Michael S. Brandstein

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Conventional time-delay estimators exhibit dramatic performance degradations in the presence of multipath signals. This limits their application in reverberant enclosures, particularly when the signal of interest is speech and it may not possible to estimate and compensate for channel e ects prior to time-delay estimation. This paper details an alternative(More)
Engineering with departmental honors. He went on to Brown University in Providence, Rhode Island to study signal processing and began research on microphone arrays. He received a Master of Science degree in Electrical Engineering in 1993 and continued to pursue his work towards a Doctor of Philosophy degree. While a student at Brown, he held several(More)
The linear intersection (LI) estimator, a closed-form method for the localization of source positions given sensor array time-delay estimate information, is presented. The LI estimator is shown to be robust and accurate, to closely model the search-based ML estimator, and to outperform a benchmark algorithm. The computational complexity of the LI estimator(More)
The relative time delay associated with a speech signal received at a pair of spatially separated microphones is a key component in talker localization and microphone array beamforming procedures. The traditional method for estimating this parameter utilizes the generalized cross correlation (GCC), the performance of which is compromised by the presence of(More)
A frequency-domain-based delay estimator is described, designed speci cally for speech signals in a microphone-array environment. It is shown to be capable of obtaining precision delay estimates over a wide range of SNR conditions and is simple enough computationally to make it practical for real-time systems. A location algorithm based upon the delay(More)
A method for tracking the positional estimates of multiple talkers in the operating region of an acoustic microphone array is presented. Initial talker location estimates are provided by a time-delay-based localization algorithm. These raw estimates are spatially smoothed by a Kalman filter derived from a set of potential source motion models. Data(More)
This paper presents a method for enhancing multi-channel reverberant speech using event-based processing of wavelet transform coefficients. Clustering of the wavelet extrema across multiple channels is employed to obtain a single multi-scale extrema representation from which the enhanced signal is synthesized. Processing is done in the LPC residual domain,(More)
This paper addresses the limitations of current approaches to distant-talker speech acquisition and advocates the development of techniques which explicitly incorporate the nature of the speech signal (e.g. statistical non-stationarity, method of production, pitch, voicing, formant structure, and source radiator model) into a multi-channel context. The goal(More)