Fabian Esqueda

• D.Sc. (Tech)
• Native Instruments GmbH

In this paper we present a virtual analog model of the voltage- controlled filter used in the EDP Wasp synthesizer. This circuit is an interesting case study for virtual analog modeling due to its characteristic nonlinear and highly dynamic behavior which can be attributed to its unusual design. The Wasp filter consists of a state variable filter t...

An assessment of filters for classic oversampled audio waveshaping schemes is carried out in this paper, pursuing aliasing reduction. For this purpose, the quality measure of the A-weighted noise-to-mask ratio is computed for test tones covering the frequency range from 27.5 Hz to 4.186 kHz, sampled at 44.1 kHz, and processed at eight-times oversam...

This paper proposes signal processing methods to extend a stationary part of an audio signal endlessly. A frequent occasion is that there is not enough audio material to build a synthesizer, but an example sound must be extended or modified for more variability. Filtering of a white noise signal with a filter designed based on high-order linear pre...

This work studies the use of signal-driven synthesis algorithms applied to an augmented guitar. A robust sub-octave generator, partially modeled after a classic audio-driven monophonic guitar synthesizer design of the 1970s is presented. The performance of the proposed system is evaluated within the context of an augmented active guitar with an act...

The Serge Triple Waveshaper (TWS) is a synthesizer module de- signed in 1973 by Serge Tcherepnin, founder of Serge Modular Music Systems. It contains three identical waveshaping circuits that can be used to convert sawtooth waveforms into sine waves. However, its sonic capabilities extend well beyond this particular application. Each processing sec...

Wavefolders are a particular class of nonlinear waveshaping circuits, and a staple of the “West Coast” tradition of analog sound synthesis. In this paper, we present analyses of two popular wavefolding circuits—the Lockhart and Serge wavefolders—and show that they achieve a very similar audio effect. We digitally model the input–output relationship...

Abstract—The analog voltage-controlled filter used in historical music synthesizers by Moog is modeled using a digital system, which is then compared in terms of audio measurements with the original analog filter. The analog model is mainly borrowed from D’Angelo’s previous work. The digital implementation of the filter incorporates a recently prop...

Aliasing is major problem in any audio signal processing chain involving nonlinearity. The usual approach to antialiasing involves operation at an oversampled rate—usually 4 to 8 times an audio sample rate. Recently, a new approach to antialiasing in the case of memoryless nonlinearities has been proposed, which relies on operations over the antide...

An antialiased digital model of the wavefolding circuit inside the Buchla 259 Complex Waveform Generator is presented. Wave-folding is a type of nonlinear waveshaping used to generate complex harmonically-rich sounds from simple periodic waveforms. Unlike other analog wavefolder designs, Buchla's design features five op-amp-based folding stages arr...

This work presents a novel virtual analog model of the Lockhart wavefolder. Wavefolding modules are among the fundamental elements of 'West Coast' style analog synthesis. These circuits produce harmonically-rich waveforms when driven by simple periodic signals such as sinewaves. Since wavefolding introduces high levels of harmonic distortion , we p...

Aliasing is a commonly-encountered problem in audio signal processing, particularly when memoryless nonlinearities are simulated in discrete time. A conventional remedy is to operate at an oversampled rate. A new aliasing reduction method is proposed here for discrete-time memoryless nonlinearities, which is suitable for operation at reduced oversa...

An aliasing reduction method for hard-clipped sampled signals is proposed. Clipping in the digital domain causes a large amount of harmonic distortion, which is not bandlimited, so spectral components generated above the Nyquist limit are reflected to the baseband and mixed with the signal. A model for an ideal bandlimited ramp function is derived,...

A study on the limits of bandlimited correction functions used to eliminate aliasing in audio signals with discontinuities is presented. Trivial sampling of signals with discontinuities in their waveform or their derivatives causes high levels of aliasing distortion due to the infinite bandwidth of these discontinuities. Geometrical oscillator wave...

The use of the bandlimited ramp (BLAMP) function as an antialiasing tool for audio signals with sharp corners is presented. Discontinuities in the waveform of a signal or its derivatives require infinite bandwidth and are major sources of aliasing in the digital domain. A polynomial correction function is modeled after the ideal BLAMP function. Thi...

A method to measure the response of a linear time-variant (LTV) audio system is presented. The proposed method uses a series of short chirps generated as the impulse response of several cascaded allpass filters. This test signal can measure the characteristics of an LTV system as a function of time. Results obtained from testing of this method on a...

An efficient method for aliasing reduction under soft clipping using a piecewise polynomial is presented. Soft clipping is commonly used to model the saturating behavior of electronic musical systems such as guitar amplifiers and voltage-controlled filters used in subtractive synthesis. Saturations introduce high levels of harmonic distortion and,...

A new method for aliasing reduction in soft-clipping nonlinearities is proposed. Digital implementations of saturating systems introduce harmonic distortion which, if untreated, gets reflected at the Nyquist limit and is mixed with the signal. This is called aliasing and is heard as a disturbance. A new correction function, derived by integrating t...

Various ways to implement infinitely rising or falling spectral notches, also known as the barberpole phaser and flanging illusions, are described and studied. The first method is inspired by the Shepard-Risset illusion, and is based on a series of several cascaded notch filters moving in frequency one octave apart from each other. The second metho...

Soft-clipping algorithms used to implement musical distortion effects are major sources of aliasing due to their nonlinear behavior. It is a research challenge to design computation-ally efficient methods for alias-free distortion without over-sampling. In the proposed approach, soft clipping is decomposed into a hard clipper and a low-order polyno...