Download Time-Varying Filters for Musical Applications
A variety of methods are available for implementing time-varying digital filters for musical applications. The considerations for musical applications differ from those of other applications, such as speech coding. This domain requires realtime parametric control of a filter such as an equalizer, allowing parameters to vary each sample, e.g. by user interaction, a low-frequency oscillator (LFO), or an envelope. It is desirable to find a filter structure that is timevarying stable, artifact-free, computationally efficient, easily supports arbitrary filter shapes, and yields sensible intermediate filter shapes when interpolating coefficients. It is proposed to use the state variable filter (SVF) for this purpose. A novel proof of its stable time-varying behavior is presented. Equations are derived for matching common equalizer filter shapes, as well as any zdomain transfer function, making the SVF suitable for efficiently implementing any recursive filter. The SVF is compared to state of the art filter structures in an objective evaluation and a subjective listening test. The results confirm that the SVF has good audio quality, while supporting the aforementioned advantageous qualities in a time-varying digital filter for music. They also show that a class of time-varying filter techniques useful for speech coding are unsuitable for musical applications.
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 Approximations for Online Computation of Redressed Frequency Warped Vocoders
In recent work, the construction of non-uniform generalized Gabor frames for the time-frequency analysis of signals has been introduced. In particular, while preserving perfect reconstruction, these frames allow for tilings of the time-frequency plane with arbitrary allocation of partially overlapping frequency bands or time intervals. In a recent paper, the author demonstrated that the construction of such frames can be entirely based on warping operators, which are specified by the required frequency or time warping maps, which, in turn, interpolate the desired frequency or time intervals edges. However, while the online computation of Gabor expansions on non-uniform time intervals presents little or no problem, the computation of Gabor expansions on non-uniform frequency bands requires knowledge of the Fourier transform of the entire signal, which precludes online computation. In this paper we introduce approximations and ideas for the design of nearly perfect reconstruction analysis and synthesis atoms, which allow for the online computation of time-frequency representations on non-uniform frequency bands.
Download Hybrid Reverberation Processor with Perceptual Control
This paper presents a hybrid reverberation processor, i.e. a realtime audio signal processing unit that combines a convolution reverb for recreating the early reflections of a measured impulse response (IR) with a feedback delay network (FDN) for synthesizing the reverberation tail. The FDN is automatically adjusted so as to match the energy decay profile of the measured IR. Particular attention is given to the transition between the convolution section and the FDN in order to avoid audible artifacts. The proposed reverberation processor offers both computational efficiency and flexible perceptual control over the reverberation effect.
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 Multi-Player Microtiming Humanisation using a Multivariate Markov Model
In this paper, we present a model for the modulation of multiperformer microtiming variation in musical groups. This is done using a multivariate Markov model, in which the relationship between players is modelled using an interdependence matrix (α) and a multidimensional state transition matrix (S). This method allows us to generate more natural sounding musical sequences due to the reduction of out-of-phase errors that occur in Gaussian pseudorandom and player-independent probabilistic models. We verify this using subjective listening tests, where we demonstrate that our multivariate model is able to outperform commonly used univariate models at producing human-like microtiming variability. Whilst the participants in our study judged the real time sequences performed by humans to be more natural than the proposed model, we were still able to achieve a mean score of 63.39% naturalness, suggesting microtiming interdependence between players captured in our model significantly enhances the humanisation of group musical sequences.
Download Streaming Spectral Processing with Consumer-Level Graphics Processing Units
This paper describes the implementation of a streaming spectral processing system for realtime audio in a consumer-level onboard GPU (Graphics Processing Unit) attached to an off-the-shelf laptop computer. It explores the implementation of four processes: standard phase vocoder analysis and synthesis, additive synthesis and the sliding phase vocoder. These were developed under the CUDA development environment as plugins for the Csound 6 audio programming language. Following a detailed exposition of the GPU code, results of performance tests are discussed for each algorithm. They demonstrate that such a system is capable of realtime audio, even under the restrictions imposed by a limited GPU capability.
Download A Two Level Montage Approach to Sound Texture Synthesis with Treatment of Unique Events
In this paper a novel algorithm for sound texture synthesis is presented. The goal of this algorithm is to produce new examples of a given sampled texture, the synthesized textures being of any desired duration. The algorithm is based on a montage approach to synthesis in that the synthesized texture is made up of pieces of the original sample concatenated together in a new sequence. This montage approach preserves both the high level evolution and low level detail of the original texture. Included in the algorithm is a measure of uniqueness, which can be used for the identification of regions in the original texture containing events that are atypical of the texture, and hence avoid their unnatural repetition at the synthesis stage.
Download Fast Signal Reconstruction from Magnitude Spectrogram of Continuous Wavelet Transform Based on Spectrogram Consistency
The continuous wavelet transform (CWT) can be seen as a filterbank having logarithmic frequency subbands spacing similar to the human auditory system. Thus, to make computers imitate the significant functions of the human auditory system, one promising approach would be to model, analyze and process magnitude spectrograms given by the CWT. To realize this approach, we must be able to convert a processed or modified magnitude CWT spectrogram, which contains no information about the phase, into a time domain signal specifically for those applications in which the aim is to generate audio signals. To this end, this paper proposes a fast algorithm for estimating the phase from a given magnitude CWT spectrogram to reconstruct an audio signal. The experimental results revealed that the proposed algorithm was around 100 times faster than a conventional algorithm, while the reconstructed signals obtained with the proposed algorithm had almost the same audio quality as those obtained with the previous study.
Download Numerical Simulation of String/Barrier Collisions: The Fretboard
Collisions play a major role in various models of musical instruments; one particularly interesting case is that of the guitar fretboard, the subject of this paper. Here, the string is modelled including effects of tension modulation, and the distributed collision both with the fretboard and individual frets, and including both effects of free string vibration, and under finger-stopped conditions, requiring an additional collision model. In order to handle multiple distributed nonlinearities simultaneously, a finite difference time domain method is developed, with a penalty potential allowing for a convenient model of collision within a Hamiltonian framework, allowing for the construction of stable energy-conserving methods. Implementation details are discussed, and simulation results are presented illustrating a variety of features of such a model.