Sound Synthesis

We describe techniques to sonify rhythmic activity of epileptic seizures as measured by human EEG. Event- based mapping of parameters is found to be informative in terms of auto- and cross-correlations of the multivariate data. For the study, a group of patients with childhood absence seizures are selected. We find consistent intra-patient conservation of the rhythmic pattern as well as

A sound synthesis algorithm which simulates and extends the behaviour of the acoustic prepared piano is presented. The algorithm is based on a finite difference approximation to multiple stiff string vibration, including an excitation method (a hammer) as well as several connected preparation elements, modeled as lumped nonlinearities. Numerical issues and implementation details are discussed, and sound examples are presented..

In this paper we report on the synthesis of sounds using cellular
automata, specifically the multitype voter model. The mapping process adopted is based on digital signal processing analysis of automata evolutions and consists in mapping histograms onto spectrograms. The main problem of cellular automata is the difficulty of control and, consequently, sound synthesis
methods based on these computational models normally present a high factor of randomness in the output. We have achieved a significant degree of control as to predict the type of sounds that we can obtain. We are able to develop a flexible sound design process with emphasis on the possibility of controlling over time the spectrum complexity.

________ SynthBuilder is a user-extensible,object-oriented , NEXTSTEP Music Kit application for interactive real-time design and performance of synthesizer patches, especially physical models [Smith 1987]. Patches are represented by networks consisting of digital signal processing elements called unit generators and MIDI event elements called note filters and note generators. As an example of the kinds of patches developed, a six-string

An eight-channel spatial sound system was developed for a Java3D environment known as the Wedge. The goal of the system was to add immersiveness to the VR experience, as well as allow 3D sound scenes to be created through direct manipulation. The system used a layered architecture, with development occurring in four incremental stages. The first layer consisted of a PortAudio interface built on top of ASIO to send data to the sound card. The second layer established an eight- channel VBAP implementation in C++. The final two layers created a Java3D interface into the Wedge by extending an existing Java3D sound class, and allowing it to interface with the VBAP code through the Java Native interface. The advantage brought about by the integration of the audio system within the Java3D environment was that both graphics and audio could now be rendered within the same scene graph. This approach seeked to eliminate concerns such as learning other languages and protocols, or attempting to maintain synchronization of separate audio and graphics systems. The integrated system was therefore much more robust, and allowed for greater programmability.

Sound synthesis based on physical models of musical instruments is, ultimately, an exercise in numerical simulation. As such, for complex systems of the type seen in musical acoustics, simulation can be a computationally costly
undertaking, particularly if simplifying hypotheses, such
as those of traveling wave or mode decompositions are not
employed. In this paper, large scale time stepping methods, such as the finite difference time domain and finite
volume time domain methods are explored for a variety of
systems of interest in musical acoustics, including brass instruments, percussion instruments based on thin plate and
shell vibration, and also their embeddings in 3D acoustic
spaces. Attention is paid here to implementation issues,
particularly on parallel hardware, which is well-suited to
time stepping methods operating over regular grids. Sound
examples are presented.

This article discusses Phase Distortion synthesis and its application to arbitrary input signals. The main elements that compose the technique are presented. Its similarities to Phase Modulation are discussed and the equivalence between the two techniques is explored. Two alternative methods of distorting the phase of an arbitrary signal are presented. The first is based on the audio-rate modulation of a first-order allpass filter coefficient. The other method relies on a re-casting of the Phase Modulation equation, which leads to a heterodyned form of waveshaping. The relationship of these implementations to the original technique is explored in detail. Complementing the article, a number of examples are discussed, demonstrating the application of the technique as an interesting digital audio effect.

Η έκρηξη της τεχνολογίας του 20ου αιώνα ιδιαίτερα στο δεύτερο μισό (IBM Computers), βοήθησε αρχικά τα Πανεπιστήμια & Ερευνητικά Κέντρα να μελετήσουν ένα μεγάλο εύρος από επιστημονικούς τομείς. Άνθρωποι όπως οι Max Mathews και John Chowning, με βασικό εργαλείο τις γλώσσες προγραμματισμού της εποχής (π.χ. MUSIC-V) μελέτησαν και ανέπτυξαν θεωρίες οι οποίες στη συνέχεια πέρασαν στο ευρύ κοινό μέσω πληθώρας από Synthesizers (Yamaha DX7, PPG Wave κ.α.). Όλη αυτή η εξέλιξη μας οδήγησε στο σήμερα και στον συνδυασμό της Hardware και της Inside the Box φιλοσοφίας (DAW, plug-ins, VST-i), με πολλά από τα VST-i να εμπνέονται αναπόφευκτα από τα Hardware Synthesizers. H παρούσα εργασία αναλαμβάνει να εξετάσει τέσσερις από τις βασικές τεχνικές σύνθεσης ήχου (Additive, Subtractive, FM, Wavetable) και να βοηθήσει τον χρήστη να τις εφαρμόσει στα Synthesizers. Σκοπός μου δεν είναι να αναπτύξω ένα “ultimate” Synthesizer, αλλά μέσω των απλών Synthesizers (Max Patches) που ανέπτυξα να κατανοήσω καλύτερα τη συμπεριφορά των τεχνικών σύνθεσης ήχου αλλά και τον τρόπο λειτουργίας των Synthesizers.

Apresentamos neste trabalho uma introdução às definições de onda sonora, vibração e estudo do movimento harmônico tal como definições musicais de melodia e harmonia. Descrevemos o processo histórico da divisão numérica do intervalo musical e apresentamos os questionamentos sobre sua aceitação devido à limitação do ouvido humano em perceber tons diferentes. Utilizando uma visão analítica sobre esses aspectos e considerando a evolução musical, mostramos que a gama musical a qual possui o maior número de frações de denominadores baixos é ideal para exprimir a música que a percepção humana, em geral, é capaz de discernir. Repetimos esta característica para investigar e comparar a percepção musical para duas espécies de pássaros e relacioná-las à percepção musical humana.

In the late 1960s, composer, architect and theoretician Iannis Xenakis (1922-2001) began his research on stochastic synthesis: an approach to microsound synthesis that uses probability distributions to manipulate individual digital samples. He continued with this research for most of the 1970s and from the late 1980s until the end of his career. This article traces the aesthetic origins of this non-standard synthesis technique and gives a historical presentation of its development, including a discussion of the works in which Xenakis used it and a full description of the dynamic stochastic synthesis algorithms. In addition, a new extension of these techniques is put forward: the stochastic concatenation of dynamic stochastic synthesis.

We present ScalaCollider, a new client framework to connect to the SuperCollider sound synthesis server. It builds on top of the general purpose language Scala. Scala’s ambition is to allow for the development of scalable systems, being equally comfortable both for small-scale scripting and large-scale modular projects.
Following an overview and comparison of the currently available clients for SuperCollider, we introduce the most important features of Scala, and show how its specific language elements can be exploited to design an elegant client that supports UGen graph composition, handles proxy objects, and restores part of the clarity lost in the entanglement of the original Super- Collider client’s class library.
The problems of type-safety and approaches to concurrency are discussed, and an outlook on a high-level extension for declarative sound process specification is given.

In the 1960s and 1970s, most electrical (or analog) synthesizers were based on a sound synthesis technique called subtractive synthesis, in which a spectrally rich source signal, typically a sawtooth or rectangular pulse wave, is filtered with a time-varying filter. In the 1980s and 1990s this synthesis technique was almost overtaken by other synthesis techniques but in the mid-1990s musicians started again to show interest in the ''warm`` sound of subtractive synthesis. To meet this interest, a Swedish company Clavia introduced in 1995 the NordLead synthesizer that used the subtractive sound synthesis but in digital domain. Furthermore, they introduced the term ''virtual analog`` to represent the digital simulation of the analog audio devices. Since then research on digital signal processing methods for subtractive sound synthesis has increased and more and more interest has been shown to the topic in the past few years. In fact, the last year (2010) was again record-breaking when the number of publication on this topic is considered. In this paper, the history of the publications of virtual analog synthesis research is reviewed. In addition to the publication count over the years, the viewpoints of this paper include the distribution of the methodologies presented in the publications.

The explosion of 20th-century technology, especially in the second half (IBM Computers), initially helped Universities & Research Centers to study a wide range of scientific fields. People like Max Mathews and John Chowning, using the programming languages of the time (e.g. MUSIC-V), researched and developed theories which become part of our music culture through a plethora of Synthesizers (Yamaha DX7, PPG Wave, etc.). This development has led us today to the combination of Hardware and Inside the Box philosophy (DAW, plug-ins, VST-i), with many of the VST-i inevitably inspired by Hardware Synthesizers.
This thesis examines four of the primary sound synthesis techniques (Additive, Subtractive, FM, Wavetable) and helps the user apply them to a synthesizer. My goal is not to create an "ultimate" synthesizer, but through the simple synthesizers (Max Patches) that I developed, to understand better the behaviour of sound synthesis techniques and how synthesizers work.

In this paper, a simple finite difference scheme for a rect-angular dynamic nonlinear plate, under free boundary con-ditions is presented. The algorithm is straightforward to program, and is capable of reproducing, to a first approx-imation, the behaviour of various percussion instruments whose timbre depends crucially on nonlinear effects (due to high-speed strikes), including transient pitch glides and the buildup of high-frequency energy. Though computa-tionally intensive, algorithms such as that presented here promise more faithful sound synthesis and, as with all phys-ical model inspired synthesis algorithms, require the specifi-cation of only a few, physically meaningful parameters. Full details of the algorithm, including the setting of boundary conditions and computational demands are provided. Nu-merical simulation results are presented.

Introduction to computer music, the novelty of Csound6 language, real-time, envelopes, tremolo, vibrato, spatialisation, generative processes, MIDI control, OSC, additive synthesis, subtractive synthesis, FM synthesis, granular synthesis, physical modeling, scanned synthesis, signal processors, graphical user interfaces creation, integration in Max / MSP and Ableton Live, VST plugins development and much more.

Mit einem Minimum an mathematischen Formeln und in leicht verständlicher Sprache bietet das vorliegende Buch auf dem aktuellen Stand der Forschung eine grundlegende Einfüh-rung in die einzelnen Fachgebiete der Musikal ischen Akustik (Instrumenten-, Raum-und Psychoakustik, Auditorische Szenenanalyse, Musikalische Wahrnehmung und Klangsyn-these) sowie in die Geschichte, Perspektiven und Methoden dieses vielschichtigen Bereichs der Musikwissenschaft. Am Beispiel einer Konzertsituation wird der Weg von der Schallerzeugung am Musikinstru-ment bzw. der Singstimme über die Schallausbreitung im Raum bis hin zum Ohr des Zuhörers und der dahinter stattfindenden kognitiven und alltagsrelevanten Schallverarbeitung erläu-tert. Hierbei werden sowohl die aktuellen Messmethoden und akustischen Ana lyseverfahren berücksichtigt als auch die Funktionsweise und Geschichte der elektronischen Musikinstru-mente und der modernen Audiotechnik.

In quasi-bandlimited classical waveform oscillators, the aliasing distortion present in a trivially sampled waveform can be reduced in the digital domain by applying a tabulated correction function. This paper presents an approach that applies the correction function in the differentiated domain by synthesizing a bandlimited impulse train (BLIT) that is integrated to obtain the desired bandlimited waveform. The ideal correction function of the BLIT method is infinitely long and in practice needs to be windowed. In order to obtain a good alias-reduction performance, long tables are typically required. It is shown that when a short look-up table is used, a windowed ideal correction function does not provide the best performance in terms of minimizing aliasing audibility. Instead, audibly improved alias-reduction performance can be obtained using a look-up table that has a parametric control over the low-order generations of aliasing. Some practical parametric look-up table designs are discussed in this paper, and their use and alias-reduction performance are exemplified. The look-up table designs discussed in this paper providing the best alias-reduction performance are parametric window functions and least-squares optimized multi-band FIR filter designs.

Le titre de ce travail peut difficilement refléter la complexité et l’étendue de son sujet essentiel: la naissance et le développement d’une conception quantique  du son. Bien entendu, le point de départ de cette thèse est centré sur un intérêt spécifique envers les techniques granulaires dans la synthèse sonore, mais la recherche m’a poussé peu à peu à découvrir leurs bases scientifiques. Or, les notions quantiques issues de ces bases ont déterminé l’aspect psychoacoustique des méthodes granulaires, et elles ont eu également un effet décisif sur leur aspect esthétique. 

Dans ce travail, je ne me suis pas limité à l’étude des méthodes granulaires. Mon objectif a été aussi celui de donner une vision globale de l’impact des idées quantiques sur la musique en général. Pourtant, le champ électroacoustique a constitué le noyau de mon travail, et à partir de là, différentes idées esthétiques ont été développées. Or, cette démarche possède une logique. Une technique de synthèse quelconque aura toujours une influence sur les sons produits, et les différentes caractéristiques de cette technique particulière détermineront toujours une esthétique spécifique.

The AurumSolis TRANQUIL SPACE MK1 is an experimental prototype noise-reducing audio system. The theory behind this project is that by picking up ambient sound ​with a microphone and reproducing it out of phase, it is possible to actively cancel or "null" out background noise.

The dissertation deals with an implementation of the woodwind instrument
physical model using the digital waveguide modelling technique. An
essential theory concerning acoustical properties of woodwind instruments
and applicable modelling methods is discussed. This includes an analysis of
two existing models of clarinet and flute instruments, which serves as a
starting point for the implementation of the “Irish whistle” physical model in
Matlab. The final result is the “Irish whistle” application capable of
producing a whistle-like sound. Graphical User Interface allows a convenient
control over the model and excitation source parameters. The evaluation of
the model consists of various comparison tests with a real Irish whistle and
an examination of the most interesting model parameters.
Praca

Structured for use in university courses, the book is an overview of the theory and practice of Max/MSP, with a glossary of terms and suggested tests that allow students to evaluate their progress. Comprehensive online support, running parallel to the explanations in the book, includes hundreds of sample patches, analyses, interactive sound-building exercises, and reverse engineering exercises. This book will provide a reader with skill and understanding in using Max/MSP for sound design and musical composition.

Second Edition updated for MAX 6) More than 500 pages, interactive examples, hundreds of patches, online support, theory and Max/MSP glossary, test, reverse engineering exercises, analyses, completion and correction of patches, etc.

Chapter I – Introduction to Sound Synthesis and Signal Processing
Chapter II – Additive Synthesis
Chapter III – Subtractive Synthesis
Chapter IV – Controls and LFO

In this thesis, the design of a music synthesizer for systems suffering from limitations in computing power and memory capacity is presented. First, different possible synthesis techniques are reviewed and their applicability in computationally efficient music synthesis is discussed. In practice, the applicable techniques are limited to additive and source-filter synthesis, and, in special cases, to frequency modulation, wavetable and sampling synthesis.

Next, the design of the structures of the applicable techniques are presented in detail, and properties and design issues of these structures are discussed. A major implementation problem is raised in digital source-filter synthesis, where the use of classic waveforms, such as sawtooth wave, as the source signal is challenging due to aliasing caused by waveform discontinuities. Methods for existing bandlimited waveform synthesis are reviewed, and a new approach using polynomial bandlimited step function is presented in detail with design rules for the applicable polynomials. The approach is also tested with two different third-order polynomials. They reduce aliasing more at high frequencies, but at low frequencies their performance is worse than with the first-order polynomial. In addition, some commonly used sound effect algorithms are reviewed with respect to their applicability in computationally efficient music synthesis.

In many cases the sound synthesis system must be capable of producing music consisting of various different sounds ranging from real acoustic instruments to electronic instruments and sounds from nature. Therefore, the music synthesis system requires careful sound design. In this thesis, sound design rules for imitation of various sounds using the computationally efficient synthesis techniques are presented. In addition, the effects of the parameter variation for the design of sound variants are presented.

This report accompanies the author’s graduation project for fulfillment of the EMMA at the Hogeschool voor de Kunsten in Utrecht. In chapter 1 the research question and objectives
of the project are stated, which can be summarized as follows: To explore the artistic possibilities of mapping a mathematical grounded model onto musical parameters to design and control electronic sounds and to use the same model to generate larger-scale musical structure and form. These experiments have culminated in an algorithmic sound composition in which the designed mappings are used as effectively as possible. The motivation for carying out this project is primarily out of a general interest with regards to making music and designing sounds from a creative perspective, developing software from a technological perspective, but also to seriously study and apply theoretical physics and mathematics.

The presentation of the results of the project is preceded by a critical review of relevant
work and repertoire by others in chapter 2. It discusses literature concerning the mapping of
equations to musical parameters, but also some more generalized software implementations,
musical works and compositional strategies.

Chapter 3 is devoted to the derivation of the Schrödinger equation and its solution for the hydrogen atom. The mathematics involved might be rather challenging for the
unexperienced reader, but the discussion will be kept as simple as possible and there will be references to more specialized literature whenever necessary. The chapter ends with some graphical examples of the designed software implementations of the solutions of the Schrödinger equation for a hydrogen atom. The SuperCollider classes that govern this implementation are part of the MathLib Quark which is publicly available for download.

Chapter 4 is a detailed technical description of the project and the algorithmic sound composition in which the technical implementation of the designed algorithmic methods and
techniques is described as well as the sound synthesis model used for the generation of all sounds.

Chapter 5 is an evaluation of the project from a number of different perspectives and tries to answer the research questions posed in chapter 1. In addition, a number of alternative tests are conducted to justify the choice of working with the Schrödinger equation from a musical point of view.

In the last chapter, the strength and weaknesses of the design process, the programmed software and sound composition are summarized and a number of suggestions are made for future experiments and subjects to focus on.

All available sound material that is related to the project is included on the cd accompanying this report or alternatively, can be listened to at http://www.soundcloud.com/michaeldzjap.

The aim of this thesis is to investigate techniques for, and applications of automatic sound synthesizer programming. An automatic sound synthesizer programmer is a system which removes the requirement to explicitly specify parameter settings for a sound synthesis algorithm from the user. Two forms of these systems are discussed in this thesis: tone matching programmers and synthesis space explorers. A tone matching programmer takes at its input a sound synthesis algorithm and a desired target sound.

There exists a tenacious dichotomy between the 'organic' and the 'synthetic' in electronic music. In spite of immediate opposing qualities, they instill sensations of each other in practice: Acoustic sounds are subject to artificial mimicry while algorithmic music can present an imitation of human creativity. This presents a post-human aesthetic that questions if there are echoes of life in the machine. This paper investigates how sonic aesthetics in the borderlands between electronic avant-garde, pop and club music have changed after 2010. I approach these aesthetics through Brian Massumi's notions of 'semblance' and 'animateness' as abstract monikers to assist in the tracing of meta-aesthetic experiences of machine-life, genre, musical structure and the listening to known-unknown noise. The idea of a post-Internet society acts as framework for the intimate relation between pop and avant-garde in contemporary electronic music. This relation is a result of a sonic contextualization of late-capitalist society via the Internet. Finally I discuss how pop is the noise in the avant-garde and how the human in a post-Internet era presents itself through synthetic plasticity.