Download Antiderivative Antialiasing with Frequency Compensation for Stateful Systems Employing nonlinear functions in audio DSP algorithms requires attention as they generally introduce aliasing. Among others, antiderivative antialiasing proved to be an effective method for static nonlinearities and gave rise to a number of variants, including our AA-IIR method. In this paper we introduce an improvement to AA-IIR that makes it suitable for use in stateful systems. Indeed, employing standard antiderivative antialiasing techniques in such systems alters their frequency response and may cause stability issues. Our method consists in cascading a digital filter after the AA-IIR block in order to fully compensate for unwanted delay and frequency-dependent effects. We study the conditions for such a digital filter to be stable itself and evaluate the method by applying it to the diode clipper circuit.
Download A Quaternion-Phase Oscillator An approach to designing dynamical systems with a three-dimensional state space is described that can be used to build a variety of non-periodic oscillators. The state space is taken to be a 3sphere, which is identified with the manifold of unit quaternions. Any such system can be described as a quaternion-valued ordinary differential equation, which is digitally realized using an approximation as a finite difference e quation. Two examples are shown. Compared to previous applications of dynamical systems used to generate audio samples, the approach described here offers a wide choice of specific flows which can neither diverge nor approach a stable limit point.
Download Deforming the Oscillator: Iterative Phases Over Parametrizable Closed Paths Iterative phase formulations allow for the generalization of many oscillatory sound synthesis methods from circles to general parametrizable loops, with or without explicit geometric contexts. This paper describes this approach, leading to the ability to perform modulation, feedback and chaotic oscillations over deformed circles that can include ill-behaved geometries, while allowing modulations or feedback to be deformed as well.
Download Continuous State Modeling for Statistical Spectral Synthesis Continuous State Markovian Spectral Modeling is a novel approach for parametric synthesis of spectral modeling parameters, based on the sines plus noise paradigm. The method aims specifically at capturing shimmer and jitter - micro-fluctuations in the partials’ frequency and amplitude trajectories, which are essential for the timbre of musical instruments. It allows for parametric control over the timbral qualities, while removing the need for the more computationally expensive and restrictive process of the discrete state space modeling method. A qualitative comparison between an original violin sound and a re-synthesis shows the ability of the algorithm to reproduce the micro-fluctuations, considering their stochastic and spectral properties.
Download Subjective Evaluation of Sound Quality and Control of Drum Synthesis with Stylewavegan In this paper we investigate into perceptual properties of StyleWaveGAN, a drum synthesis method proposed in a previous publication. For both, the sound quality as well as the control precision StyleWaveGAN has been shown to deliver state of the art performance for quantitative metrics (FAD and MSE of the control parameters). The present paper aims to provide insight into the perceptual relevance of these results. Accordingly, we performed a subjective evaluation of the sound quality as well as a subjective evaluation of the precision of the control using timbre descriptors from the AudioCommons toolbox. We evaluate the sound quality with mean opinion score and make measurements of psychophysical response to the variations of the control. By means of the perceptual tests, we demonstrate that StyleWaveGAN produces better sound quality than state-of-the-art model DrumGAN and that the mean control error is lower than the absolute threshold of perception at every point of measurement used in the experiment.
Download HD-AD: A New Approach to Audio Atomic Decomposition with Hyperdimensional Computing In this paper, we approach the problem of atomic decomposition of audio at the symbolic level of atom parameters through the lens of hyperdimensional computing (HDC) – a non-traditional computing paradigm. Existing atomic decomposition algorithms often operate using waveforms from a redundant dictionary of atoms causing them to become increasingly memory/computationally intensive as the signal length grows and/or the atoms become more complicated. We systematically build an atom encoding using vector function architecture (VFA), a field of HDC. We train a neural network encoder on synthetic audio signals to generate these encodings and observe that the network can generalize to real recordings. This system, we call Hyperdimensional Atomic Decomposition (HD-AD), avoids time-domain correlations all together. Because HD-AD scales with the sparsity of the signal, rather than its length in time, atomic decompositions are often produced much faster than real-time.
Download Combined Derivative/Antiderivative Antialiasing Nonlinear systems play an important role in musical signal processing, but their digital implementation suffers from the occurrence of aliasing distortion. Consequently, various aliasing reduction methods have been proposed in the literature. In this work, a novel approach is examined that uses samples of a signal’s derivative in addition to the signal’s samples themselves. This allows some aliasing reduction, but is usually insufficient on its own. However, it can elegantly and fruitfully be combined with antiderivative antialiasing to obtain an effective method. Unfortunately, it still compares unfavorably to oversampled antiderivative antialiasing. It may therefore be regarded as a negative result, but it may hopefully still form a basis for further developments.
Download A Low-Latency Quasi-Linear-Phase Octave Graphic Equalizer This paper proposes a low-latency quasi-linear-phase octave graphic equalizer. The structure is derived from a recent linearphase graphic equalizer based on interpolated finite impulse response (IFIR) filters. The proposed system reduces the total latency of the previous equalizer by implementing a hybrid structure. An infinite impulse response (IIR) shelving filter is used in the structure to implement the first band of the equalizer, whereas the rest of the band filters are realized with the linear-phase FIR structure. The introduction of the IIR filter causes a nonlinear phase response in the low frequencies, but the total latency is reduced by 50% in comparison to the linear-phase equalizer. The proposed graphic equalizer is useful in real-time audio processing, where only little latency is tolerated.
Download Time-Varying Filter Stability and State Matrix Products We show a new sufficient criterion for time-varying digital filter stability: that the matrix norm of the product of state matrices over a certain finite number of time steps is bounded by 1. This extends Laroche’s Criterion 1, which only considered one time step, while hinting at extensions to two time steps. Further extending these results, we also show that there is no intrinsic requirement that filter coefficients be frozen over any time scale, and extend to any dimension a helpful theorem that allows us to avoid explicitly performing eigen- or singular value decompositions in studying the matrix norm. We give a number of case studies on filters known to be time-varying stable, that cannot be proven time-varying stable with the original criterion, where the new criterion succeeds.
Download A General Antialiasing Method for Sine Hard Sync Hard sync is a feature appearing in many analog synthesizers: it consists in retriggering a slave oscillator, regardless of its phase, every time a master oscillator completes its cycle. If this process is naïvely implemented digitally, it is subject to aliasing. While for sawtooth, square, and triangle waves several effective antialiasing methods have been developed, the literature is sparser concerning sine hard sync, arguably because discontinuities of infinite order are introduced which are more difficult to handle. In this paper, we introduce a new antialiasing algorithm for sine hard sync which is obtained by filtering the hard-synced sine with a FIR lowpass kernel, as opposed to existing methods based on the windowed sinc function. We show that our method yields lower computational cost and better aliasing reduction.