffer[BlockSize][Channels];
+    using Base::output_buffer;
+    // void render_block();
+    using Base::render_block;
+    unsigned int read_ptr;
+
+    block_voice()
+    {
+        read_ptr = BlockSize;
+    }
+    virtual void reset()
+    {
+        Base::reset();
+        read_ptr = BlockSize;
+    }
+    virtual void render_to(float (*buf)[2], int nsamples)
+    {
+        int p = 0;
+        while(p < nsamples)
+        {
+            if (read_ptr == BlockSize) 
+            {
+                render_block();
+                read_ptr = 0;
+            }
+            int ncopy = std::min<int>(BlockSize - read_ptr, nsamples - p);
+            for (int i = 0; i < ncopy; i++)
+                for (int c = 0; c < Channels; c++)
+                    buf[p + i][c] += output_buffer[read_ptr + i][c];
+            p += ncopy;
+            read_ptr += ncopy;
+        }
+    }
+};
+
+/// Base class for all kinds of polyphonic instruments, provides
+/// somewhat reasonable voice management, pedal support - and 
+/// little else. It's implemented as a base class with virtual
+/// functions, so there's some performance loss, but it shouldn't
+/// be horrible.
+/// @todo it would make sense to support all notes off controller too
+struct basic_synth {
+protected:
+    /// Current sample rate
+    int sample_rate;
+    /// Hold pedal state
+    bool hold;
+    /// Sostenuto pedal state
+    bool sostenuto;
+    /// Voices currently playing
+    std::list<dsp::voice *> active_voices;
+    /// Voices allocated, but not used
+    std::stack<dsp::voice *> unused_voices;
+    /// Gate values for all 128 MIDI notes
+    std::bitset<128> gate;
+    /// Maximum allocated number of channels
+    unsigned int polyphony_limit;
+
+    void kill_note(int note, int vel, bool just_one);
+public:
+    virtual void setup(int sr) {
+        sample_rate = sr;
+        hold = false;
+        sostenuto = false;
+        polyphony_limit = (unsigned)-1;
+    }
+    virtual void trim_voices();
+    virtual dsp::voice *give_voice();
+    virtual dsp::voice *alloc_voice()=0;
+    virtual dsp::voice *steal_voice();
+    virtual void render_to(float (*output)[2], int nsamples);
+    virtual void note_on(int note, int vel);
+    virtual void percussion_note_on(int note, int vel) {}
+    virtual void control_change(int ctl, int val);
+    virtual void note_off(int note, int vel);
+    /// amt = -8192 to 8191
+    virtual void pitch_bend(int amt) {}
+    virtual void on_pedal_release();
+    virtual bool check_percussion() { return active_voices.empty(); }
+    virtual ~basic_synth();
+};
+
+}
+
+#endif

plugins/ladspa_effect/calf/calf/utils.h

@@ -0,0 +1,161 @@
+/* Calf DSP Library
+ * Utilities
+ *
+ * Copyright (C) 2008 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA 02111-1307, USA.
+ */
+#ifndef __CALF_UTILS_H
+#define __CALF_UTILS_H
+
+#include <errno.h>
+#include <pthread.h>
+#include <map>
+#include <string>
+
+namespace calf_utils
+{
+
+/// Pthreads based mutex class
+class ptmutex
+{
+public:
+    pthread_mutex_t pm;
+    
+    ptmutex(int type = PTHREAD_MUTEX_RECURSIVE)
+    {
+        pthread_mutexattr_t attr;
+        pthread_mutexattr_init(&attr);
+        pthread_mutexattr_settype(&attr, type);
+        pthread_mutex_init(&pm, &attr);
+        pthread_mutexattr_destroy(&attr);
+    }
+    
+    bool lock()
+    {
+        return pthread_mutex_lock(&pm) == 0;
+    }
+    
+    bool trylock()
+    {
+        return pthread_mutex_trylock(&pm) == 0;
+    }
+    
+    void unlock()
+    {
+        pthread_mutex_unlock(&pm);
+    }
+
+    ~ptmutex()
+    {
+        pthread_mutex_destroy(&pm);
+    }
+};
+
+/// Exception-safe mutex lock
+class ptlock
+{
+    ptmutex &mutex;
+    bool locked;
+    
+public:
+    ptlock(ptmutex &_m) : mutex(_m), locked(true)
+    {
+        mutex.lock();
+    }
+    void unlock()
+    {
+        mutex.unlock();
+        locked = false;
+    }
+    void unlocked()
+    {
+        locked = false;
+    }
+    ~ptlock()
+    {
+        if (locked)
+            mutex.unlock();
+    }
+};
+
+/// Exception-safe temporary assignment
+template<class T>
+class scope_assign
+{
+    T &data, old_value;
+public:
+    scope_assign(T &_data, T new_value)
+    : data(_data), old_value(_data)
+    {
+        data = new_value;
+    }
+    ~scope_assign()
+    {
+        data = old_value;
+    }
+};
+
+struct text_exception: public std::exception
+{
+    const char *text;
+    std::string container;
+public:
+    text_exception(const std::string &t) : container(t) { text = container.c_str(); }
+    virtual const char *what() const throw () { return text; }
+    virtual ~text_exception() throw () {}
+};
+
+struct file_exception: public std::exception
+{
+    const char *text;
+    std::string message, filename, container;
+public:
+    file_exception(const std::string &f) : message(strerror(errno)), filename(f), container(filename + ":" + message) { text = container.c_str(); }
+    file_exception(const std::string &f, const std::string &t) : message(t), filename(f), container(filename + ":" + message) { text = container.c_str(); }
+    virtual const char *what() const throw () { return text; }
+    virtual ~file_exception() throw () {}
+};
+
+/// String-to-string mapping
+typedef std::map<std::string, std::string> dictionary;
+
+/// Serialize a dictonary to a string
+extern std::string encode_map(const dictionary &data);
+/// Deserialize a dictonary from a string
+extern void decode_map(dictionary &data, const std::string &src);
+
+/// int-to-string
+extern std::string i2s(int value);
+
+/// float-to-string
+extern std::string f2s(double value);
+
+/// float-to-string-that-doesn't-resemble-an-int
+extern std::string ff2s(double value);
+
+/// Escape a string to be used in XML file
+std::string xml_escape(const std::string &src);
+
+/// Load file from disk into a std::string blob, or throw file_exception
+std::string load_file(const std::string &src);
+
+/// Indent a string by another string (prefix each line)
+std::string indent(const std::string &src, const std::string &indent);
+
+};
+
+#endif

plugins/ladspa_effect/calf/giface.cpp

@@ -0,0 +1,325 @@
+/* Calf DSP Library
+ * Module wrapper methods.
+ *
+ * Copyright (C) 2001-2007 Krzysztof Foltman
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General
+ * Public License along with this program; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, 
+ * Boston, MA  02110-1301  USA
+ */
+#include <assert.h>
+#include <memory.h>
+#include <calf/giface.h>
+#include <stdio.h>
+using namespace std;
+using namespace calf_utils;
+using namespace calf_plugins;
+
+float parameter_properties::from_01(double value01) const
+{
+    double value = dsp::clip(value01, 0., 1.);
+    switch(flags & PF_SCALEMASK)
+    {
+    case PF_SCALE_DEFAULT:
+    case PF_SCALE_LINEAR:
+    case PF_SCALE_PERC:
+    default:
+        value = min + (max - min) * value01;
+        break;
+    case PF_SCALE_QUAD:
+        value = min + (max - min) * value01 * value01;
+        break;
+    case PF_SCALE_LOG:
+        value = min * pow(double(max / min), value01);
+        break;
+    case PF_SCALE_GAIN:
+        if (value01 < 0.00001)
+            value = min;
+        else {
+            float rmin = std::max(1.0f / 1024.0f, min);
+            value = rmin * pow(double(max / rmin), value01);
+        }
+        break;
+    case PF_SCALE_LOG_INF:
+        assert(step);
+        if (value01 > (step - 1.0) / step)
+            value = FAKE_INFINITY;
+        else
+            value = min * pow(double(max / min), value01 * step / (step - 1.0));
+        break;
+    }
+    switch(flags & PF_TYPEMASK)
+    {
+    case PF_INT:
+    case PF_BOOL:
+    case PF_ENUM:
+    case PF_ENUM_MULTI:
+        if (value > 0)
+            value = (int)(value + 0.5);
+        else
+            value = (int)(value - 0.5);
+        break;
+    }
+    return value;
+}
+
+double parameter_properties::to_01(float value) const
+{
+    switch(flags & PF_SCALEMASK)
+    {
+    case PF_SCALE_DEFAULT:
+    case PF_SCALE_LINEAR:
+    case PF_SCALE_PERC:
+    default:
+        return double(value - min) / (max - min);
+    case PF_SCALE_QUAD:
+        return sqrt(double(value - min) / (max - min));
+    case PF_SCALE_LOG:
+        value /= min;
+        return log((double)value) / log((double)max / min);
+    case PF_SCALE_LOG_INF:
+        if (IS_FAKE_INFINITY(value))
+            return max;
+        value /= min;
+        assert(step);
+        return (step - 1.0) * log((double)value) / (step * log((double)max / min));
+    case PF_SCALE_GAIN:
+        if (value < 1.0 / 1024.0) // new bottom limit - 60 dB
+            return 0;
+        double rmin = std::max(1.0f / 1024.0f, min);
+        value /= rmin;
+        return log((double)value) / log(max / rmin);
+    }
+}
+
+float parameter_properties::get_increment() const
+{
+    float increment = 0.01;
+    if (step > 1)
+        increment = 1.0 / (step - 1);
+    else 
+    if (step > 0 && step < 1)
+        increment = step;
+    else
+    if ((flags & PF_TYPEMASK) != PF_FLOAT)
+        increment = 1.0 / (max - min);
+    return increment;
+}
+
+int parameter_properties::get_char_count() const
+{
+    if ((flags & PF_SCALEMASK) == PF_SCALE_PERC)
+        return 6;
+    if ((flags & PF_SCALEMASK) == PF_SCALE_GAIN) {
+        char buf[256];
+        size_t len = 0;
+        sprintf(buf, "%0.0f dB", 6.0 * log(min) / log(2));
+        len = strlen(buf);
+        sprintf(buf, "%0.0f dB", 6.0 * log(max) / log(2));
+        len = std::max(len, strlen(buf)) + 2;
+        return (int)len;
+    }
+    return std::max(to_string(min).length(), std::max(to_string(max).length(), to_string(min + (max-min) * 0.987654).length()));
+}
+
+std::string parameter_properties::to_string(float value) const
+{
+    char buf[32];
+    if ((flags & PF_SCALEMASK) == PF_SCALE_PERC) {
+        sprintf(buf, "%0.f%%", 100.0 * value);
+        return string(buf);
+    }
+    if ((flags & PF_SCALEMASK) == PF_SCALE_GAIN) {
+        if (value < 1.0 / 1024.0) // new bottom limit - 60 dB
+            return "-inf dB"; // XXXKF change to utf-8 infinity
+        sprintf(buf, "%0.1f dB", 6.0 * log(value) / log(2));
+        return string(buf);
+    }
+    switch(flags & PF_TYPEMASK)
+    {
+    case PF_STRING:
+        return "N/A";
+    case PF_INT:
+    case PF_BOOL:
+    case PF_ENUM:
+    case PF_ENUM_MULTI:
+        value = (int)value;
+        break;
+    }
+
+    if ((flags & PF_SCALEMASK) == PF_SCALE_LOG_INF && IS_FAKE_INFINITY(value))
+        sprintf(buf, "+inf"); // XXXKF change to utf-8 infinity
+    else
+        sprintf(buf, "%g", value);
+    
+    switch(flags & PF_UNITMASK) {
+    case PF_UNIT_DB: return string(buf) + " dB";
+    case PF_UNIT_HZ: return string(buf) + " Hz";
+    case PF_UNIT_SEC: return string(buf) + " s";
+    case PF_UNIT_MSEC: return string(buf) + " ms";
+    case PF_UNIT_CENTS: return string(buf) + " ct";
+    case PF_UNIT_SEMITONES: return string(buf) + "#";
+    case PF_UNIT_BPM: return string(buf) + " bpm";
+    case PF_UNIT_RPM: return string(buf) + " rpm";
+    case PF_UNIT_DEG: return string(buf) + " deg";
+    case PF_UNIT_NOTE: 
+        {
+            static const char *notes = "C C#D D#E F F#G G#A A#B ";
+            int note = (int)value;
+            if (note < 0 || note > 127)
+                return "---";
+            return string(notes + 2 * (note % 12), 2) + i2s(note / 12 - 2);
+        }
+    }
+
+    return string(buf);
+}
+
+void calf_plugins::plugin_ctl_iface::clear_preset() {
+    int param_count = get_param_count();
+    for (int i=0; i < param_count; i++)
+    {
+        parameter_properties &pp = *get_param_props(i);
+        if ((pp.flags & PF_TYPEMASK) == PF_STRING)
+        {
+            configure(pp.short_name, pp.choices ? pp.choices[0] : "");
+        }
+        else
+            set_param_value(i, pp.def_value);
+    }
+}
+
+const char *calf_plugins::load_gui_xml(const std::string &plugin_id)
+{
+#if 0
+    try {
+        return strdup(calf_utils::load_file((std::string(PKGLIBDIR) + "/gui-" + plugin_id + ".xml").c_str()).c_str());
+    }
+    catch(file_exception e)
+#endif
+    {
+        return NULL;
+    }
+}
+
+bool calf_plugins::check_for_message_context_ports(parameter_properties *parameters, int count)
+{
+    for (int i = count - 1; i >= 0; i--)
+    {
+        if (parameters[i].flags & PF_PROP_MSGCONTEXT)
+            return true;
+    }
+    return false;
+}
+
+bool calf_plugins::check_for_string_ports(parameter_properties *parameters, int count)
+{
+    for (int i = count - 1; i >= 0; i--)
+    {
+        if ((parameters[i].flags & PF_TYPEMASK) == PF_STRING)
+            return true;
+        if ((parameters[i].flags & PF_TYPEMASK) < PF_STRING)
+            return false;
+    }
+    return false;
+}
+
+#if USE_DSSI
+struct osc_cairo_control: public cairo_iface
+{
+    osctl::osc_inline_typed_strstream &os;
+    
+    osc_cairo_control(osctl::osc_inline_typed_strstream &_os) : os(_os) {}
+    virtual void set_source_rgba(float r, float g, float b, float a = 1.f)
+    {
+        os << (uint32_t)LGI_SET_RGBA << r << g << b << a;
+    }
+    virtual void set_line_width(float width)
+    {
+        os << (uint32_t)LGI_SET_WIDTH << width;
+    }
+};
+
+static void send_graph_via_osc(osctl::osc_client &client, const std::string &address, line_graph_iface *graph, std::vector<int> &params)
+{
+    osctl::osc_inline_typed_strstream os;
+    osc_cairo_control cairoctl(os);
+    for (size_t i = 0; i < params.size(); i++)
+    {
+        int index = params[i];
+        os << (uint32_t)LGI_GRAPH;
+        os << (uint32_t)index;
+        for (int j = 0; ; j++)
+        {
+            float data[128];
+            if (graph->get_graph(index, j, data, 128, &cairoctl))
+            {
+                os << (uint32_t)LGI_SUBGRAPH;
+                os << (uint32_t)128;
+                for (int p = 0; p < 128; p++)
+                    os << data[p];
+            }
+            else
+                break;
+        }
+        for (int j = 0; ; j++)
+        {
+            float x, y;
+            int size = 3;
+            if (graph->get_dot(index, j, x, y, size, &cairoctl))
+                os << (uint32_t)LGI_DOT << x << y << (uint32_t)size;
+            else
+                break;
+        }
+        for (int j = 0; ; j++)
+        {
+            float pos = 0;
+            bool vertical = false;
+            string legend;
+            if (graph->get_gridline(index, j, pos, vertical, legend, &cairoctl))
+                os << (uint32_t)LGI_LEGEND << pos << (uint32_t)(vertical ? 1 : 0) << legend;
+            else
+                break;
+        }
+        os << (uint32_t)LGI_END_ITEM;
+    }
+    os << (uint32_t)LGI_END;
+    client.send(address, os);
+}
+
+calf_plugins::dssi_feedback_sender::dssi_feedback_sender(const char *URI, line_graph_iface *_graph, calf_plugins::parameter_properties *props, int num_params)
+{
+    graph = _graph;
+    client = new osctl::osc_client;
+    client->bind("0.0.0.0", 0);
+    client->set_url(URI);
+    for (int i = 0; i < num_params; i++)
+    {
+        if (props[i].flags & PF_PROP_GRAPH)
+            indices.push_back(i);
+    }
+}
+
+void calf_plugins::dssi_feedback_sender::update()
+{
+    send_graph_via_osc(*client, "/lineGraph", graph, indices);
+}
+
+calf_plugins::dssi_feedback_sender::~dssi_feedback_sender()
+{
+    // this would not be received by GUI's main loop because it's already been terminated
+    // client->send("/iQuit");
+    delete client;
+}
+#endif