Near Field Adaptive Beamforming minimizes the problem of signal degradation for near field sources for speech enhancement by leveraging both the good low frequency performance of near field superdirectivity and the adaptability of a General Sidelobe Canceller system. The near field compensation function corrects for the attenuation of the source signal amplitude that decreases as a function of distance to the microphone array.
VOCAL’s near field adaptive beamformer software package includes algorithms to determine the source location and dynamically steer and focus a beam at the source. Our beamforming software is optimized for DSPs and conventional processors from TI, ADI, ARM, Intel and other vendors. Custom beamforming solutions are also available. Contact us to discuss your adaptive beamforming application requirements.
Near Field Adaptive Beamformer
The Near Field Adaptive Beamformer employs a Generalized Sidelobe Canceler (GSC) structure that separates the beamformer into a fixed beamformer processing path, and an adaptive path. The adaptive path includes near field compensation, a blocking matrix, and a set of adaptive filters. The desired source signal is constrained in the fixed beamformer path, whereas the adaptive path is updated using an unconstrained Least Mean Squared (LMS) algorithm.
Microphone array beamformers typically assume a planar acoustic source signal wavefront. This assumption is not optimum for the near field source. In the near field a, an acoustic source signal of frequency, ƒ, is considered in the near-field when d < 2cl2 ⁄ λ, where l is the total size of the microphone array, d is the distance of the signal from the center of the microphone array and c is the velocity of sound.
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References
[1] I. Mccowan, D. Moore, and S. Sridharan, ”Near-field Adaptive Beamformer for Robust Speech Recognition”, In Digital Signal Processing, pp 87-106, 2001.