Abstract: The invention relates to an audio processing system for processing an input sound to an output sound. The invention further relates to a method of estimating a feedback transfer function in an audio processing system. The object of the present invention is to provide an alternative scheme for minimizing feedback in audio processing systems.

Abstract: The invention concerns an audio system comprising a microphone, audio signal processing means, an output transducer and mans for detecting a possible feedback tone and the corresponding frequency of the feedback tone in the audio system between the output transducer and the microphone. According to the invention means for counteracting feedback are provided. Further, mans are provided for changing the phase of the audio signal at a given frequency.

Abstract: A digital filter includes at least a first and a second delayed summation line. One of the delay lines includes a warped finite impulse response (FIR) filter and the other line is a FIR filter. Midpoint elements from the first delayed summation line are used as input to the second delayed summation line. Output from the first delayed summation line is delayed with a delay corresponding to the total delay of the second delayed summation line and the delayed output from the first delayed summation line is added to output from the second delayed summation line to faun a new output.

Abstract: The invention relates to a hearing aid system with an electrical feedback cancellation path, for compensating acoustic feedback between an output transducer and an input transducer by subtracting an estimate of the acoustical feedback from a signal on the input side of the amplifier part, the electrical feedback cancellation path comprising an adaptive filter for providing a variable filtering function. The invention further relates to a method of compensating acoustic feedback in a hearing aid system and to its use. The object of the present invention is to provide an alternative scheme for estimating the acoustical/mechanical feedback in a hearing aid. The problem is solved in that the hearing aid system comprises a second electrical input signal consisting essentially of the direct part of said first electrical input signal (i.e.

Abstract: The invention regards a scheme for generating a probe noise signal to be used in an anti feedback system of an audio system. The audio system comprises e.g. a microphone for capturing an audio signal, an audio signal processor for adaptation of the audio signal and a receiver for generation of an audible signal. According to an embodiment of the invention, a noise signal is injected into the audio signal path between the microphone and the receiver and used for estimating acoustical feedback, the noise signal being generated by the following steps: converting a digitized audio signal to the frequency domain, in order to obtain a series of magnitude and phase values, changing the phase values such that the phase of the resulting signal becomes less correlated, preferably substantially un-correlated, to the original signal, converting the magnitude and phase back to a time domain signal using the changed phase values. The invention may e.g. be used in a hearing aid, a headset or a pair of headphones.

Abstract: Disclosed is a method of reducing feedback in a hearing aid adapted to be worn by a user, the method comprising the step of: receiving an audio input signal in an input transducer in the hearing aid; wherein the method further comprises the steps of: transforming the input signal into the frequency domain; dividing the audio signal into a plurality of frequency bands; determining a threshold frequency over which a plurality of upper frequency bands lies; multiplying each of the plurality of upper frequency bands by a random phase, thereby obtaining a plurality of phase randomized upper frequency bands; synthesizing the plurality of phase randomized upper frequency bands and the lower frequency bands to an output signal; transforming the output signal into the time-domain; and transmitting the output signal to an output transducer of the hearing aid.

Abstract: The invention relates to a hearing instrument for processing an input sound to an output sound according to a user's needs. The invention further relates to a method of estimating acoustic feedback in a hearing instrument. The object of the present invention is to reduce the impact of tonal components in the target input signal on the quality of the estimate of acoustic feedback.

Abstract: The invention relates to a listening device for processing an input sound to an output sound according to a user's needs. The invention further relates to a method of reducing acoustic feedback in a listening device and to the use of a listening device. The object of the present invention is to provide an alternative scheme for feedback estimation in a listening device.

Abstract: The invention relates to a hearing aid system comprising first and second spatially separated hearing instruments, the system being adapted for processing input sounds to output sounds according to a user's needs. The invention further relates to a method and use. The object of the present invention is to provide an alternative scheme for reducing the effect of acoustic feedback in a hearing aid system.

Abstract: The invention relates to a hearing instrument for processing an input sound to an output sound according to a user's needs. The invention further relates to a method of operating a hearing instrument and to use of a hearing instrument. The object of the present invention is to provide an alternative scheme for handling acoustic feedback in a hearing instrument.

Abstract: Disclosed is method of generating an audible signal in a hearing aid by estimating a weighting function of received audio signals, the hearing aid is adapted to be worn by a user; the method comprises the steps of: estimating a directional signal by estimating a weighted sum of two or more microphone signals from two or more microphones, where a first microphone of the two or more microphones is a front microphone, and where a second microphone of the two or more microphones is a rear microphone; estimating a direction-dependent time-frequency gain, and synthesizing an output signal; wherein estimating the direction-dependent time-frequency gain comprises: obtaining at least two directional signals each containing a time-frequency representation of a target signal and a noise signal; and where a first of the directional signals is defined as a front aiming signal, and where a second of the directional signals is defined as a rear aiming signal; using the time-frequency representation of the target signal a

Abstract: The invention relates to a listening system comprising a first input transducer for converting an input sound to an electrical input signal, the electrical input signal comprising a direct part and an acoustic feedback part, an output transducer for converting an electrical output signal to an output sound, a forward path being defined between the input and output transducer and comprising a signal processing unit, a feedback cancellation system for estimating acoustic feedback comprising an adaptive FBC filter arranged in parallel to the forward path, the adaptive FBC filter comprising a variable FBC filter part and an FBC update algorithm part for updating the variable FBC filter part, the FBC update algorithm part comprising first and second FBC algorithm input signals influenced by the electrical input and output signals, respectively, the first and second FBC update algorithm input signal paths comprising first and second variable filters, respectively, the listening system further comprising an electric

Abstract: A device for detecting the presence of wind noise in an array of microphones including two or more separate sound inlet openings, and a sound to electrical converting element or microphone in relation to each sound inlet opening. The microphones each generate time dependent signals which are fed to a signal processing device that provides one or more output signals. The signal processing device has means for generating a time dependent cross correlation function between a first and a second microphone signal, and means for generating a signal corresponding to a time dependent auto correlation function of either the first or the second of the microphone signals.