Download An Efficient Phasiness Reduction Technique for Moderate Audio Time-Scale Modification Phase vocoder approaches to time-scale modification of audio introduce a reverberant/phasy artifact into the time-scaled output due to a loss in phase coherence between short-time Fourier transform (STFT) bins. Recent improvements to the phase vocoder have reduced the presence of this artifact, however, it remains a problem. A method of time-scaling is presented that results in a further reduction in phasiness, for moderate time-scale factors, by taking advantage of some flexibility that exists in the choice of phase required so as to maintain horizontal phase coherence between related STFT bins. Furthermore, the approach leads to a reduction in computational load within the range of time-scaling factors for which phasiness is reduced.
Download Audio Processing Using Haskell The software for most today’s applications including signal processing applications is written in imperative languages. Imperative programs are fast because they are designed close to the architecture of the widespread computers, but they don’t match the structure of signal processing very well. In contrast to that, functional programming and especially lazy evaluation perfectly models many common operations on signals. Haskell is a statically typed, lazy functional programming language which allow for a very elegant and concise programming style. We want to sketch how to process signals, how to improve safety by the use of physical units, and how to compose music using this language.
Download Towards Efficient Emulation of Nonlinear Analog Circuits for Audio Using Constraint Stabilization and Convex Quadratic Programming This paper introduces a computationally efficient method for
the emulation of nonlinear analog audio circuits by combining state-space representations, constraint stabilization, and convex quadratic programming (QP). Unlike traditional virtual analog (VA) modeling approaches or computationally demanding
SPICE-based simulations, our approach reformulates the nonlinear
differential-algebraic (DAE) systems that arise from analog circuit
analysis into numerically stable optimization problems. The proposed method efficiently addresses the numerical challenges posed
by nonlinear algebraic constraints via constraint stabilization techniques, significantly enhancing robustness and stability, suitable
for real-time simulations. A canonical diode clipper circuit is presented as a test case, demonstrating that our method achieves accurate and faster emulations compared to conventional state-space
methods. Furthermore, our method performs very well even at
substantially lower sampling rates. Preliminary numerical experiments confirm that the proposed approach offers improved numerical stability and real-time feasibility, positioning it as a practical
solution for high-fidelity audio applications.
Download Interactive digital audio environments: gesture as a musical parameter This paper presents some possible relationships between gesture and sound that may be built with an interactive digital audio environment. In a traditional musical situation gesture usually produces sound. The relationship between gesture and sound is unique, it is a cause to effect link. In computer music, the possibility of uncoupling gesture from sound is due to the fact that computer can carry out all the aspects of sound production from composition up to interpretation and performance. Real time computing technology and development of human gesture tracking systems may enable gesture to be introduced again into the practice of computer music, but with a completely renewed approach. There is no more need to create direct cause to effect relationships for sound production, and gesture may be seen as another musical parameter to play with in the context of interactive musical performances.
Download Two polarisation finite difference model of bowed strings with nonlinear contact and friction forces Recent bowed string sound synthesis has relied on physical modelling techniques; the achievable realism and flexibility of gestural control are appealing, and the heavier computational cost becomes less significant as technology improves. A bowed string is simulated in two polarisations by discretising the partial differential equations governing its behaviour, using the finite difference method; a globally energy balanced scheme is used, as a guarantee of numerical stability under highly nonlinear conditions. In one polarisation, a nonlinear contact model is used for the normal forces exerted by the dynamic bow hair, left hand fingers, and fingerboard. In the other polarisation, a force-velocity friction curve is used for the resulting tangential forces. The scheme update requires the solution of two nonlinear vector equations.Sound examples and video demonstrations are presented.
Download Nicht-negativeMatrixFaktorisierungnutzendes-KlangsynthesenSystem (NiMFKS): Extensions of NMF-based Concatenative Sound Synthesis Concatenative sound synthesis (CSS) entails synthesising a “target” sound with other sounds collected in a “corpus.” Recent work explores CSS using non-negative matrix factorisation (NMF) to approximate a target sonogram by the product of a corpus sonogram and an activation matrix. In this paper, we propose a number of extensions of NMF-based CSS and present an open MATLAB implementation in a GUI-based application we name NiMFKS. In particular we consider the following extensions: 1) we extend the NMF framework by implementing update rules based on the generalised β-divergence; 2) We add an optional monotonic algorithm for sparse-NMF; 3) we tackle the computational challenges of scaling to big corpora by implementing a corpus pruning preprocessing step; 4) we generalise constraints that may be applied to the activation matrix shape; and 5) we implement new modes of interacting with the procedure by enabling sketching and modifying of the activation matrix. Our application, NiMFKS and source code can be downloaded from here: https: //code.soundsoftware.ac.uk/projects/nimfks.
Download Visualaudio-Design – Towards a Graphical Sounddesign VisualAudio-Design (VAD) is a spectral-node based approach to visually design audio collages and sounds. The spectrogram as a visualization of the frequency-domain can be intuitively manipulated with tools known from image processing. Thereby, a more comprehensible sound design is described to address common abstract interfaces for DSP algorithms that still use direct value inputs, sliders, or knobs. In addition to interaction in the timedomain of audio and conventional analysis and restoration tasks, there are many new possibilities for spectral manipulation of audio material. Here, affine transformations and two-dimensional convolution filters are proposed.
Download Combining Zeroth and First-Order Analysis With Lagrange Polynomials to Reduce Artefacts in Live Concatenative Granulation This paper presents a technique addressing signal discontinuity and concatenation artefacts in real-time granular processing
with rectangular windowing. By combining zero-crossing synchronicity, first-order derivative analysis, and Lagrange polynomials, we can generate streams of uncorrelated and non-overlapping
sonic fragments with minimal low-order derivatives discontinuities. The resulting open-source algorithm, implemented in the
Faust language, provides a versatile real-time software for dynamical looping, wavetable oscillation, and granulation with reduced artefacts due to rectangular windowing and no artefacts
from overlap-add-to-one techniques commonly deployed in granular processing.
Download A toolkit for experimentation with signal interaction This paper will describe a toolkit for experimentation with signal interaction techniques, also commonly referred to as cross adaptive processing. The technique allows analyzed features of one audio signal to inform the processing of another. Earlier used mainly for mixing and post production purposes, we now want to use it creatively as an intervention in the musical communication between two performers. The idea stems from Stockhausen’s use of intermodulation in the 1960’s, and as such we might also call the updated technique interprocessing. Our interest in the technique comes as a natural extension to previous research on live processing as an instrumental and performative activity. The automatic control of signal processing routines is related to previous work on adaptive audio effects and automatic mixing. The focus for our investigation and experimentation with the current toolkit will be how this affects the musical communication between performers, and how it changes what they can and will play. The program code for the toolkit is available as a github repository1 under an open source license.
Download Modifying Signals in Transform Domain: a Frame-Based Inverse Problem Within this paper a method for morphing audio signals is presented. The theory is based on general frames and the modification of the signals is done via frame multiplier. Searching this frame multiplier with given input and output signal, an inverse problem occurs and a priori information is added with regularization terms. A closed-form solution is obtained by a diagonal approximation, i.e. using only the diagonal entries in the signal transformations. The proposed solutions for different regularization terms are applied to Gabor frames and to the constant-Q transform, based on non-stationary Gabor frames.