Dual-Channel Speech Enhancement by Superdirective Beamforming
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Dual-Channel Speech Enhancement by Superdirective Beamforming Thomas Lotter and Peter Vary Institute of Communication Systems and Data Processing, RWTH Aachen University, 52056 Aachen, Germany Received 31 January 2005; Revised 8 August 2005; Accepted 22 August 2005 In this contribution, a dual-channel input-output speech enhancement system is introduced. The proposed algorithm is an adaptation of the well-known superdirective beamformer including postfiltering to the binaural application. In contrast to conventional beamformer processing, the proposed system outputs enhanced stereo signals while preserving the important interaural amplitude and phase differences of the original signal. Instrumental performance evaluations in a real environment with multiple speech sources indicate that the proposed computational efficient spectral weighting system can achieve significant attenuation of speech interferers while maintaining a high speech quality of the target signal. Copyright © 2006 T. Lotter and P. Vary. This is an open access article distributed under the Creative Commons Attribution License, which permits unrestricted use, distribution, and reproduction in any medium, provided the original work is properly cited.
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INTRODUCTION
Speech enhancement by beamforming exploits spatial diversity of desired speech and interfering speech or noise sources by combining multiple noisy input signals. Typical beamformer applications are hands-free telephony, speech recognition, teleconferencing, and hearing aids. Beamformer realizations can be classified into fixed and adaptive. A fixed beamformer combines the noisy signals of multiple microphones by a time-invariant filter-and-sum operation. The combining filters can be designed to achieve constructive superposition towards a desired direction (delayand-sum beamformer) or in order to maximize the SNR improvement (superdirective beamformer), for example, [1]. As practical problems such as self-noise and amplitude or phase errors of the microphones limit the use of optimal beamformers, constrained solutions have been introduced that limit the directivity to the benefit of reduced susceptibility [2–4]. Most fixed beamformer design algorithms assume the desired source to be positioned in the far field, that is, the distance between the microphone array and the source is much greater than the dimension of the array. Nearfield superdirectivity [5] additionally exploits amplitude differences between the microphone signals. Adaptive beamformers commonly consist of a fixed beamformer steered towards a desired direction and a time-varying branch, which adaptively steers beamformer spatial nulls towards interfering sources. Among various adaptive beamformers, the
Griffiths-Jim beamformer [6], or extensions, for example, in [7, 8], is most widely known. Adaptive beamformers can be considered less robust against distortions of the desired signal than fixed beamformers. Beamforming for binaural input signals, that is, signals recorded by single microphones at the left and right ear, has found
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