Patent attributes
The adaptive beamformer unit (191) comprises: a filtered sum beamformer (107) arranged to process input audio signals (u 1, u2) from an array of respective microphones (101, 103), and arranged to yield as an output a first audio signal (z) predominantly corresponding to sound from a desired audio source (160) by filtering with a first adaptive filter (f1(-t)) a first one of the input audio signals (u1) and with a second adaptive filter (f2(-t)) a second one of the input audio signals (u2), the coefficients of the first filter (f1(-t)) and the second filter (f2(-t)) being adaptable with a first step size (a1) and a second step size ((x2) respectively; noise measure derivation means (111) arranged to derive from the input audio signals (u1, u2) a first noise measure (x1) and a second noise measure (x2); and an updating unit (192) arranged to determine the first and second step size (a1, (x2) with an equation comprising in a denominator the first noise measure (x1) for the first step size (a1), respectively the second noise measure (x2) for the second step size (a2). This makes the beamformer relatively robust against the influence of correlated audio interference. The beamformer may also be incorporated in a sidelobe canceller topology yielding a more noise cleaned desired sound estimate, which can be used in a related, more advanced adaptive filter (f1(-t), f2(-t)) updating. Such a beamformer is typically useful for application in handsfree speech communication systems.
