Topic
Adaptive beamformer
About: Adaptive beamformer is a research topic. Over the lifetime, 4934 publications have been published within this topic receiving 93100 citations.
Papers published on a yearly basis
Papers
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TL;DR: Basic adaptive beamforming techniques are compared using shallow sea sonar data recorded in a towed horizontal array environment with moving interfering sources originated from shipping noise to demonstrate noticeable performance improvements that can be achieved.
Abstract: The performances of adaptive array algorithms are known to suffer from a strong degradation in scenarios with moving interfering sources. In this article, basic adaptive beamforming techniques are compared using shallow sea sonar data recorded in a towed horizontal array environment with moving interfering sources originated from shipping noise. Our experimental results show the relationship between the practical performances of adaptive and conventional beamforming techniques compared in terms of output SINR or a related measure given by the noncompensated postbeamforming interference power. These results demonstrate noticeable performance improvements that can be achieved using several robust algorithms relative to traditional adaptive beamforming schemes.
71 citations
TL;DR: In this article, the recursive maximum likelihood algorithm is proposed for estimating the parameters of the signal model and the parameter estimates are then used to form an adaptive infinite impulse response filter, which can be used to solve several problems in adaptive filtering.
Abstract: Many problems in adaptive filtering can be approached from the point of view of system identification. The recursive maximum likelihood algorithm is proposed for estimating the parameters of the signal model. The parameter estimates are then used to form an adaptive infinite impulse response filter. Several examples are discussed including: adaptive line enhancement, adaptive deconvolution, adaptive noise cancelling, and adaptive time delay estimation.
71 citations
TL;DR: Experimental results demonstrate that the proposed beamformer outperforms the transfer function GSC (TF-GSC) in reverberant environments and achieves both improved noise reduction and reduced speech distortion.
Abstract: In this paper, we propose a convolutive transfer function generalized sidelobe canceler (CTF-GSC), which is an adaptive beamformer designed for multichannel speech enhancement in reverberant environments. Using a complete system representation in the short-time Fourier transform (STFT) domain, we formulate a constrained minimization problem of total output noise power subject to the constraint that the signal component of the output is the desired signal, up to some prespecified filter. Then, we employ the general sidelobe canceler (GSC) structure to transform the problem into an equivalent unconstrained form by decoupling the constraint and the minimization. The CTF-GSC is obtained by applying a convolutive transfer function (CTF) approximation on the GSC scheme, which is a more accurate and a less restrictive than a multiplicative transfer function (MTF) approximation. Experimental results demonstrate that the proposed beamformer outperforms the transfer function GSC (TF-GSC) in reverberant environments and achieves both improved noise reduction and reduced speech distortion.
70 citations
Patent•
TL;DR: The hybrid adaptive beamformer of the present invention includes a plurality of microphones for receiving sound energy from an external environment and for producing a second order output beam taking into consideration the determined amount of reverberation as mentioned in this paper.
Abstract: The hybrid adaptive beamformer of the present invention includes a plurality of microphones for receiving sound energy from an external environment and for producing a plurality of microphone outputs from the sound energy. A processor produces a plurality of first order beams based on the microphone outputs. The processor determines an amount of reverberation in the external environment and adaptively produces a second order output beam taking into consideration the determined amount of reverberation. The processor may determine the amount of reverberation based on a comparison of the first order beams. The processor may produce the second order output beam by adaptively combining the plurality of first order beams, as further described below, or by adaptively combining the microphone outputs. The adaptation varies taking into consideration the determined amount of reverberation. Alternatively, the adaptation may vary by measuring the signal-to-noise ratio of the first order beams and adaptively combining the first order beams based on the determined signal-to-noise ratios.
70 citations
TL;DR: In this paper, a total solution for frequency-domain blind source separation (BSS) of convolutive mixtures of audio signals, especially speech, is presented, including permutation, scaling, circularity, and complex activation function solutions.
Abstract: This paper overviews a total solution for frequency-domain blind source separation (BSS) of convolutive mixtures of audio signals, especially speech. Frequency-domain BSS performs independent component analysis (ICA) in each frequency bin, and this is more efficient than time-domain BSS. We describe a sophisticated total solution for frequency-domain BSS, including permutation, scaling, circularity, and complex activation function solutions. Experimental results of 2 × 2, 3 × 3, 4 × 4, 6 × 8, and 2 × 2 (moving sources), (#sources × #microphones) in a room are promising.
70 citations