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Abstracts of Talks/Presentations
(in alphabetical order)
AEOLUS - a church organ in your PC
Fons Adriaensen (Antwerp, Belgium)
Many organist will see any attempt to imitate the sound of a real pipe
organ
by electronic means as pure heresy. They are not entirely wrong, as
there
are many examples of 'electronic organs', some of them quite expensive
and
surrounded by much hype, that fail miserably to even capture the most
essential sonic qualities of a real instrument. There are many reasons
for this.
While the bare sound of most organ pipes is not really complicated and
can
be imititated by for example additive synthesis, this is only the
beginning.
Every real pipe organ is designed and voiced for the environment it is
to be
used in. 'Voicing' an organ stop, wich means tuning the loudness,
timbre,
and maybe other parameters of each individual pipe to arrive at a set
that
is balanced within itself and combines well with the other stops, is
the
art of a skilled craftsman.
Another important point is the acoustic environment. A real organ
placed
in an anechoic, 'dead' room, doesn't sound much more inspiring than a
square wave. So it is essential to provide a high quality reverb,
including
realistic early reflections.
This talk will introduce the first release of AEOLUS, a GPL-ed pipe
organ
program for Linux that should be able to provide a passable imitation
of a
small to medium sized instrument that a musician would actually enjoy
playing.
AEOLUS provides the end user with the necessary controls and parameters
to
do a proper voicing of his instrument and will also include a high
quality 3-D reverb and Ambisonics B-format outputs. If the technical
provisions at ZKM allow, the program will be demonstrated in full
surround-sound mode.
Using JAAA for audio measurements
Fons Adriaensen (Antwerp, Belgium)
JAAA (the Jack and Alsa Audio Analyser) is a combined signal generator
and spectrum analyser designed for precision audio measurements. First
beta release is planned for January 2004.
This talk will present a short introduction to the theory of spectral
analysis, covering topics such as bandwidth, windowing and noise
measurement. In a second part it will be demonstrated how to use the
program to verify the performance and quality of your audio card or
any other piece of audio equipment.
Application of Wave Field Synthesis in electronic music and sound installations
Marije Baalman (Technical University Berlin, Germany)
Wave Field Synthesis offers new possibilities for composers of electronic
music and to sound artists to add the dimension of space to a
composition. Unlike most other spatialisation techniques,Wave Field
Synthesis is suitable for concert situations, where the listening area
needs to be large. Using the software program "WONDER", developed at the TU
Berlin, compositions can be made or setups can be created for realtime
control from other programs, using the Open Sound Control protocol.
During the talk, the principles of wave field synthesis are discussed and
how to set up a system. Then the features of WONDER are discussed, as well
as some pieces that were created using the software.
Rapid and Reuseable Audio Development With Pd
Frank Barknecht (Cologne, Germany)
Modular synthesizers traditionally use the concept of rather low level
sound modules ("unit generators") from which more complex instruments
are built. But programming with low level generators is a difficult
task and requires not only a lot of experience and knowledge of
synthesis techniques but also a lot of time.
The graphical sound environment Pure Data (Pd) by MAX/MSP-inventor
Miller S. Puckette allows developers to create reusable high-level
instruments with integrated sound engine and user interface. These
patches then allow a rapid development cycle and also make Pd itself
more accessible to less
experienced or beginning users.
The talk will present several examples of such RRAD-tools and how they
can be used to build for example trendy techno toys remotely similar to
commercial software like Propellerhead's Reason. The talk will include
a basic beginner's introduction to Pd and explore if RRAD-patches could
accelerate the otherwise steep learning curve of Pd.
Hurdles and Benefits of Introducing Linux as a Viable
Digital Audio Workstation in the Academic Environment
Ivica Ico Bukvic
(Oberlin College and
College-Conservatory of Music, Center for Computer Music, University of
Cincinnati,
Ohio, USA)
As a composer focusing on computer music pursuing career in Academia and an
avid Linux fan I've made a considerable effort to introduce Linux as a viable
digital audio workstation (DAW) alternative to the established mainstream
OS's, namely MacOS and Win32 platforms. In my pursuit, I've encountered a
number of hurdles as well as realized some very important benefits of such
efforts that reach much farther than the point of "evangelizing" the general
computer-dependent populace. It is therefore my interest to present my findings
to the fellow Linux audio enthusiasts who may have come in contact with similar
circumstances and/or will have opportunity to do so in the near future.
Overview
It is my intention to present the given topic using slides with a detailed
breakdown of the known issues in a form of a lecture that will ultimately
encourage discussion among the audience members in order to generate possible
suggestions as to how to surmount some of the hurdles, as well as emphasize its
obvious advantages. I will present my first-hand findings while lobbying for
Linux adoption at two reputable Colleges in US and the surprising responses
I've run across, namely College-Conservatory of Music at the University of
Cincinnati and the Oberlin College. Finally, I will also point out interesting
changes in focus that were a direct result of the unrealistic expectations I
established among the faculty who expressed interest in adopting the Linux
platform. Finally, I will generate a detailed overview explaining why this
course of action is an extremely important catalyst for the adoption of Linux
as a viable DAW.
Goals
By presenting my experiences and furthermore instigating a discussion I expect
to generate a fruitful discourse that will lead towards generation of a set of
desirable strategies for promotion of Linux in the academic audio-related
circles. Such strategies will not only help make future endeavors of this sort
a more fruitful and productive experience, but will also help avoid current
most obvious and detrimental caveats of such action.
Unlocking the Full Potential of RTMix real-time interactive multimedia Art
Performance, Composition, and Coaching Interface
Ivica Ico Bukvic
(Oberlin College and
College-Conservatory of Music, Center for Computer Music, University of
Cincinnati,
Ohio, USA)
RTMix is a result of my personal compositional endeavors through which I
realized an apparent lack of a unifying front-end software interface geared
towards live performance of interactive electroacoustic art in a traditional
concert setting, especially in respect towards interactive music that utilizes
one or more of the Music-N languages for a real-time manipulation of an audio
signal, Csound, Supercollider, and RTcmix being one of the more popular
choices. Already two years in development, RTMix has grown well beyond its
initial and relatively humble goals and has become a focal interface furnishing
MIDI and OSC capabilities, networkability, and an elaborate scripting language
that incorporates probability-related functions, as well as advanced timing
mechanisms. With its growing popularity and already thousands of downloads,
there is an increasing need to expose its ever-changing and improving set of
capabilities in a form of a workshop that would help users to unlock its full
potential.
Overview
The proposed demo session will be presented in a form of a workshop that would
address a number of advanced features through series of brief tutorials, as
well as emphasize advantages of its utilization over other available interfaces
that may furnish a similar functionality. In addition, it is my intention to
focus specifically on addressing the interconnectivity of RTMix with other
popular software packages geared towards live and interactive performance, such
as pure-data, RTcmix, and Supercollider. Finally, I intend to elicit a response
from the audience in a form of comments and feature requests. Such data will be
then utilized to generate a roadmap of RTMix's further development, giving
priority to the most sought functionalities. Through interaction with the
audience members I will also look into locating potential co-developers that
will help me expand the RTMix as well as quicken the pace of its development.
Goals
The goal of this demonstration is to help the existing RTMix's users utilize
the software in a more efficient fashion, conceivably expand the user-base, as
well as gather invaluable data that will suggest the roadmap for its further
development. Finally, it is my hope that through this exposure I will attract
additional developers that will assist me in adding the newly-sought features
to this exciting software package.
Recombinant Spatialization for Ecoacoustic Immersive Environments
Matthew Burtner and
David Topper
(The Virginia Center for Computer Music (VCCM), University of Virginia, USA)
An approach to digital audio synthesis is implemented using recombinant
spatialization for signal processing. This technique, which we call
Spatio-Operational Spectral Synthesis (SOS), relies on recent theories of
auditory perception, especially research in auditory perception by Kubovy and
Bregman. Here, the perceptual spatial phenomenon of objecthood is explored as
an expressive musical tool. In musical applications of these theories, we
observe the emergence of a "persistence of audition" exposing interesting
opportunities for compositional development. In essence, SOS, breaks an audio
signal into salient components then recombines and spatializes them in a
multichannel environment. The presentation will detail some of our recent work
using SOS in an 8 channel environment.
Following an introduction to the technique and several examples demonstrating
potential applications, this paper concentrates on some applications of the
technique in ecoacoustic compositions by Matthew Burtner, Anugi Unipkaaq,
Sikniq Unipkaaq and Siku Unipaaq. These works draw on environmental systems as
models for multichannel processing.
S.O.S has been implemented in RTCMIX on Linux.
Linux console - a textbased studio
Julien Patrick Claassen (SBS C-LAB,
Paderborn)
Programs (probably) used:
mplayer, timidity, fluidsynth, ecasound and ladspa-FX
Aim: What can I do with a linux console and audio?
Rough overview:
- Introduction to the topic
- My way of working (short explanation of a blind-person's work with a braille-display)
- Short demonstrations: playing a .wav-file, a video, a midi-file, an
mp3 and an ogg.
- a small recording of a fluidsynth-sound via a jack-connection
- FX-processing of audio-material with ecasound and ladspa
- Mixing a few tracks
- Last words
Idea: the idea of the talk is to _PRACTICALLY_ demonstrate the abilities
of a linux console in the field of audio. My play is not only to give an
idea of the possibilities and leaving it at that, but to present and
perhaps create some "realworld" material with the text-based tools I
use.
Note: There won't be any slides, but some html-documents and scripts,
which I use during my presentation, so you can try it yourself later.
Adding VST support to Linux audio applications
Paul Davis (Linux Audio Systems, Bala
Cynwyd,
Pennsylvania, USA)
For users of proprietary audio applications, plugins have come to play
an increasingly important role, sometimes generating more allegiance
than host applications themselves. Supporting existing plugins for
Windows and MacOS is therefore an important part of convincing
potential users to try Linux as a platform for audio work. I will
outline previous attempts to support VST plugins under Linux, and will
then continue to discuss in detail a new approach worked on by myself
and Torben Hohn (gAlan) that is available to most existing Linux audio
applications. Demonstrations of several free win32/x86 VST plugins
will be included, both as standalone JACK applications and within the
Linux DAW Ardour.
Ardour - The Digital Audio Workstation for Linux
Paul Davis (Linux Audio Systems, Bala
Cynwyd,
Pennsylvania, USA)
Ardour is a digital audio workstation, providing comprehensive
multitrack, multichannel non-linear editing facilities for high
resolution audio projects. This talk will briefly outline Ardour's
design and history, and will then focus on a demonstration of the
program's current capabilities including recording, editing, mixing,
processing, exporting and more.
Audio networking
François Déchelle,
Patrice Tisserand and
Simon Schampijer
(IRCAM, Paris, France)
We will talk about distributed architectures for audio and music, focusing on
real time audio streaming over Internet and grid computing.
Distant musical interaction via Internet has been tackled by numerous project.
The 'Distributed virtual concert' project, which aims to provide musicians
(electronic or acoustic) a way to play together via Internet, is a
collaboration between IRCAM and CEDRIC, the computer science research lab of
CNAM/Paris, and is currently implemented as a jMax package. The musicians are
connected by audio streams other RTP and a distributed algorithm resynchronize
the different streams to a common tempo. The different issues associated with
real time audio streaming on Internet will be presented, together with the free
softwares available.
We will also present some experimentations done at IRCAM on grid computing.
Grid computing covers a number of technologies that aim to provide to end users
standardized access to processing power and storage using high speed networks.
We'll make an overview of current grid computing technologies and describe the
free software that are available today on GNU/Linux to build grids and to
distribute computations other a network of desktop machines. We will then
discuss how grids could be used for musical and audio applications and what
they can bring to musicians and audio engineers.
Linux as a Workstation for Composers
Orm Finnendahl (Folkwang-Hochschule Essen, Germany)
Using the computer for contemporary composition means working with
lots of different tools: graphics applications, programming languages,
sound synthesis languages, tools for score generation etc. The
GNU/ Linux operating system not only supports a wide range of
specialized applications for these purposes but is also an ideal
platform for the integration and customization of the different parts
of the working process. The presentation gives some practical examples
including less common usages of these tools.
PD Workshop
Orm Finnendahl (Folkwang-Hochschule Essen, Germany)
PD has built in graphic capabilities not well known to most users and
not existing in plain max/ msp. These graphic capabilities have a big
potential for user interface design. In the workshop some examples are
given including a performance demonstration with musicians of the
opening concert, Sascha Armbruster and Burkhard Beins.
AGNULA: the past, the present, the future
Andrea Glorioso (Centro Tempo Reale, Florence, Italy)
AGNULA was born in 2002 as the first totally Libre Software project
funded by the European Commission, under the 5th Framework
Programme. Initially focused on building two reference GNU/Linux
distributions for audio & video (DeMuDi and RehMuDi) the project has
slowly extended its goal towards educating users on the topics of
Libre Software, with specific attention paid to audio/video
applications and content distribution.
The funded lifetime of AGNULA will be finished by the time of this
talk - but AGNULA intends to be alive and kicking.
In this talk Andrea Glorioso, former AGNULA technical manager, will
provide a brief history of the project, a bird's eye on the current
situation, and the plans for the future - near, medium and long
term (i.e. world domination).
flext - C++ layer for cross-platform development of Max/MSP and pd externals
Thomas Grill (Vienna, Austria)
flext seeks to represent a uniform programming interface for extending the
most common
modular real-time audio systems Max/MSP and Pure Data (PD) with external
modules, or
short externals. These modules provide a way to tailor such a system for one
's special needs
and supply additional functionality. Source code based on flext is able to
exploit nearly all
features of the respective real-time framework while staying completely
independent of the
actual host system and platform (hardware and operating system). flext
currently supports PD
for Linux, Windows and OSX as well as Max/MSP for OS9 and OSX (and shortly
Windows).
Support for jMax under Linux, OSX and Windows and other systems can follow
in the near
future.
LASH
Bob Ham (Nottingham, UK)
LASH is a session manager for linux audio applications that
automatically saves and restores port connections. This talk will
discuss the motivation and genesis of LASH; the operation of the
system and the roles played by different software; and will look at
future areas of development.
JACK, JAMin, Mastering
Steve Harris (IAM - Intelligence, Agents and Multimedia Research Group.
University of Southampton, Hampshire, UK)
This talk will cover the operation and intended usage of the JAMin mastering
tool and explain some of the design decisions. The first part will explain
how to operate the software in conjunction with other JACK-based tools and
the second will be cover mastering techniques with demonstrations.
Contents
- Overview of JAMin UI
- Signal flow description + functionality
- Integration with JACK
- Delayed mix-down philosophy
- Usage hints and tricks
- Demo
Audio Engineering in a Nutshell
Steve Harris (IAM - Intelligence, Agents and
Multimedia Research Group. University of Southampton, Hampshire, UK) and
Jörn Nettingsmeier
(Folkwang-Hochschule Essen, Germany)
"Audio Engineering in a Nutshell" is meant to be an in-depth crash
course all the way from the fundamentals (sound waves, sampling) to some
of the more tricky aspects of sound manipulation. Along the way, you
will get an idea how digital signal processing actually works.
Specifically, we will cover control of audio spectrum (by equalization),
dynamics (by compression), timing (by delays) and space and
spatialization (by reverb, stereo miking and panning techniques).
after a brief recap about sound signals and their digital
representation, we will explore each of these topics in detail.
We begin each section with some general "philosophy" on the topic. Next,
we'll examine the workings of "traditional" analog audio devices to
understand the basic principles, and then move on to their digital
implementation, along with demonstrations of Linux effects plugins. At
the end of each section, we offer some practical usage tips, and will
try to answer any questions that might arise.
About the lecturers:
Steve Harris is the author and maintainer of swh-plugins, the most
comprehensive set of audio plugins under linux. Apart from numerous
contributions to other linux audio projects, he is one of the main
developers of jamin, the jack audio mastering interface. Steve works as
a researcher in the "intelligence, agents, multimedia group" at the
university of southampton.
Jörn Nettingsmeier is studying music at the Folkwang-Hochschule Essen
and computer science at the University of Duisburg-Essen, works as sound
engineer for theatre and live music, linux system administrator and
teacher of music theory, and he plays the odd gig as a jazz bassist.
ALSA Sucks? - Trouble Shooting for Your Healthy Music Life
Takashi Iwai (SuSE Linux AG, Nuremberg, Germany)
ALSA provides superior functionality for highend audio systems.
At the same time, however, it involves matters for many end users.
As ALSA is regarded as the new standard sound system on linux, it's
time to consider about this theme.
In this talk, a kind of FAQ regarding ALSA is presented:
what and how you can do with it, how you can debug, and
how you can communicate with ALSA developers.
SuperCollider3 on Linux - A Status Report
Stefan Kersten (Technical University Berlin, Germany)
SuperCollider3 (SC3) is James McCartney's realtime synthesis
and composition framework, based on a client-server
model.
An efficient, portable and embeddable server application
provides sound synthesis capabilities, and can be controlled
asynchronously by any number of OpenSoundControl (OSC)
client applications.
The object-oriented SuperCollider language is based on a
virtual machine with realtime execution semantics. Apart
from providing a rich stream-based environment for musical
composition, it tightly integrates with SuperCollider
synthesis servers running in the same address space, on the
same machine or distributed in a local network.
This talk gives an overview of SC3's architecture and its
current status on Linux. It also provides an outlook on a
possible integrated editing, control and visualization
framework for performance and composition.
ALSA project - the past year
Jaroslav Kysela (SuSE Linux AG, Ceske Budejovice, Czech Republic)
The ALSA project reached version 1.0. The focus in talk will be
given to the last year events in this project but also newcoming
features will be mentioned.
Developing Spectral Signal Processing Applications
Victor Lazzarini (Music Technology Laboratory,
Dept. Of Music,
National University of Ireland, Maynooth)
This talk will focus on aspects of application development
for spectral signal processing using C++ and the Sound
Object Library. It will start by introducing the basic concepts:
the FFT/STFT, convolution, the Phase Vocoder, instantaneous
frequency distribution, sinuoidal analysis and resynthesis.
It will then discuss the SndObj classes designed for spectral
processing and their applications. Using examples, the
development of spectral processing programs will be explored.
The talk will also discuss the use of the library as a framework
for audio application development, in conjunction with
C++ GUI frameworks, such as V.
Jack port and integration on Mac OSX
Stephane Letz (Grame, Lyon, France)
JackOSX is an implementation of Jack on MacOSX that fully integrate the Jack
server inside the CoreAudio architecture. It provides interoperability between
native Jack clients developed using the Jack API and standard MacOSX audio
applications that use the CoreAudio API. Any CoreAudio application like
iTunes or Logic become "jackified" to take profit of Jack audio communication
features.
The talk will present :
- Jack server adpatation and optimisation on MacOSX, particularly the
client activation code.
- Integration of Jack inside the CoreAudio architecture built using the
following components :
- the Jack Router : a CoreAudio user space driver that allows any
CoreAudio application to become a jack client and take profit of Jack audio
communication features.
- the Jack aware audio plug-ins provided to expand the audio routing
capabilities
- JackPilot : allows to control the Jack server and provide a
connection manager.
- Jack Transport API and synchronisation integration with CoreMidi
architecture.
Translating
Lots of
Insipid
Silly
Parenthesis into noise: Sound
Synthesis Techniques and Algorithmic Composition
Fernando Pablo Lopez-Lezcano (CCRMA/Stanford
University, California, USA)
Before rodents ran wild and permanently trained users to track a tiny
cursor in a screen, mere words were the interface and old fashioned
mechanical keys were the input device. From that time in history come
text based sound generation and processing languages like CLM...
I'll briefly introduce sound synthesis and processing concepts and
theory, and algorithmic composition techniques using Common Lisp based
non-realtime tools (Common Music [CM] / Common Lisp Music [CLM]).
Additive / Subtractive / Modulation synthesis, Granular synthesis,
Digital Filtering, Automata, Fractals, Chaos and much more (especially
chaos :-)
How to create a Client for SuperCollider3 to produce Sound in
Realtime
Christian Mühlethaler and
Alexander Schuppisser (Bern, Switzerland)
SuperCollider3 (SC3) is James McCartney's realtime synthesis and
composition framework, based on a client-server model. For more
Information, see the abstract to
"SuperCollider3 on Linux - A Status
Report".
SC3 can be controlled by a network protocol called OpenSoundControl
(OSC) over TCP/IP or UDP.
This talk gives an overview about Sonificator, an open-source
client-framework for SC3 in Java. The purpose of
the framework was to sonificate data in realtime and to position the
sound sources in a virtual plane, so, that a listener with headphones
would actually feel surrounded by these sounds.
As an example application of this framework, a GUI will be presented,
based on the Java network Packet Capture Library (JPCAP) to sonificate
network traffic in this virtual landscape.
The framework is the product of a diploma at the University of Applied
Science of the canton of Berne, Switzerland
AlsaModularSynth - An instrument for the electrified virtuoso
Matthias Nagorni (SUSE Linux AG,
Nürnberg,
Germany)
The modular synthesizer AlsaModularSynth has been designed as an instrument
for live performance. Therefore all sound parameters can be controlled by
MIDI controllers. In addition to that, the parameters that define the sound
of a specific patch can be arranged into a separate dialog named "Parameter
View". These parameters can be saved as presets and are selectable via MIDI
program change. This way, AlsaModularSynth offers the full flexibility of
a modular synthesizer but allows also to build up convenient interfaces
for playing it on stage.
The talk will focus on the possibilities of getting a large variety of
sounds by using the MIDI velocity parameter to control sound variations
within a patch. After explaining basic patches, it will be shown how they
can be extended to obtain an expressive synthesizer,
Music Notation with LilyPond
Han-Wen Nienhuys (Utrecht, The Netherlands)
LilyPond is a modular, extensible and
programmable compiler for producing high-quality music notation. In
this talk we shall discuss the background of automated music printing,
describe how our system works. Finally, we will demonstrate the latest
and greatest new features.
3D game engines for sound synthesis
Julian Oliver (Selectparks and The Interactive Institute of Sweden)
I will survey several 3D engines available to the Linux
Desktop, and how they may be used as alternative interfaces both in
performance
and the generation of unique compositions.
Particular interest will be placed on 3D Game Engines, whose flexible
event structures can be used as a dynamic framework for composition.
Here, 'playing music' is given a fresh context.
Topics covered include Virtual Sensors, Event Threading, Switch
Networks, Navigation as Gesture, Virtual Acoustics, Non-Proprietary
Toolkits and Collaborative Performance in Virtual Environments.
Presented at the talk will be several of the author's own works, one of which
(in collaboration with Stephen Pickles) interfaces a popular game-engine
with the signal processing toolkit, Pure Data.
The workshop component will be based around this particular piece.
Participants of the workshop will develop a small sound-server that
receives control-data from AI Agents and players on a live game server.
Secondly they will develop a small 3D interface in a graphical
toolkit that they can then use in conjunction with their sound-server.
No formal programming experience is required.
The Faust audio programming language
Yann Orlarey (Grame, Lyon, France)
This talk will present Faust, a functional programming language for realtime
audio applications and plugins. The Faust compiler translates signal
processing specifications, written in a textual block diagram syntax, into
optimized C++ code.
The main features of the system are :
- A powerful high level language specifically designed for realtime
audio applications and plugin.
- An algebraic block diagram approach with a well defined formal
semantic.
- A semantic driven compiler able to translate Faust programs into
efficient C++ code that can compete with (and some time exceed) hand written
programs.
- The support of a large variety of plugin formats and standalone
applications from a single Faust specification.
Keynote
One Year After: Notable Linux Audio Development in 2003
Dave Phillips (Detroit, USA)
A brief summary of the most significant events in Linux
audio software development
since the first LAD meeting, followed by a brief
introduction to LAD 2004.
Still Nailing Jelly: An Update on Linux Audio Documentation Projects
Dave Phillips (Detroit, Michigan, USA)
A look at documentation progress in AGNULA, The Book Of Linux Music &
Sound ed.2,
Planet CCRMA,
Csound, and other projects. Articles about Linux audio
software have appeared in the
Linux Journal, Sound On Sound, Electronic Musician,
Keyboards, and the Computer
Music Journal. I'll discuss the increase of hard-copy and
on-line documentation and
I'll assess some projects currently in need of documentation.
Once again text & parentheses: sound synthesis with 'foo'
Martin Rumori (Technical University Berlin, Germany)
'foo' is a sound synthesis tool based on
the Scheme language, a clean and powerful Lisp dialect. 'foo' is used
for high-quality non-realtime sound synthesis and -processing. By
scripting 'foo' like a shell it is also a neat tool for implementing
common tasks like soundfile conversion, resampling, multichannel
extraction etc.
'foo' was developed by Gerhard Eckel and Ramón González-Arroyo
in 1993
at ZKM. Its kernel is written in Objective-C for the NeXTStep
platform. Eleven years later, a first port to Linux has been done
using the GNUstep framework. Porting to Cocoa/Mac OS X is also
planned.
'foo' does not distinguish between score and patch nor between audio
and control rate and has a simple and powerful temporal semantics. A
control library written in Scheme allows to access these features in
an abstract musical way.
In the talk, I will demonstrate some key features of 'foo' and give an
overview about porting and further development issues.
LinuxSampler
Benno Senoner (Lionstracs, Merano, Italy)
LinuxSampler is a work in progress and aims to become a professional grade
software sampler comparable to the most advanced windows/mac counterparts.
The main characteristics are that it supports disk streaming of sample
libraries
including the libraries in .GIG format thus allowing the playback of high
quality instruments.
The talk will describe the aspects of the project:
- brief history
- goals, long/short term
- current status
- interoperability with other OSes (VST, native Mac version)
- sample formats (libgig, akai etc)
- collaboration with companies (like sample library developers, hardware
manufacturers like Lionstracs and others).
- pratical demonstration on a Laptop
GAIA - Graphical Audio Interface Application
David Topper (The Virginia Center for Computer Music (VCCM), University of Virginia, USA)
Paper (pdf)
GAIA (Graphical Audio Interface Application) is an open source interface
for controlling the RTcmix synthesis and effects processing engine
running in a Linux environment.. Until recently, most RTcmix research
has been limited to using text-based scorefiles. The primary motivation
behind GAIA is to build upon this paradigm by providing a graphical
front end. An emphasis has been placed on creating an environment that
is easy to learn, robust and open source to allow for third party
contribution. GAIA breaks new ground in that it supports both graphical
and text based programming in the same application. Objects (or nodes
within the program's graph-like control structure) can themselves be
small scripts, written in Perl. These scripts can operate on data
within the application as well as trigger RTcmix events in real time.
GAIA unites two powerful open source projects, RTcmix and Perl, and
provides a powerful high level GUI for working with both. Control
mechanisms include the GAIA environment itself, MIDI, serial port data,
TCP/UDP socket data, and real time video processing via the Video 4
Linux (v4l) API. Through these mechanisms, GAIA creates a flexible and
powerful environment for controlling any number of synthesis and effects
processing parameters, bringing various techniques to a new level of
realization.
^
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