Designing and synthesizing delay-based audio effect using the CORDIS-ANIMA physical modeling formalism

International audienceThroughout this paper, several CORDIS-ANIMA physical models will be presented to offer an alternative synthesis of some classical delay-based digital audio effects: a delay model, two comb filter models, three flanger models and a sound spatialization model. Several of these realizations support a control scheme based on the ''Physical Instrumental Interaction''. Additionally they provide several sonic characteristics which do not appear in the original ones. Especially the flanger model, for certain parameter values may give a new digital audio effect between flanging and filtering

Computationally efficient music synthesis : methods and sound design

Tässä diplomityössä esitetään musiikkisyntetisaattorin suunnittelua systeemille, jonka laskentateho ja muistikapasiteetti ovat rajoitettuja. Ensiksi kerrataan mahdollisia synteesitekniikoita sekä arvioidaan niiden käyttökelpoisuutta laskennallisesti tehokkaassa musiikkisynteesissä. Käytännössä käyttökelpoiset tekniikat ovat lisäävä ja lähde-suodinsynteesit, ja erikoistapauksissa taajuusmodulaatio-, aaltotaulukko- ja samplaussynteesit. Tämän jälkeen käyttökelpoisten tekniikoiden rakenteiden suunnittelua esitetään tarkemmin, sekä esitetään näiden rakenteiden ominaisuuksia ja suunnitteluongelmia. Suurin ongelma kohdataan digitaalisessa lähde-suodinsynteesissä, jossa klassisten aaltomuotojen, kuten saha-aallon käyttö lähdesignaalina on ongelmallista laskostumisen takia, joka johtuu aaltomuodossa olevista epäjatkuvuuksista. Olemassa olevia kaistarajoitettuja aaltomuotosynteesimenetelmiä kerrataan, ja polynomimuotoiseen kaistarajoitetuun askelfunktioon perustuvaa menetelmää esitellään tarkemmin antamalla suunnittelusääntöjä käyttökelpoisille polynomeille. Menetelmää testataan lisäksi kahdella kolmannen asteen polynomilla. Nämä polynomit vähentävät laskostumista korkeilla taajuuksilla enemmän verrattuna ensimmäisen asteen polynomiin, mutta pienillä taajuksilla ensimmäisen asteen polynomi tuottaa parempia tuloksia. Lisäksi kerrataan muita mahdollisia ääniefektialgoritmeja ja arvioidaan niiden käyttökelpoisuutta laskennallisesti tehokkaassa musiikkisynteesissä. Useasti äänisynteesisysteemin täytyy pystyä generoimaan musiikkia, jossa käytetään monia erilaisia ääniä, jotka ulottuvat oikeista akustisista soittimista elektronisiin soittimiin ja luonnon ääniin. Siksi tällainen systeemi tarvitsee huolellista äänten suunnittelua. Tässä diplomityössä esitetään suunnittelusääntöjä erilaisten äänien imitoimiseksi. Lisäksi esitellään synteesimenetelmien parametrien vaikutus äänivarianttien suunnitteluun.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

Designing sound : procedural audio research based on the book by Andy Farnell

In procedural media, data normally acquired by measuring something, commonly described as sampling, is replaced by a set of computational rules (procedure) that defines the typical structure and/or behaviour of that thing. Here, a general approach to sound as a definable process, rather than a recording, is developed. By analysis of their physical and perceptual qualities, natural objects or processes that produce sound are modelled by digital Sounding Objects for use in arts and entertainments. This Thesis discusses different aspects of Procedural Audio introducing several new approaches and solutions to this emerging field of Sound Design.Em Media Procedimental, os dados os dados normalmente adquiridos através da medição de algo habitualmente designado como amostragem, são substituídos por um conjunto de regras computacionais (procedimento) que definem a estrutura típica, ou comportamento, desse elemento. Neste caso é desenvolvida uma abordagem ao som definível como um procedimento em vez de uma gravação. Através da análise das suas características físicas e perceptuais , objetos naturais ou processos que produzem som, são modelados como objetos sonoros digitais para utilização nas Artes e Entretenimento. Nesta Tese são discutidos diferentes aspectos de Áudio Procedimental, sendo introduzidas várias novas abordagens e soluções para o campo emergente do Design Sonoro

Hardward and algorithm architectures for real-time additive synthesis

Additive synthesis is a fundamental computer music synthesis paradigm tracing its origins to the work of Fourier and Helmholtz. Rudimentary implementation linearly combines harmonic sinusoids (or partials) to generate tones whose perceived timbral characteristics are a strong function of the partial amplitude spectrum. Having evolved over time, additive synthesis describes a collection of algorithms each characterised by the time-varying linear combination of basis components to generate temporal evolution of timbre. Basis components include exactly harmonic partials, inharmonic partials with time-varying frequency or non-sinusoidal waveforms each with distinct spectral characteristics. Additive synthesis of polyphonic musical instrument tones requires a large number of independently controlled partials incurring a large computational overhead whose investigation and reduction is a key motivator for this work. The thesis begins with a review of prevalent synthesis techniques setting additive synthesis in context and introducing the spectrum modelling paradigm which provides baseline spectral data to the additive synthesis process obtained from the analysis of natural sounds. We proceed to investigate recursive and phase accumulating digital sinusoidal oscillator algorithms, defining specific metrics to quantify relative performance. The concepts of phase accumulation, table lookup phase-amplitude mapping and interpolated fractional addressing are introduced and developed and shown to underpin an additive synthesis subclass - wavetable lookup synthesis (WLS). WLS performance is simulated against specific metrics and parameter conditions peculiar to computer music requirements. We conclude by presenting processing architectures which accelerate computational throughput of specific WLS operations and the sinusoidal additive synthesis model. In particular, we introduce and investigate the concept of phase domain processing and present several “pipeline friendly” arithmetic architectures using this technique which implement the additive synthesis of sinusoidal partials

Probabilistic characterization and synthesis of complex driven systems

Thesis (Ph.D.)--Massachusetts Institute of Technology, School of Architecture and Planning, Program in Media Arts and Sciences, 2000.Includes bibliographical references (leaves 194-204).Real-world systems that have characteristic input-output patterns but don't provide access to their internal states are as numerous as they are difficult to model. This dissertation introduces a modeling language for estimating and emulating the behavior of such systems given time series data. As a benchmark test, a digital violin is designed from observing the performance of an instrument. Cluster-weighted modeling (CWM), a mixture density estimator around local models, is presented as a framework for function approximation and for the prediction and characterization of nonlinear time series. The general model architecture and estimation algorithm are presented and extended to system characterization tools such as estimator uncertainty, predictor uncertainty and the correlation dimension of the data set. Furthermore a real-time implementation, a Hidden-Markov architecture, and function approximation under constraints are derived within the framework. CWM is then applied in the context of different problems and data sets, leading to architectures such as cluster-weighted classification, cluster-weighted estimation, and cluster-weighted sampling. Each application relies on a specific data representation, specific pre and post-processing algorithms, and a specific hybrid of CWM. The third part of this thesis introduces data-driven modeling of acoustic instruments, a novel technique for audio synthesis. CWM is applied along with new sensor technology and various audio representations to estimate models of violin-family instruments. The approach is demonstrated by synthesizing highly accurate violin sounds given off-line input data as well as cello sounds given real-time input data from a cello player.by Bernd Schoner.Ph.D

Courseware and curriculum development for a wireless electronics class

Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2007.Includes bibliographical references (p. 169-170).The design of basic wireless building blocks (such as oscillators, amplifiers, and modulation circuits) and modern encoding techniques such as CDMA (Code Division Multiple Access) are in high demand by employers of recent graduates. This thesis sets forth a lesson plan and laboratory kit design to be used in the development of a new class that teaches these wireless system design techniques. Such a class will help students gain both the theoretical and practical experience required of them in today's industry.by Ariel Rodriguez.M.Eng

Physically Informed Subtraction of a String's Resonances from Monophonic, Discretely Attacked Tones : a Phase Vocoder Approach

A method for the subtraction of a string's oscillations from monophonic, plucked- or hit-string tones is presented. The remainder of the subtraction is the response of the instrument's body to the excitation, and potentially other sources, such as faint vibrations of other strings, background noises or recording artifacts. In some respects, this method is similar to a stochastic-deterministic decomposition based on Sinusoidal Modeling Synthesis [MQ86, IS87]. However, our method targets string partials expressly, according to a physical model of the string's vibrations described in this thesis. Also, the method sits on a Phase Vocoder scheme. This approach has the essential advantage that the subtraction of the partials can take place \instantly", on a frame-by-frame basis, avoiding the necessity of tracking the partials and therefore availing of the possibility of a real-time implementation. The subtraction takes place in the frequency domain, and a method is presented whereby the computational cost of this process can be reduced through the reduction of a partial's frequency-domain data to its main lobe. In each frame of the Phase Vocoder, the string is encoded as a set of partials, completely described by four constants of frequency, phase, magnitude and exponential decay. These parameters are obtained with a novel method, the Complex Exponential Phase Magnitude Evolution (CSPME), which is a generalisation of the CSPE [SG06] to signals with exponential envelopes and which surpasses the nite resolution of the Discrete Fourier Transform. The encoding obtained is an intuitive representation of the string, suitable to musical processing

Millimetre wave quasi-optical signal processing and spread spectrum techniques

The use of quasi-optical techniques for signal processing at millimetre wave frequencies and their application to spread spectrum systems is presented in this thesis. Millimetre waves offer a number of advantages when compared to microwave and optical signals and as a result they are finding an increasing number of applications in the area of communications. Traditionally, millimetre wave research has been centred on scientific and experimental interests. In recent years, however, practical communications systems have been emerging at these frequencies. New technologies are being developed which aim to exploit the potential advantages of millimetric signals and the work undertaken here relates to one particular avenue - that of quasi-optics. This thesis introduces measurement methods for assessing millimetric components and quasi-optical systems, followed by an appraisal of Gunn oscillators and their associated power supplies as suitable signal sources for communications. A type of Gunn oscillator displaying semi-chaotic behaviour is described which may have potential as an inherently broadband source suitable for spread spectrum applications. The application of quasi-optical signal processing methods to a prototype spread spectrum millimetre wave system is presented. The system has been shown to operate satisfactorily in a number of demonstrations. Theoretical models of the quasi-optical circuit show good agreement with experiment and an analysis of the signal-to-noise behaviour predicts the potential receiver performance. Some ideas for future work, building on that presented here, are suggested