Victor Lazzarini

The generation of natural-synthetic spectra by means of adaptive frequency modulation

Frequency-modulation (FM) synthesis is widely known as a computationally efficient method for synthesizing musically interesting timbres. However, it has suffered from neglect owing to the difficulty in creating natural-sounding spectra and mapping gestural input to synthesis parameters. Recently, a revival has occurred with the advent of adaptive audio-processing methods, and this work proposes a technique called adaptive FM synthesis. This article derives two novel ways by which an arbitrary input signal can be used to modulate a carrier. We show how phase modulation (PM) can be achieved first by using delay lines and then by heterodyning. By applying these techniques to real-world signals, it is possible to generate transitions between natural-sounding and synthesizerlike sounds. Examples are provided of the spectral consequences of adaptive FM synthesis using inputs of various acoustic instruments and a voice. An assessment of the timbral quality of synthesized sounds demonstrates its effectiveness.

New perspectives on distortion synthesis for virtual analog oscillators

��� The term “virtual analog” (VA) first appeared in the 1990s with the commercial introduction of digital synthesizer instruments that were intended to emulate the earlier analog subtractive synthesizers, such as those produced by, among others, Moog or Sequential Circuits (Smith 2008). In creating a “virtual analog” digital model, two approaches are possible: the first is to build an explicit digital model derived from the various electrical components that form the original analog circuit; the second is to use digital processing structures that will produce outputs that mimic those of the analog system. To date, the second approach has been used for the implementation of the elements of virtual analog subtractive synthesizers, as real-time computational constraints limit the efficacy of circuit models in usable software instruments (Civolani and Fontana 2008). Among the features of a subtractive synthesis system, one aspect that has received more attention than others is algorithms for real-time digital generation of the periodic waveforms associated with voltage-controlled oscillators (VCO), such as sawtooths, square waves, and triangle waves, also known as the classic analog waveforms (Stilson 2006).

Real-time detection of musical onsets with linear prediction and sinusoidal modeling

Real-time musical note onset detection plays a vital role in many audio analysis processes, such as score following, beat detection and various sound synthesis by analysis methods. This article provides a review of some of the most commonly used techniques for real-time onset detection. We suggest ways to improve these techniques by incorporating linear prediction as well as presenting a novel algorithm for real-time onset detection using sinusoidal modelling. We provide comprehensive results for both the detection accuracy and the computational performance of all of the described techniques, evaluated using Modal, our new open source library for musical onset detection, which comes with a free database of samples with hand-labelled note onsets.

Spectrally rich phase distortion sound synthesis using an allpass filter

This paper examines a recently introduced technique for sound synthesis that uses a coefficient modulated allpass filter to cause phase modifications to its input signal. The intention in this work is to outline some of the properties of the coefficient modulated allpass filter and then to establish a connection between this new method and the older technique of Phase Distortion. Results are presented to demonstrate how the allpass technique provides a spectrally richer output signal.

Recent Advances in Real-Time Musical Effects, Synthesis, and Virtual Analog Models

This paper reviews some of the recent advances in real-time musical effects processing and synthesis. The main emphasis is on virtual analog modeling, specifically digital emulation of vintage delay and reverberation effects, tube amplifiers, and voltage-controlled filters. Additionally, adaptive effects algorithms and sound synthesis and processing languages are discussed.

Relational Properties in Interaction Aesthetics: The Ubiquitous Music Turn

Taking as a point of departure recent theoretical advances in Interaction Design and Human-Computer Interaction (Lowgren 2009), we discuss a body of knowledge gathered in Ubiquitous Music practices (Keller et al. 2011a) during the last six years. New concepts and methods have been proposed to describe aspects of the ideation and materialisation of experiences with technology. Pliability (Lowgren 2007) and anchoring (Keller et al. 2010) are two of the multiple design qualities that surfaced in interaction design that impact information technology creative practices. We present results of experiments addressing creativity support for ubiquitous music making through the time tagging metaphor (Radanovitsck et al. 2011). The studies serve to exemplify how relational properties can be integrated within creativity-centred design. Our research indicates that sonic relational properties may provide affordances for proto-musical phenomena. We discuss the theoretical and methodological implications of this proposal highlighting its impact on everyday musical creativity.

Feedback Amplitude Modulation Synthesis

A recently rediscovered sound synthesis method, which is based on feedback amplitude modulation (FBAM), is investigated. The FBAM system is interpreted as a periodically linear time-varying digital filter, and its stability, aliasing, and scaling properties are considered. Several novel variations of the basic system are derived and analyzed. Separation of the input and the modulation signals in FBAM structures is proposed which helps to create modular sound synthesis and digital audio effects applications. The FBAM is shown to be a powerful and versatile sound synthesis principle, which has similarities to the established distortion synthesis methods, but which is also essentially different from them.

Discrete-time modelling of the moog sawtooth oscillator waveform

Discrete-time modelling strategies of analogue Moog sawtooth oscillator waveforms are presented. Two alternative approaches suitable for real-time implementation are proposed, one modelling the analogue waveform in time domain using phase distortion synthesis and another matching the spectrum of an existing antialiasing sawtooth oscillator to the corresponding analogue spectrum using a first-order IIR post-equalising filter. A parameter estimation procedure for both approaches is explained and performed. Performance evaluation using polynomial fits for the estimated parameters is carried out, and good matches between the model outputs and recorded waveforms are obtained. The best match of the tested algorithms is produced by the phase distortion model and by post-equalising the fourth-order B-spline bandlimited step function sawtooth oscillator.

The SndObj Sound Object Library

The SndObj Sound Object Library is a C++ object-oriented audio processing framework and toolkit. This article initially examines some of the currently available sound processing packages, including sound compilers, programming libraries and toolkits. It reviews the processes involved in the use of these systems and their strengths and weaknesses. Some application examples are also provided. In this context, the SndObj library is presented and its components are discussed in detail. The article shows the library as composed of a set of classes that encapsulate all processes involved in synthesis, processing and IO operations. Programming examples are included to show some uses of the system. These, together with library binaries, source code and complete documentation, are included in the downloadable package, available on the Internet. Possible future developments and applications are considered. The library is demonstrated to provide a useful base for research and development of audio processing software.

Real-time segmentation of the temporal evolution of musical sounds

Since the studies of Helmholtz, it has been known that the temporal evolution of musical sounds plays an important role in our perception of timbre. The accurate temporal segmentation of musical sounds into regions with distinct characteristics is therefore of interest to researchers in the field of timbre perception as well as to those working with different forms of sound modelling and manipulation. Following recent work by Hajda (1996), Peeters (2004) and Caetano et al (2010), this paper presents a new method for the automatic segmentation of the temporal evolution of isolated musical sounds in real-time. We define attack, sustain and release segments using cues from a combination of the amplitude envelope, the spectro- temporal evolution and a measurement of the stability of the sound that is derived from the onset detection function. We conclude with an evaluation of the method.