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—We consider a sensor array located in an enclosure , where arbitrary transfer functions (TFs) relate the source signal and the sensors. The array is used for enhancing a signal contaminated by interference. Constrained minimum power adaptive beamforming, which has been suggested by Frost and, in particular, the generalized sidelobe canceler (GSC) version,(More)
—Speech quality and intelligibility might significantly deteriorate in the presence of background noise, especially when the speech signal is subject to subsequent processing. In particular , speech coders and automatic speech recognition (ASR) systems that were designed or trained to act on clean speech signals might be rendered useless in the presence of(More)
In speech enhancement applications microphone array postfilte-ing allows additional reduction of noise components at a beam-former output. Among microphone array structures the recently proposed General Transfer function Generalized Sidelobe Can-celler (TF–GSC) has shown impressive noise reduction abilities in a directional noise field, while still(More)
—In speech communication systems the received microphone signals are degraded by room reverberation and ambient noise that decrease the fidelity and intelligibility of the desired speaker. Reverberant speech can be separated into two components, viz. early speech and late reverberant speech. Recently, various algorithms have been developed to suppress late(More)
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(More)
—In many practical environments we wish to extract several desired speech signals, which are contaminated by non-stationary and stationary interfering signals. The desired signals may also be subject to distortion imposed by the acoustic room impulse responses (RIRs). In this paper, a linearly constrained minimum variance (LCMV) beamformer is designed for(More)
In this paper, we present a novel approach for real-time multichannel speech enhancement in environments of non-stationary noise and time-varying acoustical transfer functions (ATFs). The proposed system integrates adaptive beamforming, ATF identification, soft signal detection, and multichannel postfiltering. The noise canceller branch of the beamformer(More)
In this paper we introduce a novel algorithm for extracting desired speech signals uttered by moving speakers contaminated by competing speakers and stationary noise in a reverberant environment. The proposed beamformer uses eigenvectors spanning the desired and interference signals subspaces. It relaxes the common requirement on the activity patterns of(More)
Besides noise reduction an important objective of binaural speech enhancement algorithms is the preservation of the bin-aural cues of both desired and undesired sound sources. Recently , the binaural Linearly Constrained Minimum Variance (BLCMV) beamformer has been proposed that aims to preserve the desired speech component and suppress the unde-sired(More)