Paper Archive

This paper describes the implementation of an innovative musical interface based on the sound localization capability of a microphone array. Our proposal is to allow a musician to plan and conduct the expressivity of a performance, by controlling in realtime an audio processing module through the spatial movement of a sound source, i.e. voice, traditional musical instruments, sounding mobile devices. The proposed interface is able to locate and track the sound in a two-dimensional space with accuracy, so that the x-y coordinates of the sound source can be used to control the processing parameters. In particular, the paper is focused on the localization and tracking of harmonic sound sources in real moderate reverberant and noisy environment. To this purpose, we designed a system based on adaptive parameterized Generalized Cross-Correlation (GCC) and Phase Transform (PHAT) weighting with Zero-Crossing Rate (ZCR) threshold, a Wiener filter to improve the Signal to Noise Ratio (SNR) and a Kalman filter to make the position estimation more robust and accurate. We developed a Max/MSP external objects to test the system in a real scenario and to validate its usability.

Download

In the digital simulation of non-linear audio effect circuits, the arising non-linear equation system generally poses the main challenge for a computationally cheap implementation. As the computational complexity grows super-linearly with the number of equations, it is beneficial to decompose the equation system into several smaller systems, if possible. In this paper we therefore develop an approach to determine such a decomposition automatically. We limit ourselves to cases where an exact decomposition is possible, however, and do not consider approximate decompositions.

Download

This paper investigates the passivity of the Moog Ladder Filter and its simulation. First, the linearized system is analyzed. Results based on the energy stored in the capacitors lead to a stability domain which is available for time-varying control parameters meanwhile it is sub-optimal for time-invariant ones. A second storage function is proposed, from which the largest stability domain is recovered for a time-invariant Q-parameter. Sufficient conditions for stability are given. Second, the study is adapted to the nonlinear case by introducing a third storage function. Then, a simulation based on the standard bilinear transform is derived and the dissipativity of this numerical version is examined. Simulations show that passivity is not unconditionally guaranteed, but mostly fulfilled, and that typical behaviours of the Moog filter, including self-oscillations, are properly reproduced.

Download

This paper describes a sound synthesis technique that modulates the coefficients of allpass filter chains using audio-rate frequencies. It was found that modulating a single allpass filter section produces a feedback AM–like spectrum, and that its bandwidth is extended and further processed by non-sinusoidal FM when the sections are cascaded. The cascade length parameter provides dynamic bandwidth control to prevent upper range aliasing artifacts, and the amount of spectral content within that band can be controlled using a modulation index parameter. The technique is capable of synthesizing rich and evolving timbres, including those resembling classic virtual analog waveforms. It can also be used as an audio effect with pitch-tracked input sources. Software and sound examples are available at http://www.acoustics.hut.fi/publications/papers/dafx09-cm/

Download

Virtual acoustics and auralizations have been previously used to study the perceptual properties of concert hall acoustics in a descriptive profiling framework. The results have indicated that the apparent auditory distance to the orchestra might play a crucial role in enhancing the listening experience and the appraisal of hall acoustics. However, it is unknown how the acoustics of the hall influence auditory distance perception in such large spaces. Here, we present one step towards studying auditory distance perception in concert halls with virtual acoustics. The aims of this investigation were to evaluate the feasibility of the auralizations and the system to study perceived distances as well as to obtain first evidence on the effects of hall acoustics and the source materials to distance perception. Auralizations were made from measured spatial impulse responses in two concert halls at 14 and 22 meter distances from the center of a calibrated loudspeaker orchestra on stage. Anechoic source materials included symphonic music and pink noise as well as signals produced by concatenating random segments of anechoic instrument recordings. Forty naive test subjects were blindfolded before entering the listening room, where they verbally reported distances to sound sources in the auralizations. Despite the large variance in distance judgments between the individuals, the reported distances were on average in the same range as the actual distances. The results show significant main effects of halls, distances and signals, but also some unexpected effects associated with the presentation order of the stimuli.

Download

This paper will describe a toolkit for experimentation with signal interaction techniques, also commonly referred to as cross adaptive processing. The technique allows analyzed features of one audio signal to inform the processing of another. Earlier used mainly for mixing and post production purposes, we now want to use it creatively as an intervention in the musical communication between two performers. The idea stems from Stockhausen’s use of intermodulation in the 1960’s, and as such we might also call the updated technique interprocessing. Our interest in the technique comes as a natural extension to previous research on live processing as an instrumental and performative activity. The automatic control of signal processing routines is related to previous work on adaptive audio effects and automatic mixing. The focus for our investigation and experimentation with the current toolkit will be how this affects the musical communication between performers, and how it changes what they can and will play. The program code for the toolkit is available as a github repository1 under an open source license.

Download

Audio feature extraction underpins a massive proportion of audio processing, music information retrieval, audio effect design and audio synthesis. Design, analysis, synthesis and evaluation often rely on audio features, but there are a large and diverse range of feature extraction tools presented to the community. An evaluation of existing audio feature extraction libraries was undertaken. Ten libraries and toolboxes were evaluated with the Cranfield Model for evaluation of information retrieval systems, reviewing the coverage, effort, presentation and time lag of a system. Comparisons are undertaken of these tools and example use cases are presented as to when toolboxes are most suitable. This paper allows a software engineer or researcher to quickly and easily select a suitable audio feature extraction toolbox.

Download

Contemporary musicians commonly face the challenge of finding new, characteristic sounds that can make their compositions more distinct. They often resort to computers and algorithms, which can significantly aid in creative processes by generating unexpected material in controlled probabilistic processes. In particular, algorithms that present emergent behaviors, like genetic algorithms and cellular automata, have fostered a broad diversity of musical explorations. This article proposes an original technique for the computer-assisted creation and manipulation of sound textures. The technique uses Probabilistic Cellular Automata, which are yet seldom explored in the music domain, to blend two audio tracks into a third, different one. The proposed blending process works by dividing the source tracks into frequency bands and then associating each of the automaton’s cell to a frequency band. Only one source, chosen by the cell’s state, is active within each band. The resulting track has a non-repeating textural pattern that follows the changes in the Cellular Automata. This blending process allows the musician to choose the original material and the blend granularity, significantly changing the resulting blends. We demonstrate how to use the proposed blending process in sound design and its application in experimental and popular music.

Download

In this paper we present a novel algorithm for automatic analysis, transcription, and parameter extraction from isolated polyphonic guitar recordings. In addition to general score-related information such as note onset, duration, and pitch, instrumentspecific information such as the plucked string, the applied plucking and expression styles are retrieved automatically. For this purpose, we adapted several state-of-the-art approaches for onset and offset detection, multipitch estimation, string estimation, feature extraction, and multi-class classification. Furthermore we investigated a robust partial tracking algorithm with respect to inharmonicity, an extensive extraction of novel and known audio features as well as the exploitation of instrument-based knowledge in the form of plausability filtering to obtain more reliable prediction. Our system achieved very high accuracy values of 98 % for onset and offset detection as well as multipitch estimation. For the instrument-related parameters, the proposed algorithm also showed very good performance with accuracy values of 82 % for the string number, 93 % for the plucking style, and 83 % for the expression style. Index Terms - playing techniques, plucking style, expression style, multiple fundamental frequency estimation, string classification, fretboard position, fingering, electric guitar, inharmonicity coefficient, tablature

Download

Periodic or quasi-periodic low-frequency components (i.e. vibrato and tremolo) are present in steadystate portions of sustained instrumental sounds. If we are interested both in studying its expressive meaning, or in building a hierarchical multi-level representation of sound in order to manipulate it and transform it with musical purposes those components should be isolated and separated from the amplitude and frequency envelopes. Within the SMS analysis framework it is now feasible to extract high level time-evolving attributes starting from basic analysis data. In the case of frequency envelopes we can apply STFTs to them, then check if there is a prominent peak in the vibrato/tremolo range and, if it is true, we can smooth it away in the frequency domain; finally, we can apply an IFFT to each frame in order to re-construct an envelope that has been cleaned of those quasi-periodic low-frequency components. Two important problems nevertheless have to be tackled, and ways of overcoming them will be discussed in this paper: first, the periodicity of vibrato and tremolo, that is quite exact only when the performers are professional musicians; second: the interactions between formants and fundamental frequency trajectories, that blur the real tremolo component and difficult its analysis.

Download