This paper presents a survey of various audio effects that can be physically applied to a rigidly-terminated vibrating string. The string’s resonant behavior is described, and then the ability of active feedback control to “reprogram" the physics of the string is explained. Active damping, which is a direct result of applying classical control techniques, provides for an effect based on amplitude modulation (AM). Traditional electric guitar sustain techniques are elaborated upon, which suggest another approach for ensuring marginal stability of the system even in the presence of an arbitrary nonlinear and/or time-varying effect unit in the feedback loop. This approach involves placing a dynamic range limiter in the feedback loop and does not introduce significant harmonic distortion other than that due to the effect unit. The maximum RMS level of the system’s output can be easily bounded if reasonable conditions are met by the dynamic range limiter. Finally, nonlinear and time-varying feedback control loops are applied experimentally to artificially induce frequency modulation (FM) at a low rate and AM at a high rate. These effects can be interpreted musically as vibrato and as a sort of resonant ring modulation, respectively.

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Table lookup oscillators form a basic building block of most software synthesis systems. Several of the classical digital and analog synthesis techniques require their use. The classical table lookup oscillator [1, 2] is commonly discussed regarding its amplitude error, but another source of error is mostly disregarded, although it influences the sound quality to a much higher degree, especially when running the oscillators at elevated frequencies. This error is due to the aliasing effect that occurs when shifting (or resampling) the original lookup table at higher rates. One obstacle that has been encountered when trying to emulate classical analog synthesizers, is that the naive implementation of waveforms for subtractive synthesis generate the same aliasing frequencies. Over the years, several solutions for the problem of generating high quality band-limited waveforms for emulating classical analog synthesis have been proposed, solving the problem for specific waveforms. Never-the-less, the general question of how to manage aliasing behavior efficiently in general table lookup oscillators is still not resolved. This article reviews existing techniques for analog synthesis and tries to expand them for the general case.

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In this article we propose a method to improve the accuracy of sinusoid modeling by introducing parameter variation models into both the analyzer and the synthesizer. Using the least-square-error estimator as an example, we show how the sinusoidal parameters estimated under a stationary assumption relate to the real nonstationary process, and propose a way to reestimate the parameters using some parameter variation model. For the synthesizer, we interpolate the parameters using the same model, with the phase unwrapping process reformulated to adapt to the change. Results show that the method effectively cuts down the systematic error of a conventional system based on a least-square-error estimator and the McAulay-Quatieri synthesizer.

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This paper presents an implementation of the phase vocoder within a Gaussian state-space framework. Rather than formulate the problem as a deterministic evolution of frequencies centered around a given bin, this evolution is treated stochastically by introducing noise into the dynamics matrix of the recursive state equation. This produces effects on the roughness of the input sound, which vary depending on the position within the matrix where the noise is added, how it is propagated throughout the matrix and further by the variance of the noise input.

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This paper presents a new fast sound synthesis method using polynomials. This is an additive method, where polynomials are used to approximate sine functions. Traditional additive synthesis requires each sample to be generated for each partial oscillator. Then all these partial samples are summed up to obtain the resulting sound sample, thus making the synthesis time proportional to the product of the number of oscillators and the sampling rate. By using polynomial approximations, we instead sum up only the oscillator coefficients and then generate directly the sound sample from these new coefficients. Most of computation time is consumed by a data structure that manages the update of the generator coefficients as a priority queue. Practical implementations show that Polynomial Additive Sound Synthesis (PASS) is particularly efficient for low-frequency signals.

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The circle map is a general non-linear iterated function that maps the circle onto itself. In its standard form it can be interpreted as a simple sinusoidal oscillator which is perturbed by a non-linear term. By varying the strength of the non-linear contribution a rich array of non-linear responses can be achieved, including waveshaping, pitch-bending, period-doubling and highly irregular patterns. We describe a number of such examples and discuss their subjective auditory perception.

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This paper discusses the implementation of a complex event generation model with a simple feedback loop and its sound synthesis results while investigating the overall system behaviour. The system is based on the Cosmos model, which is a self similar structure, and distributes events on different time-scales with certain interdependency. The user intervenes with the system in real-time by inputting a live sound source and interacting with the user interface by controlling the parameters for the time scales macro, meso and micro. Because of the complex dynamic behaviour and modulation scheme, it is possible to create a timbre space of unique textures.

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Standard acoustic source localization algorithms attempt to estimate the instantaneous location of a source based only on current data from a microphone sensor array. This is done regardless of previous location estimates. However more recent Sequential Monte Carlo based approaches have instead posed the problem using an evolving state-space framework. In this paper we take this approach further by exploiting the directionality of human speech sources. This allows us to estimate the orientation of the source within the room. Finally combining previous source localization methods with this work we outline how both parameters - location and orientation - may be estimated jointly. Examples are given of performance in a typically reverberant real office environment for both a stationary and a moving source.

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Wave Field Synthesis (WFS) is a physical based sound reproduction technique. It relies on linear arrays of regularly spaced omnidirectional loudspeakers. A fundamental limitation of WFS is that the synthesis remains correct only up to a corner frequency referred to as spatial aliasing frequency. This paper addresses irregular spacing of loudspeaker array for WFS. Adapted driving functions are defined. New formulations of the spatial aliasing frequency are proposed. It is shown that the use of logarithmically spaced loudspeaker arrays can significantly increase the spatial aliasing frequency for non focused virtual sources.

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A novel analysis method for binaural room impulse responses (BRIRs) is presented. It is based on the analysis of ear canal signals with continuous wavelet transform (CWT). Then, the crosswavelet transform (XWT) is used for detection of the direct sound and individual reflections from a BRIR. The new method seems to time-localize the reflections quite accurately. In addition, the proposed analysis method enables detailed study of the frequency content of the early reflections. The algorithm is tested with both measured and modeled impulse responses. A comparison with an FFT-based cross-spectrogram is made. The results show that XWT has potential in audio signal analysis.

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