Download A pickup model for the Clavinet In this paper recent findings on magnetic transducers are applied to the analysis and modeling of Clavinet pickups. The Clavinet is a stringed instrument having similarities to the electric guitar, it has magnetic single coil pickups used to transduce the string vibration to an electrical quantity. Data gathered during physical inspection and electrical measurements are used to build a complete model which accounts for nonlinearities in the magnetic flux. The model is inserted in a Digital Waveguide (DWG) model for the Clavinet string for its evaluation.
Download 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.
Download Higher-Order Integrated Wavetable Synthesis Wavetable synthesis is a popular sound synthesis method enabling the efficient creation of musical sounds. Using sample rate conversion techniques, arbitrary musical pitches can be generated from one wavetable or from a small set of wavetables: downsampling is used for raising the pitch and upsampling for lowering it. A challenge when changing the pitch of a sampled waveform is to avoid disturbing aliasing artifacts. Besides bandlimited resampling algorithms, the use of an integrated wavetable and a differentiation of the output signal has been proposed previously by Geiger. This paper extends Geiger’s method by using several integrator and differentiator stages to improve alias-reduction. The waveform is integrated multiple times before it is stored in a wavetable. During playback, a sample rate conversion method is first applied and the output signal is then differentiated as many times as the wavetable has been integrated. The computational cost of the proposed technique is independent of the pitch-shift ratio. It is shown that the higher-order integrated wavetable technique reduces aliasing more than the first-order technique with a minor increase in computational cost. Quantization effects are analyzed and are shown to become notable at high frequencies, when several integration and differentiation stages are used.
Download The Helmholtz Resonator Tree The Helmholtz resonator is a prototype of a single acoustic resonance, which can be modeled with a digital resonator. This paper extends this concept by coupling several Helmholtz resonators. The resulting structure is called a Helmholtz resonator tree. The height of the tree is defined by the number of resonator layers that are interconnected. The overall number of resonance frequencies of a Helmholtz resonator tree is the same as its height. A Helmholtz resonator tree can be modeled using wave digital filters (WDF), when electro-acoustic analogies are applied. A WDF tool for implementing Helmholtz resonator trees has been developed in C++. A VST plugin and an Android mobile application were created, which can run short Helmholtz resonator trees in real time. Helmholtz resonator trees can be used for the real-time synthesis of percussive sounds and for realizing novel filtering which can be tuned using intuitive physical parameters.
Download Physically Informed Synthesis of Jackhammer Tool Impact Sounds This paper introduces a sound synthesis method for jackhammer tool impact sounds. The model is based on parallel waveguide models for longitudinal and transversal vibrations. The longitudinal sounds are produced using a comb filter that is tuned to match the longitudinal resonances of a steel bar. The dispersive transversal vibrations are produced using a comb filter which has a cascade of first-order allpass filters and time-varying feedback coefficient. The synthesis model is driven by an input generator unit that produces a train of Hann pulses at predetermined time-intervals. Each pulse has its amplitude modified slightly by a random process. For increased realism each impact is followed by a number of repetitive impacts with variable amplitude and time difference according to the initial pulse. The sound output of the model is realized by mixing both transversal and longitudinal signals and the effect is finalized by an equalizer.
Download A New Reverberator based on Variable Sparsity Convolution An efficient algorithm approximating the late part of room reverberation is proposed. The algorithm partitions the impulse response tail into variable-length segments and replaces them with a set of sparse FIR filters and lowpass filters, cascaded with several Schroeder allpass filters. The sparse FIR filter coefficients are selected from a velvet noise sequence, which consists of ones, minus ones, and zeros only. In this application, it is sufficient perceptually to use very sparse velvet noise sequences having only about 0.1 to 0.2% non-zero elements, with increasing sparsity along the impulse response. The algorithm yields a parametric approximation of the late part of the impulse response, which is more than 100 times more efficient computationally than the direct convolution. The computational load of the proposed algorithm is comparable to that of FFT-based partitioned convolution techniques, but with nearly half the memory usage. The main advantage of the new reverberator is the flexible parameterization.
Download Perceptual Linear Filters: Low-Order ARMA Approximation for Sound Synthesis This paper deals with the approximation of a given frequency response by a low-order linear ARMA filter (Auto-Regressive Moving Average). The aim of this work is the audio synthesis, then to improve the perceptual quality, a criterion based on human listening is defined and minimized. Two complementary approaches are proposed here for solving this non-linear and non-convex problem: first, a weighted version of the Iterative Prefiltering, second, an adaptation of the Gauss-Newton method. This algorithm is adapted to guarantee the causality/stability of the obtained filter, and eventually its minimum phase property. The benefit of the new method is illustrated and evaluated.
Download 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.
Download Granular analysis/synthesis of percussive drilling sounds This paper deals with the automatic and robust analysis, and the realistic and low-cost synthesis of percussive drilling like sounds. The two contributions are: a non-supervised removal of quasistationary background noise based on the Non-negative Matrix Factorization, and a granular method for analysis/synthesis of this drilling sounds. These two points are appropriate to the acoustical properties of percussive drilling sounds, and can be extended to other sounds with similar characteristics. The context of this work is the training of operators of working machines using simulators. Additionally, an implementation is explained.
Download Barberpole Phasing and Flanging Illusions 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 method, called a synchronized dual flanger, realizes the desired effect in an innovative and economic way using two cascaded time-varying comb filters and cross-fading between them. The third method is based on the use of single-sideband modulation, also known as frequency shifting. The proposed techniques effectively reproduce the illusion of endlessly moving spectral notches, particularly at slow modulation speeds and for input signals with a rich frequency spectrum. These effects can be programmed in real time and implemented as part of a digital audio processing system.