Download Improved adjustable boundary condition for the 2-D finite-difference time-domain mesh The digital waveguide (DWG) mesh is a method for simulating wave propagation in multiple dimensions. Boundary conditions are needed for modeling changes in wave propagation media such as walls and furniture in a room or boundaries of a resonating membrane of a musical instrument. The boundary conditions have been solved for a one-dimensional DWG structure, but there is no known exact solution for the multi-dimensional mesh. In this work, a new boundary structure is introduced for modeling reflection coefficient values −1 ≤ r ≤ 1 in two dimensions. The new method gives remarkably more accurate results than the earlier approximations, especially at the low absolute values of r. At incident angles of Θ < 60o , the absolute error of reflection coefficient r is below 0.1 at frequencies 0.004 < f < 0.222 relative to the sampling frequency and at 60o ≤ Θ ≤ 80o the same result is reached at 0.005 < f < 0.114.
Download Gestural exploitation of ecological information in continuous sonic feedback – The case of balancing a rolling ball Continuous sensory–motor loops form a topic dealt with rather rarely in experiments and applications of ecological auditory perception. Experiments with a tangible audio–visual interface around a physics-based sound synthesis core address this aspect. Initially dealing with the evaluation of a specific work of sound and interaction design, they deliver new arguments and notions for non-speech auditory display and are also to be seen in a wider context of psychoacoustic knowledge and methodology.
Download Improved method for extraction of partial’s parameters in polyphonic transcription of piano higher octaves Polyphonic transcription is specially challenging in piano higher octaves due to the complexity of the spectrum of notes and therefore, chords. Besides the fundamental and second partial components, other spectral elements appears. The three peaks related to the unison as well as the second harmonic of the fundamental unison can be distinguished in most measures. Furthermore, intermodulation components are also present when non-linearity is high enough. This paper compares several methods to improve the training process that allows to synthesize the spectral patterns and masks used in transcription methods.
Download Separation of overlapping impulsive sounds by bandwise noise interpolation The task of extracting harmonic content of multiple pitched sources from a mono audio mix has been investigated on several occasions [1, 2, 3, 4]. However, most pitched notes contain an inharmonic component, which is an important perceptual attribute of the sound. This content is usually not dealt with during separation. It would also be interesting in its own right to develop separation techniques for extracting percussive sounds for polyphonic mixes. This paper describes an attempt at separating overlapping impulsive content of multiple sources from a mono mix. The method uses an interpolation within individual frequency bands of the decaying noise envelope of each source across overlapping sections with other sources. Three analysis methods determining the distribution of these bands were tested: the DFT followed by processing in Bark bands, the discrete wavelet transform (DWT), and the dyadic wavelet packet transform (DWPT).
Download Intermodulation Effects Analysis using Complex Bandpass Filterbanks The objective of this paper is to show the ability of complex bandpass filterbanks to extract the intermodulation information that appears when two audio signals interact inside the same analysis band. To perform the analysis a sinusoidal model of the signals has been assumed. Three kinds of signals have been analyzed: a sum of two cosines, a sum of two linear chirps and a sum of two exponential chirps. The complex bandpass filtering of the signals is carried out using a new algorithm based on the Complex Continuous Wavelet Transform. The developed algorithm has been validated comparing the practical results with the theoretical instantaneous amplitude and instantaneous phase of the obtained model of the signals. With the appropriate width, the complex bandpass filters show the same behaviour as our perceptual ability to discriminate interacting tones when they fall inside a critical band of the human ear.
Download CMOS Implementation of an Adaptive Noise Canceller into a Subband Filter In recent years the demand for mobile communication has increased rapidly. While in the early years of mobile phones battery life was one of the main concerns for developers speech quality is now becoming one of the most important factors in the development of the next generation of mobile phones. This paper describes the CMOS implementation of an adaptive noise canceller (ANC) into a subband filter. The ANC-Subband filter is able to reduce noise components of real speech without prior knowledge of the noise properties. It is predestined to be used in mobile devices and therefore, uses a very low clock frequency resulting in a small power consumption. This low power consumption combined with its small physical size enables the circuit also be used in hearing aids to efficiently reduce noise contained in the speech signal.
Download GABOR, multi-representation real-time analysis/synthesis This article describes a set of modules for Max/MSP for real-time sound analysis and synthesis combining various models, representations and timing paradigms. Gabor provides a unified framework for granular synthesis, PSOLA, phase vocoder, additive synthesis and other STFT techniques. Gabor’s processing scheme allows for the treatment of atomic sound particles at arbitrary rates and instants. Gabor is based on FTM, an extension of Max/MSP, introducing complex data structures such as matrices and sequences to the Max data flow programming paradigm. Most of the signal processing operators of the Gabor modules handle vector and matrix representations closely related to SDIF sound description formats.
Download Morphing techniques for enhanced scat singing In jazz, scat singing is a phonetic improvisation that imitates instrumental sounds. In this paper, we propose a system that aims to transform singing voice into real instrument sounds, extending the possibilities for scat singers. Analysis algorithms in the spectral domain extract voice parameters, which drive the resulting instrument sound. A small database contains real instrument samples that have been spectrally analyzed offline. Two different prototypes are introduced, producing sounds of a trumpet and a bass guitar respectively.
Download Generalised Prior Subspace Analysis for Polyphonic Pitch Transcription A reformulation of Prior Subspace Analysis (PSA) is presented, which restates the problem as that of fitting an undercomplete signal dictionary to a spectrogram. Further, a generalization of PSA is derived which allows the transcription of polyphonic pitched instruments. This involves the translation of a single frequency prior subspace of a note to approximate other notes, overcoming the problem of needing a separate basis function for each note played by an instrument. Examples are then demonstrated which show the utility of the generalised PSA algorithm for the purposes of polyphonic pitch transcription.
Download Hidden Markov Models for spectral similarity of songs Hidden Markov Models (HMM) are compared to Gaussian Mixture Models (GMM) for describing spectral similarity of songs. Contrary to previous work we make a direct comparison based on the log-likelihood of songs given an HMM or GMM. Whereas the direct comparison of log-likelihoods clearly favors HMMs, this advantage in terms of modeling power does not allow for any gain in genre classification accuracy.