In this paper we present a descriptive study of perceived intensity in popular music. We have designed a taxonomy for the purpose, and created an intensity model based on low-level descriptors extracted from an objective data set, which can reliably predict a category label for the “subjective intensity ” of most popular music recordings. 1.

Modeling audio signals by the long-term statistical distribution of their local spectral features- often denoted as bag of frames approach (BOF)- is a popular and powerful method to describe audio content. While modeling the distribution of local spectral features by semi-parametric distributions (e.g. Gaussian Mixture Models) has been studied intensively, we investigate a non-parametric variant based on vector quantization (VQ) in this paper. The essential advantage of the proposed VQ approach over stateof-the-art similarity measures is that the proposed audio similarity metric forms a normed vector space. This allows for more powerful search strategies, e.g. KD-Trees or Local Sensitive Hashing (LSH), making content-based audio similarity available for even larger music archives. Standard VQ approaches are known to be computationally very expensive; to counter this problem, we propose a multi-level clustering architecture. Additionally, we show that the multi-level vector quantization approach (ML-VQ), in contrast to standard VQ approaches, is comparable to state-ofthe-art frame-level similarity measures in terms of quality. Another important finding w.r.t. the ML-VQ approach is that, in contrast to GMM models of songs, our approach does not seem to suffer from the recently discovered hub problem. 1.

The IM4Sports music system helps exercisers select music that suits their training programs, reflects and guides sport performance, and collects data for adapting training programs and music selections.

We describe a new approach to the “desoloing ” problem, in which one tries to isolate the accompanying instruments from a monaural recording of a soloist with accompaniment. Our approach is based on explicit knowledge of the audio in the form of a score match – a correspondence between a symbolic score and the music audio, giving the times of all musical events. We employ the familiar idea of masking the short time Fourier transform to eliminate the solo part. The ideal mask is estimated by fitting a model to the data, whose note-based components are derived from the score match. The parameters for our probabilistic model are estimated using the EM algorithm. 1

Swim Swan is a multimedia work for clarient, live electronics and computer graphics. The principles underlying the work are exposed and a set of sound-proccesing and synthesis algorithms that are used in this work are described. The issue of the control of the computer by the clarinet is emphasized. The artistic issues of such a work are addressed

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This paper presents musical and technological aspects of real-time digital audio processing and visual rendering applied to a grand nineteen-century pipe organ. The organ is “augmented ” in both its musical range and visual dimensions, thus increasing its potential for expression. The discussed project was presented to a public audience in the form of concerts. First, a brief project description is given, followed by in-depth discussions of the signal processing strategies and general musical considerations. Digital audio effects allow for the addition of new electronic registers to the organ stops. The “direct ” sound is captured inside the organ case close to the pipes in order to provide “dry ” audio signals for further processing. The room acoustic strongly affects the pipe organ sound perceived by the listener; hence, to combine the processed sound with the organ sound both room simulation and spatial audio rendering are applied. Consequently, the transformed sound is played back via a multitude of loudspeakers surrounding the audience. Finally, considerations of musical aspects are discussed, comprising reflections on virtuosity and technique in the musical play and how the new possibilities could affect composition practice and the use of the organ in contemporary music. 1.

We present a new group of audio effects that use beat tracking, the detection of beats in an audio signal, to relate effect parameters to the beats in an input signal. Conventional audio effects are augmented so that their operation is related to the output of a beat tracking system. We present a temposynchronous delay effect and a set of beat synchronous low frequency oscillator effects including tremolo, vibrato and auto-wah. All effects are implemented in real-time as VST plug-ins to allow for their use in live performance.

The ability of a sound synthesizer to provide realistic sounds depends to a great extent on the availability of expressive controls. One of the most important expressive features a user of the synthesizer would desire to have control of, is timbre. Timbre is a complex concept related to many musical indications in a score such as dynamics, accents, hand position, string played, or even indications referring timbre itself. Musical indications are in turn related to low level performance controls such as bow velocity or bow force. With the help of a data acquisition system able to record sound synchronized to performance controls and aligned to the performed score and by means of statistical analysis, we are able to model the interrelations among sound (timbre), controls and musical score indications. In this paper we present a procedure for score-controlled timbre transformations of violin sounds within a sample based synthesizer. Given a sound sample and its trajectory of performance controls: 1) a transformation of the controls trajectory is carried out according to the score indications, 2) a new timbre corresponding to the transformed trajectory is predicted by means of a timbre model that relates timbre with performance controls and 3) the timbre of the original sound is transformed by applying a timevarying filter calculated frame by frame as the difference of the original and predicted envelopes. 1.

This paper studies the reduction of nonlinearity of digital peak limiters used for maximizing signal levels. The goal is to control the time-varying gain smoothly enough to avoid frequency artifacts in the output signal. Smoother gain control is traditionally obtained by lowpass ltering the gain or the signal envelope. However, simple ltering causes overshoots and leads to either clipped output or non-maximal signal levels, depending on the gain applied to the limiter output. In order to obtain smooth gain control without clipping, this paper proposes an envelope detection method based on order-statistics ltering.