Abstract: The invention relates to method for determining a feature of the circadian rhythm of the subject, comprising measuring a first input signal indicating a cardiac function of the subject, measuring at least one second input signal indicating the activity of the subject, combining the first input signal and the second input signal to a periodic output signal representing the circadian rhythm of the subject, and determining at least one feature of this periodic output signal.

Abstract: 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

Abstract: A system (1) is described which is suited for suppressing audio distortion. The system comprises echo cancelling means g1, g2 coupled between an audio output (4) and a distorted desired audio sensing microphone array (3), and a filter arrangement (7) coupled to the echo cancelling means g1, g2 and/or the microphone array (3). The filter arrangements' coefficients represent reverberation distortion in the desired audio sensed by the microphone array (3).

Abstract: The relatively robust adaptive beamformer, comprises: a filtered sum beamformer (107) to process input audio signals (u1, u2, u3) from an array of respective microphones (101, 103, 105), and arranged to yield as an output a first audio signal (z) predominantly corresponding to sound from a desired audio source (160); and a noise estimation e.g. when incorporated in a sidelobe canceller topology an adaptive noise estimator (150), arranged to derive a noise signal (y) which is subtracted from the first audio signal (z) to obtain a noise cleaned second audio signal (r), and further comprises a scaling factor determining unit (170) arranged to provide a scale factor (S) as a function of a ratio (Q) of the sidelobe canceling, and being arranged to scale the adaptation step size with the scale factor (S), so that the sidelobe canceller only adapts quickly if it is relatively well locked on the desired audio source, but is rather insensitive to interference from noise sources.