Gannot, Sharon; Dvorkind, Tsvi Gregory Microphone array speaker localizers using spatial-temporal information. (English) Zbl 1122.94322 EURASIP J. Appl. Signal Process. 2006, No. 12, Article ID 59625, 17 p. (2006). Summary: A dual-step approach for speaker localization based on a microphone array is addressed in this paper. In the first stage, which is not the main concern of this paper, the time difference between arrivals of the speech signal at each pair of microphones is estimated. These readings are combined in the second stage to obtain the source location. In this paper, we focus on the second stage of the localization task. In this contribution, we propose to exploit the speaker’s smooth trajectory for improving the current position estimate. Three localization schemes, which use the temporal information, are presented. The first is a recursive form of the Gauss method. The other two are extensions of the Kalman filter to the nonlinear problem at hand, namely, the extended Kalman filter and the unscented Kalman filter. These methods are compared with other algorithms, which do not make use of the temporal information. An extensive experimental study demonstrates the advantage of using the spatial-temporal methods. To gain some insight on the obtainable performance of the localization algorithm, an approximate analytical evaluation, verified by an experimental study, is conducted. This study shows that in common TDOA-based localization scenarios–where the microphone array has small interelement spread relative to the source position–the elevation and azimuth angles can be accurately estimated, whereas the Cartesian coordinates as well as the range are poorly estimated. MSC: 94A13 Detection theory in information and communication theory PDF BibTeX XML Cite \textit{S. Gannot} and \textit{T. G. Dvorkind}, EURASIP J. Appl. Signal Process. 2006, No. 12, Article ID 59625, 17 p. (2006; Zbl 1122.94322) Full Text: DOI References: [1] IEEE Transactions on Acoustics, Speech, and Signal Processing 24 (4) pp 320– (1976) [2] Signal Processing 59 (3) pp 253– (1997) [3] The Journal of the Acoustical Society of America 107 (1) pp 384– (2000) [4] EURASIP Journal on Applied Signal Processing 2003 (11) pp 1110– (2003) [5] Signal Processing 85 (1) pp 177– (2004) · Zbl 1052.94002 [6] IEEE Transactions on Signal Processing 42 (8) pp 1905– (1994) [7] IEEE Transactions on Speech and Audio Processing 5 (1) pp 45– (1997) [8] IEEE Transactions on Acoustics, Speech, and Signal Processing 35 (8) pp 1223– (1987) [9] IEEE Transactions on Acoustics, Speech, and Signal Processing 35 (12) pp 1661– (1987) [10] IEEE Transactions on Speech and Audio Processing 9 (8) pp 943– (2001) [11] Numerical Recipes in C: The Art of Scientific Computing (1988) · Zbl 0661.65001 [12] IEEE Transactions on Signal Processing 50 (8) pp 1843– (2002) [13] IEEE Transactions on Signal Processing 39 (1) pp 1– (1991) [14] IEEE Transactions on Speech and Audio Processing 11 (6) pp 826– (2003) [15] Advances in Radio Science 1 pp 113– (2003) [16] Information and System Sciences, in: Adaptive Filter Theory, 4th. ed. (2002) [17] Proceedings of the IEEE 92 (3) pp 401– (2004) [18] The Journal of the Acoustical Society of America 65 (4) pp 943– (1979) This reference list is based on information provided by the publisher or from digital mathematics libraries. Its items are heuristically matched to zbMATH identifiers and may contain data conversion errors. It attempts to reflect the references listed in the original paper as accurately as possible without claiming the completeness or perfect precision of the matching.