Botond Sandor Kirei

Botond Sandor Kirei

orcid

Research Focus & Metrics

Academic Career Length

20Years (2007-Present)

(Calculated once Scilit publication data is synced: current year − year of first publication + 1)

Representative Publications

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1 May 2013 in Signal Processing
Verified
Journal Article
Closed access
CCCristian Contan
Cristian Contan
BKBotond Sandor Kirei
Botond Sandor Kirei
MTMarina Dana T¸oPa
Marina Dana T¸oPa

29 March 2011 in Acoustical Physics
Verified
Journal Article
Closed access
PDF file
MŢMarina Dana Ţopa
Marina Dana Ţopa
NTNorbert Toma
Norbert Toma
BKBotond Sandor Kirei
Botond Sandor Kirei
IHIoana Homana
Ioana Homana
MNMarius Neag
Marius Neag
GMGilbert De Mey
Gilbert De Mey
The paper presents the acoustics analysis of three different enclosed spaces. These spaces (rooms) have different geometrical shapes and sizes and serve for different purposes. The early decay time, reverberation time, clarity and center time are evaluated with Dirac, WinMLS, Aurora and Caracad software using simple, low-cost equipment. The listed acoustic parameters were determined using linear sine sweep and impulsive sources. Comparisons between experimental measurements, simulations and analytic results were done. The room impulse response measurement proved to be the most reliable method to evaluate the properties of different rooms even if the measurements are perturbed by non-idealities of the low-cost equipment.

1 September 2009
Verified
Conference Paper
Closed access
IHIoana Homana
Ioana Homana
MTMarina Dana Topa
Marina Dana Topa
BKBotond Sandor Kirei
Botond Sandor Kirei
In the paper, an echo canceller based on a system identification scheme with adaptive algorithms is presented. The algorithms consider FIR filters with the taps chosen to minimize an error signal derived from the system according to a stochastic gradient-based method. The following adaptive filters are discussed and simulated in Matlab: classical LMS and its variants: NLMS, VSLMS, VSNLMS and RLS. The echo was simulated by adding several delayed and attenuated replica to a vocal input. The comparison between the algorithms was made taking into account the mean-square error (MSE) and average attenuation (AV), as well as the required number of multiplications.

19 April 2012 in IEEE Transactions on Circuits and Systems II: Express Briefs
Verified
Journal Article
Closed access
BKBotond Sandor Kirei
Botond Sandor Kirei
MNM. G. Neag
M. G. Neag
MTM. D. Topa
M. D. Topa
This brief proposes a new approach to frequency-selective in-phase (I)/quadrature (Q) mismatch compensation, based on subband processing: The spectrum of the signal received by the digital baseband processor-a combination of wanted and image signals-is split into subbands so that, within each subband, the I/Q mismatch can be considered constant with respect to frequency. Therefore, standard compensation algorithms, devised for frequency-invariant I/Q mismatch, can be applied to compensate the signal within each subband; finally, the outputs of each subband processing path are recombined to yield the wanted signal. This procedure is demonstrated by a case study that describes an I/Q mismatch compensation algorithm particularly well suited for the proposed method and proves that, at least for certain I/Q imbalances, the efficiency of compensation can be significantly improved even if only a small number of subbands are employed. The resulting frequency-selective I/Q mismatch compensation solution is suitable for wideband multistandard receivers.

16 April 2015 in Signal, Image and Video Processing
Verified
Journal Article
Closed access
PDF file
CCCristian Contan
Cristian Contan
BKBotond Sandor Kirei
Botond Sandor Kirei
MTMarina Dana Topa
Marina Dana Topa

1 June 2011
Verified
Conference Paper
Closed access
CCCristian Contan
Cristian Contan
MTMarina Topa
Marina Topa
BKBotond Sandor Kirei
Botond Sandor Kirei
IHIoana Homana
Ioana Homana
The paper proposes a nonlinear system identification method that uses a combination of adaptive linear, Volterra and power filters. Adaptation of the kernels is made using a Normalized Least Mean Square algorithm. The method is applied in echo cancellation, where several sources of nonlinearities exist: the overdriven amplifier, the small loudspeaker at high volume, the room with different absorbent walls. Functions with nonlinear characteristics are chosen to model these distortions. The evaluation is made in terms of Echo Return Loss Enhancement. Results show that the overall convex combination approach performs better or at least as well as the best single adaptive filter.