Abstract: A method of adaptively mapping a plurality of microphone signals to a multi-channel audio signal, for example to capture stereo audio from a device held in variable orientations. At least first and second microphone signals, from respective first and second spaced apart microphones, are obtained. A device orientation signal is also obtained from a device orientation sensor. The microphone signals are adaptively mapped, based on the device orientation signal, to produce a first audio signal channel of a multi-channel audio signal. The first and second microphone signals are also adaptively mapped, based on the device orientation signal, to produce a second audio signal channel of a multi-channel audio signal.

Abstract: A method of wind noise reduction. Left side and right side microphone signals are obtained. In a first stage wind noise reduction is applied to a first sub-band of one of the signals, below a spectral threshold NA. In a second stage the wind noise reduced first side signal, and the second side signal, are both split into a sub-band below a threshold NB less than NA. The sub-band of the first side signal is mixed with the sub-band of the second side signal to produce an aggregate third sub-band signal having reduced wind noise, which is recombined with the respective sub-bands above the threshold NB to produce output first and second side signals.

Abstract: This invention concerns a method, and a device, for feedback cancellation. This invention also concerns a computer program product comprising computer program code means to make a computer execute a procedure for feedback cancellation. The method comprises providing an adaptive feedback cancellation filter which adapts under the control of a control module, and filtering at least one input of the control module to suppress correlated signals from the input prior to the control module operating upon the input. The device comprises an adaptive feedback cancellation filter, a control module and at least one filter. The control module controls adaptation of the adaptive feedback cancellation filter. The filter suppresses correlated signals from an input to the control module prior to the control module operating upon the input.

Abstract: A method of processing digitized microphone signal data in order to detect wind noise. First and second sets of signal samples are obtained simultaneously from two microphones. A first number of samples in the first set which are greater than a first predefined comparison threshold is determined. A second number of samples in the first set which are less than the first predefined comparison threshold is determined. A third number of samples in the second set which are greater than a second predefined comparison threshold is determined. A fourth number of samples in the second set which are less than the second predefined comparison threshold is determined. If the first number and second number differ from the third number and fourth number to an extent which exceeds a predefined detection threshold, e.g. as determined by a Chi-squared test, then an indication that wind noise is present is output.

Abstract: Adaptive processing of an input signal is achieved by offline analysis, with inline processing comprising an adaptive filter. The method comprises passing the input signal through an adaptive time domain filter to produce an output signal. The input signal and/or output signal is used as an offline analysis signal. The analysis signal is transformed into a transform domain (eg frequency domain) to produce a transformed analysis signal. The transformed analysis signal is analyzed, for example by ADRO, to produce a plurality of desired gains each corresponding to a respective transform domain sub-band. A time domain filter characteristic is synthesized to at least approach the desired gains. The adaptive filter is updated with the synthesized filter characteristic. Minimum phase adaptive filter techniques are found to possess particular benefits in this scheme.

Abstract: Adaptively processing an input signal, such as an input signal of a hearing aid. The input signal is passed through an adaptive time domain filter to produce an output signal. At least one of the input signal and the output signal is used as an analysis signal. The analysis signal is transformed into a transform domain to produce a transformed analysis signal, which is analyzed to produce a desired gain for each respective transform domain sub-band. A minimum phase time domain filter characteristic is synthesized which approaches the desired gains. The adaptive filter is updated with the synthesized filter characteristic.

Abstract: The invention relates to adaptive directional systems, and more particularly to a method and apparatus for producing adaptive directional signals. The invention may be applied to the provision of audio frequency adaptive directional microphone systems for devices such as hearing aids and mobile telephones. The method involves constructing the adaptive directional signal (46) from a weighted sum of a first signal (42A) having an omni-directional polar pattern and a second signal (42B) having a bi-directional polar pattern, wherein the weights are calculated to give the combined signal a constant gain in a predetermined direction and to minimize the power of the combined signal. The method has particular application in producing signals in digital hearing aids, the predetermined direction being in the forward direction with respect to the wearer.