Abstract: This specification describes technologies relating to editing digital audio data. In general, one aspect of the subject matter described in this specification can be embodied in methods that include the actions of receiving an audio signal including audio data in multiple channels; identifying noise in the audio signal including identifying panning information for the audio data in the signal at each of multiple frequency bands; and attenuating the audio data at one or more frequency bands to generate an edited audio signal when the panning exceeds a specified threshold for each of the one or more frequency bands. Other embodiments of this aspect include corresponding systems, apparatus, and computer program products.

Abstract: A system locates a speaker in a room containing a loudspeaker and a microphone array. The loudspeaker transmits a sound that is partly reflected by a speaker. The microphone array detects the reflected sound and converts the sound into a microphone signal. A processor determines the speaker's direction relative to the microphone array, the speaker's distance from the microphone array, or both, based on the characteristics of the microphone signals.

Abstract: Systems and methods for adaptive processing of a close microphone array in a noise suppression system are provided. A primary acoustic signal and a secondary acoustic signal are received. In exemplary embodiments, a frequency analysis is performed on the acoustic signals to obtain frequency sub-band signals. An adaptive equalization coefficient may then be applied to a sub-band signal of the secondary acoustic signal. A forward-facing cardioid pattern and a backward-facing cardioid pattern are then generated based on the sub-band signals. Utilizing cardioid signals of the forward-facing cardioid pattern and backward-facing cardioid pattern, noise suppression may be performed. A resulting noise suppressed signal is output.

Abstract: A sound transducer, comprising at least one acoustically neutral body with which it is possible to associate two sound conveyance elements which are shaped approximately like a stylized funnel so as to each form an auricle, which protrudes outside the acoustically neutral body and is blended with a duct with which a three-pole microphone cartridge is associated, the cartridge being arranged so that its front end, adapted to acquire the sound, is proximate to the inlet of the duct. The two cold poles of the microphone cartridges are mutually inverted, so that the cold pole of one of the microphone cartridges and the hot pole and the ground of the other of the microphone cartridges are or can be connected to a same connector or socket which is or can be associated with an amplifying and/or recording and/or processing device.

Abstract: An apparatus processes an acoustic input signal to provide an output signal with reduced noise. The apparatus weights the input signal based on a frequency-dependent weighting function. A frequency-dependent threshold function bounds the weighting function from below.

Abstract: A method and an apparatus for designing a sound compensation filter of a portable terminal are provided. The method includes synchronizing a signal input through a microphone of the system and a test signal, estimating a loss interval of the synchronized signal, compensating for a frame signal delayed by a signal loss in a time axis when the signal loss of the estimated loss interval is greater than a threshold and restoring the loss interval of the signal.

Abstract: Systems and methods for utilizing inter-microphone level differences (ILD) to attenuate noise and enhance speech are provided. In exemplary embodiments, primary and secondary acoustic signals are received by omni-directional microphones, and converted into primary and secondary electric signals. A differential microphone array module processes the electric signals to determine a cardioid primary signal and a cardioid secondary signal. The cardioid signals are filtered through a frequency analysis module which takes the signals and mimics a cochlea implementation (i.e., cochlear domain). Energy levels of the signals are then computed, and the results are processed by an ILD module using a non-linear combination to obtain the ILD. In exemplary embodiments, the non-linear combination comprises dividing the energy level associated with the primary microphone by the energy level associated with the secondary microphone.

Abstract: To reliably and consistently detect desirable sounds, a system detects the presence of wind noise based on the power levels of audio signals. A first transducer detects sound originating from a first direction and a second transducer detects sound originating from a second direction. The power levels of the sound are compared. When the power level of the sound received from the second transducer is less than the power level of the sound received from the first transducer by a predetermined value, wind noise may be present. A signal processor may generate an output from one or a combination of the audio signals, based on a wind noise detection.

Abstract: An audio device includes a first audio path with a loudspeaker for reproducing an audio signal, and a second audio path. The second audio path includes in series a band-pass filter for filtering an audio signal, a detector for detecting the amplitude of the band-pass filtered audio signal, a multiplier for multiplying a periodic signal by the amplitude of the band-pass filtered audio signal, and a vibration device for reproducing the multiplied periodic signal. The frequency of the periodic signal is substantially equal to the resonance frequency of the vibration device.

Abstract: A system for estimating the background noise in a loudspeaker-room-microphone system is presented herein where the loudspeaker is supplied with a source signal and the microphone picks up the source signal distorted by the room and provides a distorted signal. The system comprises an adaptive filter receiving the source signal and the distorted signal, and providing an error signal, a post filter connected downstream of the adaptive filter and a smoothing filter arrangement connected downstream of the adaptive filter. The smoothing filter arrangement includes a spectral domain smoothing filter that provides a spectral domain estimated-noise signal, and a time domain smoothing filter that provides a time domain estimated-noise signal.

Abstract: A signal processor uses input devices to detect speech or aural signals. Through a programmable set of weights and/or time delays (or phasing) the output of the input devices may be processed to yield a combined signal. The noise contributions of some or each of the outputs of the input devices may be estimated by a circuit element or a controller that processes the outputs of the respective input devices to yield power densities. A short-term measure or estimate of the noise contribution of the respective outputs of the input devices may be obtained by processing the power densities of some or each of the outputs of the respective input devices. Based on the short-term measure or estimate, the noise contribution of the combined signal may be estimated to enhance the combined signal when processed further. An enhancement device or post-filter may reduce noise more effectively and yield robust speech based on the estimated noise contribution of the combined signal.

Abstract: In one embodiment, a two-way telecommunication device may perform acoustic echo cancellation on incoming signals. An audio decoding module may produce an audio render signal. An audio capture interface may receive an audio capture signal. A short length adaptive filter may determine a time delay between the audio render signal and the audio capture signal by adaptively predicting a sub-band of the audio capture signal using a corresponding sub-band of the audio render signal.

Abstract: Methods and systems for masking audio noise are disclosed. One apparatus includes a silence detector configured to detect a period of substantial silence in an audio signal; a masking noise source operably coupled to the silence detector, the masking noise source configured to generate a noise signal in response to the silence detector detecting the period of substantial silence; and at least one combining device operably coupled to the masking noise source, the at least one combining device configured to contribute to combining the audio signal and the noise signal. A method includes detecting a period of substantial silence in an audio signal; and combining masking noise with the audio signal during the period of substantial silence.

Abstract: An acoustic echo cancellation device for canceling an echo in a microphone signal in response to first and second input signals includes a first combination unit for combining the first and second input signals into an aggregate input signal. The device further includes an adaptive filter unit for filtering the aggregate input signal so as to produce an aggregate echo cancellation signal. A second combination unit combines the aggregate echo cancellation signal with the microphone signal so as to produce a residual signal, and an additional filter unit filters either the first or second input signal so as to produce a first or a second partial echo cancellation signal. A post-processor unit suppresses remaining echo components in the residual signal. The post-processor unit uses at least one partial echo cancellation signal to suppress echo components corresponding with the first input signal to a greater extent than echo components corresponding with the second input signal.

Abstract: An audio signal separation apparatus for separating observation signals in the time domain of a mixture of a plurality of signals including audio signals into individual signals by means of independent component analysis to produce isolated signals adapted to produce isolated signals in the time-frequency domain from the observation signals in the time-frequency domain and a separation matrix substituted by initial values, compute the modified value of the separation matrix by using a score function using the isolated signals in the time-frequency domain and a multidimensional probability density function and the separation matrix, modify the separation matrix until the separation matrix substantially converges by using the modified value and produce isolated signals in the time-frequency domain by using the substantially converging separation matrix.

Abstract: Disclosed is a method and an apparatus for estimating noise included in a sound signal during sound signal processing. The method includes estimating harmonics components in a frame of an input sound signal; using the estimated harmonics components, computing a Voice Presence Probability (VPP) on the frame of the input sound signal; determining a weight of an equation necessary to estimate a noise spectrum, depending on the computed VPP; and using the determined weight and the equation necessary to estimate a noise spectrum, estimating the noise spectrum, and updating the noise spectrum.

Abstract: A speaker array and microphone arrays positioned on both sides of the speaker array are provided. A plurality of focal points each serving as a position of a talker are set in front of the microphone arrays respectively symmetrically with respect to a centerline of the speaker array, and a bundle of sound collecting beams is output toward the focal points. Difference values between sound collecting beams directed toward the focal points that are symmetrical with respect to the centerline are calculated to cancel sound components that detour from the speaker array to microphones. Then, it is estimated based on totals of squares of peak values of the difference values for a particular time period that the position of the talker is close to which one of the focal points, and the position of the talker is decided by comparing the totals of the squares of the peak values of the sound collecting beams directed to the focal points that are symmetrical mutually.

Abstract: The present invention discloses a multi-level output signal converter, which is connected to an audio amplifier. The audio amplifier comprises a comparing/measuring device, an encoder and an output unit. The multi-level output signal converter comprises a timing processing unit and a multi-level converter. The timing processing unit is connected to the comparing/measuring device and the encoder. The timing processing unit includes a plurality of flip-flops and a timing summing element. The flip-flop receives a first signal from the comparing/measuring device and outputs the first signal to the timing summing element. The encoder converts the first signal into a second signal. The multi-level converter is connected to the encoder and the output unit. The encoder transmits the second signal to the multi-level converter, and the multi-level converter thus outputs a third signal to the output unit.

Abstract: The present invention is to provide a sound image localization apparatus which can prevent the lowering of the amplitude of the sound image localizing signal, the occurrence of clipping, and deterioration of the sound image localization component of the sound image localizing signal.

Abstract: Disclosed herein is a noise removing apparatus, including: an object sound emphasis section adapted to carry out an object sound emphasis process for observation signals of first and second microphones to produce an object sound estimation signal; a noise estimation section adapted to carry out a noise estimation process for the observation signals to produce a noise estimation signal; a post filtering section adapted to remove noise components remaining in the object sound estimation signal using the noise estimation signal; a correction coefficient calculation section adapted to calculate, for each frequency, a correction coefficient for correcting the post filtering process based on the object sound estimation signal and the noise estimation signal; and a correction coefficient changing section adapted to change those of the correction coefficients which belong to a frequency band suffering from spatial aliasing such that a peak appearing at a particular frequency is suppressed.

Abstract: A concept is proposed for a MEMS microphone which may be operated at a relatively low voltage level and still have comparatively high sensitivity. The component according to the present invention includes a micromechanical microphone structure having an acoustically active diaphragm which functions as a deflectable electrode of a microphone capacitor (1), and a stationary acoustically permeable counterelement which functions as a counter electrode of the microphone capacitor (1).

Abstract: In a filter coefficient calculation device according to the present invention, a gain correction characteristic calculation section calculates impulse responses corresponding to a linear-phase filter having an inverse characteristic of a gain characteristic of a reproduction system, and calculates, as a gain correction characteristic, a frequency characteristic of continuous-time impulse responses that include a peak value, the continuous-time impulse responses being impulse responses, clipped from the calculated impulse responses, whose number is identical to the preset number of filter taps.

Abstract: An audio apparatus has a function of correcting an audio signal in response to a noise level. The audio apparatus includes a correction unit that corrects an input audio signal on the basis of a weighting factor, an output unit that produces a played-back audio sound on the basis of the corrected audio signal, a microphone for receiving an external sound that includes the played-back audio sound and noise, a noise-extracting unit that extracts a noise signal from an external sound signal, the noise-extracting unit including a speech-removing unit that removes a speech signal from the noise signal on the basis of noise spectrum data, and a weighting factor calculation unit that calculates the weighting factor on the basis of the extracted noise signal and supplies the calculated weighting factor to the correction unit.

Abstract: In signal-component extraction, an input signal is delayed to generate a delayed input signal. The input signal is adaptively filtered with filter coefficients, to generate a filtered signal. The filtered signal is subtracted from the delayed input signal to generate an error signal. A preset reference value is divided by an amplitude of the input signal to generate a gain value. The filter coefficients are derived based on a value obtained by multiplying the input signal and error signal by the gain value or a square of the gain value.

Abstract: A system and method are disclosed for enhancing audio signals by nonlinear spectral operations. Successive portions of the audio signal are processed using a subband filter bank. A nonlinear modification is applied to the output of the subband filter bank for each successive portion of the audio signal to generate a modified subband filter bank output for each successive portion. The modified subband filter bank output for each successive portion is processed using an appropriate synthesis subband filter bank to construct a modified time-domain audio signal. High modulation frequency portions of the audio signal may be emphasized or de-emphasized, as desired. The modification may be applied within one or more frequency bands.

Abstract: Disclosed herein are detectors of audio ringing feedback, that is decaying feedback with a gain of less than one, those detectors utilizing a repeated gain measurement that applied to a range of gain values characteristic of ringing-type feedback. Those gain measurements, while in the range, increase a probability measurement of feedback. When the probability of feedback reaches a threshold, a detection of feedback is made and feedback countermeasures, such as the application of a notch filter, may be applied. Optionally, the audio gain around likely frequencies of feedback may be enhanced for a time to increase the resolution of identification of a feedback frequency, which may be identified through an interpolative method. Repeated gain measurements may also identify building-type feedback. A ringing detector may include more than one range of detection, for example for building, strong-ringing and weak-ringing feedback.

Abstract: Spatial noise suppression for audio signals involves generating a ratio of powers of difference and sum signals of audio signals from two microphones and then performing noise suppression processing, e.g., on the sum signal where the suppression is limited based on the power ratio. In certain embodiments, at least one of the signal powers is filtered (e.g., the sum signal power is equalized) prior to generating the power ratio. In a subband implementation, sum and difference signal powers and corresponding the power ratio are generated for different audio signal subbands, and the noise suppression processing is performed independently for each different subband based on the corresponding subband power ratio, where the amount of suppression is derived independently for each subband from the corresponding subband power ratio. In an adaptive filtering implementation, at least one of the audio signals can be adaptively filtered to allow for array self-calibration and modal-angle variability.

Abstract: A method and system to reduce the noise floor of a communications system is disclosed. The system may be incorporated into any device that provides binary samples from a datastream, such as a cordless telephone system. The system is configured to determine a number of bits of the binary samples that are affected by noise. The system is then able to remove the noise by setting those bits to a fixed value. The fixed value may depend on whether the sample is positive or negative. The value to set may be chosen so that the least significant bits of each sample come as close as possible to 0 for that particular numerical representation system. The system can be integrated with other known signal processing methods.

Abstract: A feedback calibration system and a method for controlling an electronic signal are disclosed. The feedback calibration system includes an input controller adapted to modify an input signal in response to a control signal and generate a modified input signal, a signal processing block including a signal analyzer, wherein the signal processing block is adapted to process the modified input signal to generate an output signal and the signal analyzer is adapted to detect an undesirable condition of the output signal and transmit a detection signal corresponding to the undesirable condition, a transfer function estimator adapted to model and transmit a transfer function estimate of the signal processing block in real-time in response to the detection signal, and a programmable device adapted to transmit the control signal to the input controller for modifying the input signal, wherein the control signal is based upon the transfer function estimate.

Abstract: Disclosed herein is a signal processing apparatus including: a first decimation processing section for generating, based on a digital signal in a first form, a digital signal in a second form; a second decimation processing section for generating, based on the digital signal in the second form, a digital signal in a third form; a first signal processing section for processing the digital signal in the third form; an interpolation processing section for converting a digital signal in the third form outputted from the first signal processing section into a digital signal in the second form; a second signal processing section for processing the digital signal in the second form outputted from the first decimation processing section; and a combining section for combining the digital signals in the second form outputted from the interpolation processing section and the second signal processing section.

Abstract: Masking thresholds are obtained for each frequency component of sound data and ambient noise. It is determined whether each frequency component of the sound data is masked by at least one of the other frequency components of the sound data. It is further determined whether each frequency component of the sound data is masked by ambient noise. Correction coefficients are set for each frequency component of the sound data according to whether the frequency component is masked by at least one of the other frequency components of the sound data and whether the frequency component is masked by the ambient noise. And each frequency component of the sound data is corrected by using the respective correction coefficients.

Abstract: A signal separator, a method and computer product for determining a first output signal describing an audio content of a useful-signal source in a first microphone signal, and for determining a second output signal describing an audio content of the useful-signal source in a second microphone signal.

Abstract: Disclosed herein are detectors of audio ringing feedback, that is decaying feedback with a gain of less than one, those detectors utilizing a repeated gain measurement that applied to a range of gain values characteristic of ringing-type feedback. Those gain measurements, while in the range, increase a probability measurement of feedback. When the probability of feedback reaches a threshold, a detection of feedback is made and feedback countermeasures, such as the application of a notch filter, may be applied. Optionally, the audio gain around likely frequencies of feedback may be enhanced for a time to increase the resolution of identification of a feedback frequency, which may be identified through an interpolative method. Repeated gain measurements may also identify building-type feedback. A ringing detector may include more than one range of detection, for example for building, strong-ringing and weak-ringing feedback.

Abstract: A method and apparatus for detecting microphone interference includes first and second built-in microphones producing first and second microphone signals. A first filter bank creates first high-frequency-band and first low-frequency-band signals from the first microphone signal. A second filter bank creates second high-frequency-band and second low-frequency-band signals from the second microphone signal. A first measurement calculator determines a high-frequency-band energy value from the first high-frequency-band signal and the second high-frequency-band signal when the first and second high-frequency-band signals' magnitudes exceeds predetermined thresholds. A second measurement calculator calculates a low-frequency-band energy value from the first low-frequency-band signal and the second low-frequency-band signal when the first and second low-frequency-band signals' magnitudes exceed predetermined thresholds.

Abstract: A listening device for processing an input sound to an output sound, includes an input transducer for converting an input sound to an electric input signal, an output transducer for converting a processed electric output signal to an output sound, a forward path being defined between the input transducer and the output transducer and including a signal processing unit for processing an input signal in a number of frequency bands and an SBS unit for performing spectral band substitution from one frequency band to another and providing an SBS-processed output signal, and an LG-estimator unit for estimating loop gain in each frequency band thereby identifying plus-bands having an estimated loop gain according to a plus-criterion and minus-bands having an estimated loop gain according to a minus-criterion.

Abstract: Disclosed is an apparatus and method for processing signals such as sound signals. The sound processing apparatus includes a sound signal input unit for receiving sound signals, a harmonic noise separator for separating a harmonic region and a noise region from the received sound signals, a noise restraint index determination unit for determining an optimal noise restraint index k according to a system and circumstance, and a noise restrainer for restraining the separated noise region depending on the noise restraint index k so as to output noise attenuated signals.

Abstract: A dereverberation device includes a reverberation estimation unit for estimating a later reflection component by using information on an impulse response from a signal source to an observation point, a noise estimation unit, and a mixing unit. As a result, it is possible to obtain a high-quality dereverberated signal with a small calculation amount even in a noisy environment.

Abstract: A hands-free unit comprises a noise tolerant audio sensor to generate a first audio signal based on detection of audible sounds and an external audio sensor to generate a second audio signal based on detection of the audible sounds. A tunable distortion reduction filter adds high frequency information to the first audio signal and reduces distortion. A control unit detects noise levels based on comparison of first and second audio signals; and selects one of the first and second audio signals based on the detected noise level.

Abstract: A transient signal encoding method and device, decoding method and device, and processing system, where the transient signal encoding method includes: obtaining a reference sub-frame where a maximal time envelope having a maximal amplitude value is located from time envelopes of all sub-frames of an input transient signal; adjusting an amplitude value of the time envelope of each sub-frame before the reference sub-frame in such a way that a first difference is greater than a preset first threshold, in which the first difference is a difference between the amplitude value of the time envelope of each sub-frame before the reference sub-frame and the amplitude value of the maximal time envelope; and writing the adjusted time envelope into bitstream.

Abstract: Apparatus for generating a first acoustic signal and simultaneously sensing a second acoustic signal. The apparatus comprises: an input (15) for receiving a first electrical signal from a signal source; a loudspeaker terminal, directly or indirectly connected to the input (15), for connection to a loudspeaker (30) for generating the first acoustic signal in response to the first electrical signal and for generating a second electrical signal in response to the second acoustic signal; and an output (35), for outputting the second electrical signal. The loudspeaker terminal is connected to the output (35) via isolation means (25) comprising a first filter which is adapted to suppress a signal component related to the first electrical signal while simultaneously passing the second electrical signal to the output (35).

Abstract: Disclosed herein are detectors of audio ringing feedback, that is decaying feedback with a gain of less than one, those detectors utilizing a repeated gain measurement that applied to a range of gain values characteristic of ringing-type feedback. Those gain measurements, while in the range, increase a probability measurement of feedback. When the probability of feedback reaches a threshold, a detection of feedback is made and feedback countermeasures, such as the application of a notch filter, may be applied. Optionally, the audio gain around likely frequencies of feedback may be enhanced for a time to increase the resolution of identification of a feedback frequency, which may be identified through an interpolative method. Repeated gain measurements may also identify building-type feedback. A ringing detector may include more than one range of detection, for example for building, strong-ringing and weak-ringing feedback.

Abstract: The present application relates to a hearing aid with suppression of wind noise wherein wind noise detection is provided involving only a single comparison of the input signal power level at first low frequencies with the input signal power level at frequencies that may include the first low frequencies whereby a computational cost effective and simple wind noise detection is provided. The determination of relative power levels of the input signal reflects the shape of the power spectrum of the signal, and the detection scheme is therefore typically capable of distinguishing music from wind noise so that attenuation of desired music is substantially avoided.

Abstract: A signal processing apparatus includes a signal transforming unit transforming a first sound signal obtained by dividing a sound signal represented by a time function and by the predetermined time width into a second sound signal represented by a frequency function; a calculating unit determining a third sound signal which has a reduced influence of the operation sound by using the second sound signal and a sound signal representing the operation sound; a correcting unit performing a correction, by setting a sound signal representing the target sound as a reference signal, on the third sound signal based on the reference signal; and a signal inverse transforming unit inverse-transforming a sound signal on which the correction is performed from the sound signal represented by the frequency function into the sound signal represented by the time function.

Abstract: A novel adaptive beamforming technique with enhanced noise suppression capability. The technique incorporates the sound-source presence probability into an adaptive blocking matrix. In one embodiment the sound-source presence probability is estimated based on the instantaneous direction of arrival of the input signals and voice activity detection. The technique guarantees robustness to steering vector errors without imposing ad hoc constraints on the adaptive filter coefficients. It can provide good suppression performance for both directional interference signals as well as isotropic ambient noise.

Abstract: An active noise reduction system is provided for receiving an audio input signal and a noise interference signal and calculating an audio broadcasting signal according to a Feedback Filtered-X Least-Mean-Square (FFXLMS) algorithm, wherein the FFXLMS algorithm optimizes a (convergence factor) ? so as to decrease the numbers of divisions operated by the active noise reduction system and increase the operation speed of the active noise reduction system.

Abstract: The invention relates to a method for assessing interfering noise in motor vehicles, according to which noise occurring during a predefined measuring time is divided into different frequency ranges, the changes in level relative to the background noise are determined within said frequency ranges, and the determined changes in level are evaluated.

Abstract: A method for processing an audio signal received through a microphone array coupled to an interfacing device is provided. The method is processing at least in part by a computing device that communicates with the interfacing device. The method includes receiving a signal at the microphone array and applying adaptive beam-forming to the signal to yield an enhanced source component of the signal. Also, an inverse beam-forming is applied to the signal to yield an enhanced noise component of the signal. The method combines the enhanced source component and the enhanced noise component to produce a noise reduced signal, where the noise reduced signal is a target voice signal. Then, monitoring an acoustic set-up associated with the audio signal as a background process using the adaptive beam-forming inverse beam-forming to track the target signal component, and periodically setting a calibration of the monitored acoustic set-up.

Abstract: Systems and methods provided herein decrease an idle channel noise floor and reduce power during an idle channel input for low power audio devices that include a digital pulse width modulation (PWM) amplifier having a noise shaper. An audio data signal is monitored for an idle channel condition. The noise shaper performs quantization of the audio data signal and uses noise shaper filter coefficients to shape noise resulting from the quantization. Predetermined values for the noise shaper filter coefficients are used to shape the noise resulting from quantization while the idle channel condition is not being detected. The values of the noise shaper filter coefficients are reduced so that the values move toward zeros, and the reduced values of the noise shaper filter coefficients are used to attenuate noise resulting from quantization, while the idle channel condition is being detected.

Abstract: A mixed audio separation system (100) which separates a specific audio from among a mixed audio (S100) includes a local frequency information generation unit (105) which obtains pieces of local frequency information (S103) corresponding to local reference waveforms (S102), based on the local reference waveforms (S102) and an analysis waveform which is the waveform of the mixed audio (S100). Each of the local reference waveforms (S102) (i) constitutes a part of a reference waveform for analyzing a predetermined frequency, (ii) has a predetermined temporal/spatial resolution and (iii) includes at least one of an amplification spectrum and a phase spectrum in the predetermined frequency.

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