Phaseshaping oscillator algorithms for musical sound synthesis

This paper focuses on phaseshaping techniques and their relation to classical abstract synthesis methods. Elementary polynomial and geometric phaseshapers, such as those based on the modulo operation and linear transformations, are investigated. They are then applied to the generation of classic and novel oscillator effects by using nested phaseshaping compositions. New oscillator algorithms introduced in this paper include single-oscillator hard sync, triangle modulation, efficient supersaw simulation, and sinusoidal waveshape modulation effects. The digital waveforms produced with phaseshaping techniques are generally discontinuous, which leads to aliasing artifacts. Aliasing can be effectively reduced by modifying samples around each discontinuity using the previously proposed polynomial bandlimited step function (polyBLEP) method

Phaseshaping oscillator algorithms for musical sound synthesis

This paper focuses on phaseshaping techniques and their relation to classical abstract synthesis methods. Elementary polynomial and geometric phaseshapers, such as those based on the modulo operation and linear transformations, are investigated. They are then applied to the generation of classic and novel oscillator effects by using nested phaseshaping compositions. New oscillator algorithms introduced in this paper include single-oscillator hard sync, triangle modulation, efficient supersaw simulation, and sinusoidal waveshape modulation effects. The digital waveforms produced with phaseshaping techniques are generally discontinuous, which leads to aliasing artifacts. Aliasing can be effectively reduced by modifying samples around each discontinuity using the previously proposed polynomial bandlimited step function (polyBLEP) method

Examining the Oscillator Waveform Animation Effect

An enhancing effect that can be applied to analogue oscillators in subtractive synthesizers is termed Animation, which is an efficient way to create a sound of many closely detuned oscillators playing in unison. This is often referred to as a supersaw oscillator. This paper first explains the operating principle of this effect using a combination of additive and frequency modulation synthesis. The Fourier series will be derived and results will be presented to demonstrate its accuracy. This will then provide new insights into how other more general waveform animation processors can be designed

Virtual Analog Oscillator Hard Synchronisation: Fourier Series and an Efficient Implementation

This paper investigates a number of digital methods to produce the Analog subtractive synthesis effect of ‘Hard Synchronisation.’ While the original effect is produced by an explicit waveform phase reset, other approaches are given that produce an equivalent output. In particular, based on measurements taken from a real-analog synthesizer, a comb filtering model is proposed. This description ties in with earlier work but here an explicit structure is provided. This filter-based approach is then shown to be far more computationally efficient than the synchronisation by phase reset. This efficiency is at a minor cost as it is shown that it has a minimal impact on the sonic accuracy

Virtual Analog Oscillator Hard Synchronisation: Fourier series and an efficient implementation

This paper investigates a number of digital methods to produce the Analog subtractive synthesis effect of ‘Hard Synchronisation.’ While the original effect is produced by an explicit waveform phase reset, other approaches are given that produce an equivalent output. In particular, based on measurements taken from a real-analog synthesizer, a comb filtering model is proposed. This description ties in with earlier work but here an explicit structure is provided. This filter-based approach is then shown to be far more computationally efficient than the synchronisation by phase reset. This efficiency is at a minor cost as it is shown that it has a minimal impact on the sonic accuracy

Virtual Analog Oscillator Hard Synchronisation: Fourier Series and an Efficient Implementation

This paper investigates a number of digital methods to produce the Analog subtractive synthesis effect of ‘Hard Synchronisation.’ While the original effect is produced by an explicit waveform phase reset, other approaches are given that produce an equivalent output. In particular, based on measurements taken from a real-analog synthesizer, a comb filtering model is proposed. This description ties in with earlier work but here an explicit structure is provided. This filter-based approach is then shown to be far more computationally efficient than the synchronisation by phase reset. This efficiency is at a minor cost as it is shown that it has a minimal impact on the sonic accuracy

Virtual Analog Oscillator Hard Synchronisation: Fourier series and an efficient implementation

This paper investigates a number of digital methods to produce the Analog subtractive synthesis effect of ‘Hard Synchronisation.’ While the original effect is produced by an explicit waveform phase reset, other approaches are given that produce an equivalent output. In particular, based on measurements taken from a real-analog synthesizer, a comb filtering model is proposed. This description ties in with earlier work but here an explicit structure is provided. This filter-based approach is then shown to be far more computationally efficient than the synchronisation by phase reset. This efficiency is at a minor cost as it is shown that it has a minimal impact on the sonic accuracy

Aliasing reduction in clipped signals

Most real-world audio devices, particularly those of interest in musical applications, fall under the category of nonlinear systems. Examples of these devices include overdrive and distortion circuits used by guitar and bass players, dynamic range processors, and vintage synthesizer circuits. Nonlinear algorithms are known to expand the bandwidth of the input signal by introducing harmonic and intermodulation distortion. Naive digital emulations of these systems are susceptible to aliasing due to the inherent frequency constraints of discrete systems.  This thesis focuses on new digital signal processing techniques designed to reduce the level of aliasing introduced by memoryless nonlinearities. The underlying motivation of this work is to incorporate these tools within the framework of virtual analog (VA) modeling, an area of study that concentrates on the emulation of analog audio devices in the digital domain. In VA modeling, aliasing reduction has been studied extensive for the case of synthesis of classical oscillator waveforms like those used in subtractive synthesis. However, in audio effects processing oversampling has traditionally been the only available tool to ameliorate this problem. The first part of this work proposes the use of bandlimited correction functions previously used in waveform synthesis, to reduce the aliasing caused by special nonlinearities that introduce discontinuities in the derivatives of a signal. This family of novel methods includes the use of the bandlimited ramp function (BLAMP), its efficient polynomial approximations, and its integrated form. A new VA model of a highly nonlinear wavefolder circuit, which incorporates one of these techniques, is proposed.  The second family of techniques elaborated in this thesis is that of the antiderivative method. This innovative approach to aliasing reduction is based on the discrete differentiation of integrated nonlinearities and can be applied to arbitrary explicit memoryless nonlinearities regardless of their form. The use of the antiderivative forms in VA modeling is proposed by introducing two novel transistor/diode-based wavefolder models, and two static diode clipper models that incorporate these techniques.  Results obtained show the proposed algorithms effectively reduce the level of aliasing in nonlinear processing and can help reduce, and in some cases even eliminate, the oversampling requirements of the system. The proposed algorithms are suitable for real-time software implementations of VA instruments and effects processors

Design of a Scalable Polyphony-MIDI Synthesizer for a Low Cost DSP

Tässä diplomityössä esitetään Scalable Polyphony-MIDI-standardin soitinvalikoiman toteuttavan musiikkisyntetisaattorin suunnittelu edulliselle signaaliprosessorille. Ensiksi esitellään SPMIDI- standardi ja käytettävä signaaliprosessori. Sen jälkeen kerrataan yleisesti käytössä olevia synteesitekniikoita, ja niiden soveltuvuutta järjestelmiin, joissa laskentateho ja muistin määrä ovat rajoittuneita. Seuraavaksi käydään yksityiskohtaisesti läpi vähentävässä synteesissä käytettäviä oskillaattori- ja suodintekniikoita. Erityistä huomiota kiinnitetään laskostumiseen, joka johtuu perinteisissä aaltomuodoissa, kuten sahalaita- ja pulssi-aallossa, olevista epäjatkuvuuksista, ja olemassa olevia kaistarajoitettuja aaltomuotosynteesimenetelmiä kerrataan. Tämän jälkeen kerrataan olemassaolevia rakenteita laskennallisesti tehokkaille aikavarianteille suotimille. Kokonaan uusi, rajataajuuden ja resonanssin ohjauksen erottava, suodinrakenne esitellään. Analogiseen Korg MS-20-alipäästösuotimeen perustuva rakenne on laskennallisesti erittäin tehokas ja soveltuu hyvin toteutettavaksi vähäbittisillä arkkitehtuureilla. Lopuksi käsitellään toteutukseen liittyviä yksityiskohtia kiinnittäen erityistä huomiota differentioituun paraabeliaaltoon ja MS-20-suotimeen ja rajoitetun laskentakapasiteetin ja laskentaresoluution vaikutuksiin. Tämän jälkeen esitetään joidenkin esimerkkisoitinten toteutus.In this thesis, the design of a music synthesizer implementing the Scalable Polyphony-MIDI soundset on a low cost DSP system is presented. First, the SP-MIDI standard and the target DSP platform are presented followed by review of commonly used synthesis techniques and their applicability to systems with limited computational and memory resources. Next, various oscillator and filter algorithms used in digital subtractive synthesis are reviewed in detail. Special attention is given to the aliasing problem caused by discontinuities in classical waveforms, such as sawtooth and pulse waves and existing methods for bandlimited waveform synthesis are presented. This is followed by review of established structures for computationally efficient time-varying filters. A novel digital structure is presented that decouples the cutoff and resonance controls. The new structure is based on the analog Korg MS-20 lowpass filter and is computationally very efficient and well suited for implementation on low bitdepth architectures. Finally, implementation issues are discussed with emphasis on the Differentiated Parabole Wave oscillator and MS-20 filter structures and the effects of limited computational capability and low bitdepth. This is followed by designs for several example instruments