Konrad Kowalczyk
University of Erlangen-Nuremberg
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Featured researches published by Konrad Kowalczyk.
IEEE Transactions on Audio, Speech, and Language Processing | 2011
Konrad Kowalczyk; Maarten van Walstijn
This paper presents methods for simulating room acoustics using the finite-difference time-domain (FDTD) technique, focusing on boundary and medium modeling. A family of nonstaggered 3-D compact explicit FDTD schemes is analyzed in terms of stability, accuracy, and computational efficiency, and the most accurate and isotropic schemes based on a rectilinear grid are identified. A frequency-dependent boundary model that is consistent with locally reacting surface theory is also presented, in which the wall impedance is represented with a digital filter. For boundaries, accuracy in numerical reflection is analyzed and a stability proof is provided. The results indicate that the proposed 3-D interpolated wideband and isotropic schemes outperform directly related techniques based on Yees staggered grid and standard digital waveguide mesh, and that the boundary formulations generally have properties that are similar to that of the basic scheme used.
Journal of the Acoustical Society of America | 2012
Haohai Sun; Konrad Kowalczyk; Walter Kellermann
This paper presents an experimental and comparative study of several spherical microphone array eigenbeam (EB) processing techniques for localization of early reflections in room acoustic environments, which is a relevant research topic in both audio signal processing and room acoustics. This paper focuses on steered beamformer-based and subspace-based localization techniques implemented in the spherical EB domain, including the plane-wave decomposition, eigenbeam delay and sum, eigenbeam minimum variance distortionless response, eigenbeam multiple signal classification (EB-MUSIC), and eigenbeam estimation of signal parameters via rotational invariance techniques (EB-ESPRIT) methods. The directions of arrival of the original sound source and the associated reflection signals in acoustic environments are estimated from acoustic maps of the rooms, which are obtained using a spherical microphone array. The EB-domain-based frequency smoothing and white noise gain control techniques are derived and employed to improve the performance and robustness of reflection localization. The applicability of the presented methods in practice is confirmed by experiments carried out in real rooms.
Acta Acustica United With Acustica | 2008
Konrad Kowalczyk; Maarten van Walstijn
In this paper, we present new methods for constructing and analysing formulations of locally reacting surfaces that can be used in finite difference time domain (FDTD) simulations of acoustic spaces. Novel FDTD formulations of frequency-independent and simple frequency-dependent impedance boundaries are proposed for 2D and 3D acoustic systems, including a full treatment of corners and boundary edges. The proposed boundary formulations are designed for virtual acoustics applications using the standard leapfrog scheme based on a rectilinear grid, and apply to FDTD as well as Kirchhoff variable digital waveguide mesh (K-DWM) methods. In addition, new analytic evaluation methods that accurately predict the reflectance of numerical boundary formulations are proposed. numerical experiments and numerical boundary analysis (NBA) are analysed in time and frequency domains in terms of the pressure wave reflectance for different angles of incidence and various impedances. The results show that the proposed boundary formulations structurally adhere well to the theoretical reflectance. In particular, both reflectance magnitude and phase are closely approximated even at high angles of incidence and low impedances. Furthermore, excellent agreement was found between the numerical boundary analysis and the experimental results, validating both as tools for researching FDTD boundary formulations.
IEEE Transactions on Audio, Speech, and Language Processing | 2010
Konrad Kowalczyk; M. van Walstijn
In this paper, a complete method for finite-difference time-domain modeling of rooms in 2-D using compact explicit schemes is presented. A family of interpolated schemes using a rectilinear, nonstaggered grid is reviewed, and the most accurate and isotropic schemes are identified. Frequency-dependent boundaries are modeled using a digital impedance filter formulation that is consistent with locally reacting surface theory. A structurally stable and efficient boundary formulation is constructed by carefully combining the boundary condition with the interpolated scheme. An analytic prediction formula for the effective numerical reflectance is given, and a stability proof provided. The results indicate that the identified accurate and isotropic schemes are also very accurate in terms of numerical boundary reflectance, and outperform directly related methods such as Yees scheme and the standard digital waveguide mesh. In addition, one particular scheme-referred to here as the interpolated wideband scheme-is suggested as the best scheme for most applications.
IEEE Signal Processing Magazine | 2015
Konrad Kowalczyk; Oliver Thiergart; Maja Taseska; Giovanni Del Galdo; Ville Pulkki; Emanuel A. P. Habets
Flexible and efficient spatial sound acquisition and subsequent processing are of paramount importance in communication and assisted listening devices such as mobile phones, hearing aids, smart TVs, and emerging wearable devices (e.g., smart watches and glasses). In application scenarios where the number of sound sources quickly varies, sources move, and nonstationary noise and reverberation are commonly encountered, it remains a challenge to capture sounds in such a way that they can be reproduced with a high and invariable sound quality. In addition, the objective in terms of what needs to be captured, and how it should be reproduced, depends on the application and on the user?s preferences. Parametric spatial sound processing has been around for two decades and provides a flexible and efficient solution to capture, code, and transmit, as well as manipulate and reproduce spatial sounds.
IEEE Signal Processing Letters | 2013
Konrad Kowalczyk; Emanuel A. P. Habets; Walter Kellermann; Patrick A. Naylor
Localization of early room reflections can be achieved by estimating the time-differences-of-arrival (TDOAs) of reflected waves between elements of a microphone array. For an unknown source, we propose to apply sparse blind system identification (BSI) methods to identify the acoustic impulse responses, from which the TDOAs of temporally sparse reflections are estimated. The proposed time- and frequency-domain adaptive algorithms based on crossrelation formulation are regularized by incorporating an l1 -norm sparseness constraint, which is realized using a split Bregman method. These algorithms are shown to outperform standard crossrelation-based BSI techniques when estimating TDOAs of reflections in the presence of background noise.
international conference on acoustics, speech, and signal processing | 2011
Haohai Sun; Konrad Kowalczyk; Walter Kellermann
Methods of 3D direction of arrival (DOA) estimation, coherent source detection and reflective surface localization are studied, based on recordings by a spherical microphone array. First, the spherical harmonics domain minimum variance distortionless response (EB-MVDR) beamformer is employed for the localization of broadband coherent sources, which is characterized by simpler frequency focusing matrices than the corresponding element-space implementation, and by a higher resolution than conventional spherical array beamformers. After the DOA estimation step, the source signals are extracted by EB-MVDRs. Then, by computing the crosscorrelation functions between the extracted signals, the coherent sources are detected and their time differences of arrival (TDOA) are estimated. Given the positions of the array and the reference source, and the estimated DOA and TDOA of the coherent sources, the positions of the major reflectors can be inferred. Experimental results in a real room validate the proposed method.
IEEE Transactions on Audio, Speech, and Language Processing | 2011
Konrad Kowalczyk; Maarten van Walstijn; Damian T. Murphy
In this paper, a method for modeling diffusive boundaries in finite-difference time-domain (FDTD) room acoustics simulations with the use of impedance filters is presented. The proposed technique is based on the concept of phase grating diffusers, and realized by designing boundary impedance filters from normal-incidence reflection filters with added delay. These added delays, that correspond to the diffuser well depths, are varied across the boundary surface, and implemented using Thiran allpass filters. The proposed method for simulating sound scattering is suitable for modeling high frequency diffusion caused by small variations in surface roughness and, more generally, diffusers characterized by narrow wells with infinitely thin separators. This concept is also applicable to other wave-based modeling techniques. The approach is validated by comparing numerical results for Schroeder diffusers to measured data. In addition, it is proposed that irregular surfaces are modeled by shaping them with Brownian noise, giving good control over the sound scattering properties of the simulated boundary through two parameters, namely the spectral density exponent and the maximum well depth.
workshop on applications of signal processing to audio and acoustics | 2013
Konrad Kowalczyk; Oliver Thiergart; Emanuel Habets
In hands-free communication applications, the main goal is to capture desired sounds, while reducing noise and interfering sounds. However, for natural-sounding telepresence systems, the spatial sound image should also be preserved. Using a recently proposed method for generating the signal of a virtual microphone (VM), one can recreate the sound image from an arbitrary point of view in the sound scene (e.g., close to a desired speaker), while being able to place the physical microphones outside the sound scene. In this paper, we present a method for synthesizing a VM signal in noisy and reverberant environments, where the estimation of the required direct and diffuse sound components is performed using two multichannel linear filters. The direct sound component is estimated using a multichannel Wiener filter, while the diffuse sound component is estimated using a linearly constrained minimum variance filter followed by a single-channel Wiener filter. Simulations in a noisy and reverberant environment show the applicability of the proposed method for sound acquisition in a scenario in which two microphone arrays are installed in a large TV.
IEEE Transactions on Audio, Speech, and Language Processing | 2015
Jonathan Sheaffer; Maarten van Walstijn; Boaz Rafaely; Konrad Kowalczyk
Due to its efficiency and simplicity, the finite-difference time-domain method is becoming a popular choice for solving wideband, transient problems in various fields of acoustics. So far, the issue of extracting a binaural response from finite difference simulations has only been discussed in the context of embedding a listener geometry in the grid. In this paper, we propose and study a method for binaural response rendering based on a spatial decomposition of the sound field. The finite difference grid is locally sampled using a volumetric array of receivers, from which a plane wave density function is computed and integrated with free-field head related transfer functions, in the spherical harmonics domain. The volumetric array is studied in terms of numerical robustness and spatial aliasing. Analytic formulas that predict the performance of the array are developed, facilitating spatial resolution analysis and numerical binaural response analysis for a number of finite difference schemes. Particular emphasis is placed on the effects of numerical dispersion on array processing and on the resulting binaural responses. Our method is compared to a binaural simulation based on the image method. Results indicate good spatial and temporal agreement between the two methods.