Paper Archive

We show a new sufficient criterion for time-varying digital filter stability: that the matrix norm of the product of state matrices over a certain finite number of time steps is bounded by 1. This extends Laroche’s Criterion 1, which only considered one time step, while hinting at extensions to two time steps. Further extending these results, we also show that there is no intrinsic requirement that filter coefficients be frozen over any time scale, and extend to any dimension a helpful theorem that allows us to avoid explicitly performing eigen- or singular value decompositions in studying the matrix norm. We give a number of case studies on filters known to be time-varying stable, that cannot be proven time-varying stable with the original criterion, where the new criterion succeeds.

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In this article we investigate how the local Gaussian model (LGM) can be applied to separate sound sources in the higher-order ambisonics (HOA) domain. First, we show that in the HOA domain, the mathematical formalism of the local Gaussian model remains the same as in the microphone domain. Second, using an off-the shelf source separation toolbox (FASST) based on the local Gaussian model, we validate the efficiency of the approach in the HOA domain by comparing the performance of toolbox in the HOA domain with its performance in the microphone domain. To do this we discuss and run some simulations to ensure a fair comparison. Third, we check the efficiency of the local Gaussian model compared to other available source separation techniques in the HOA domain. Simulation results show that separating sources in the HOA domain results in a 1 to 12 dB increase in signal-to-distortion ratio, compared to the microphone domain. Multichannel source separation, local Gaussian model, Wiener filtering, 3D audio, Higher Order Ambisonics (HOA).

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Sound matching allows users to automatically approximate existing sounds using a synthesizer. Previous work has mostly focused on algorithms for automatically programming an existing synthesizer. This paper proposes a system for selecting between different synthesizer designs, each one with a corresponding automatic programmer. An implementation that allows designing ensembles based on a template is demonstrated. Several experiments are presented using a simple subtractive synthesis design. Using an ensemble of synthesizer-programmer pairs is shown to provide better matching than a single programmer trained for an equivalent integrated synthesizer. Scaling to hundreds of synthesizers is shown to improve match quality.

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Modern digital sound formats such as aiff, mpeg1/2/4/7, wav and ra support the use of cue points. A cue point may also be referred to as a seek point or a key frame. These mechanisms store meta data about the sound files. In this paper, we identity and describe how these formats are encoded, and process meta data information with a focus on cue points. Finally, we conclude with the direction of our future research for improving multimedia browsing mechanisms and additional applications by leveraging the use of cue points within those applications. Keywords: sound file formats, cue points, sound file, audio files, seek point, key frame, audio indexing

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This work presents a novel framework for music synthesis, based on the perceptual structure analysis of pre-existing musical signals, for example taken from a personal MP3 database. We raise the important issue of grounding music analysis on perception, and propose a bottom-up approach to music analysis, as well as modeling, and synthesis. A model of segmentation for polyphonic signals is described, and is qualitatively validated through several artifact-free music resynthesis experiments, e.g., reversing the ordering of sound events (notes), without reversing their waveforms. Then, a compact “timbre” structure analysis, and a method for song description in the form of an “audio DNA” sequence is presented. Finally, we propose novel applications, such as music cross-synthesis, or time-domain audio compression, enabled through simple sound similarity measures, and clustering.

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In ensemble music performance, such as a string quartet or duet, the musicians interact and influence each other’s performance via a multitude of parameters – including tempo, dynamics, articulation of musical phrases and, depending on the type of instrument, intonation. This paper presents our ongoing research on the effect of interaction between violinists, in terms of intonation. We base our analysis on a series of experiments with professional as well as amateur musicians playing in duet and solo experimental set-ups, and then apply a series of interdependence measures on each violinist’s pitch deviations from the score. Our results show that while it is possible to, solely based on intonation, distinguish between solo and duet performances for simple cases, there is a multitude of underlying factors that need to be analyzed before these techniques can be applied to more complex pieces and/or non-experimental situations.

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In this paper, we propose the Modulation Scale Spectrum as an extension of the Modulation Spectrum through the Scale domain. The Modulation Spectrum expresses the evolution over time of the amplitude content of various frequency bands by a second Fourier Transform. While its use has been proven for many applications, it is not scale-invariant. Because of this, we propose the use of the Scale Transform instead of the second Fourier Transform. The Scale Transform is a special case of the Mellin Transform. Among its properties is "scale-invariance". This implies that two timestretched version of a same music track will have (almost) the same Scale Spectrum. Our proposed Modulation Scale Spectrum therefore inherits from this property while describing frequency content evolution over time. We then propose a specific implementation of the Modulation Scale Spectrum in order to represent rhythm content. This representation is therefore tempo-independent. We evaluate the ability of this representation to catch rhythm characteristics on a classification task. We demonstrate that for this task our proposed representation largely exceeds results obtained so far while being highly tempo-independent.

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A FPGA-based system suitable for augmented reality audio applications is presented. The sample application described here is adaptive noise cancellation (ANC). The system consists of a Spartan -3 FPGA XC3S400 board connected to a Philips Stereo-AudioCodec UCB 1400. The algorithms for the FIR filtering and for the adaption of the filter coefficients according to the Widrow-Hoff LMS algorithm are implemented on the FPGA board. Measurement results obtained with a dummy head measuring system are reported, and a detailed analysis of system performance and possible system improvements is given.

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Digital simulation of guitar tube amplifiers is still an opened topic. The efficient implementation of several parts of the guitar amplifier is presented in this paper. This implementation is based on the pre-computation of the solution of the nonlinear differential system and further approximation of the solution. It reduces the computational complexity while the accuracy is comparable with the numerical solution. The method is used for simulation of different parts of the guitar amplifier, namely a triode preamp stage, a phase splitter and a push-pull amplifier. Finally, the results and comparison with other methods are discussed.

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There are several algorithms and approaches to Room Impulse Response (RIR) estimation. To the best of the author’s knowledge, there is no documentation of accuracy, speed, or even the feasibility of using signed distance functions (SDFs) in combination with sphere tracing for this task. A proof of concept with a focus on real-time performance is presented here, which still lacks many features such as frequency-dependent absorption and scattering coefficients, arbitrary source and receiver directives etc. The results are then compared to real room impulse responses and to a different simulation algorithm. Also, the rather special merits of such an approach, such as 4D reverberation and simple rounding of geometry, are briefly discussed and presented.

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