34 results about "Adaptive feedback cancellation" patented technology

The present invention discloses an adaptive feedback cancellation based on inserted and / or intrinsic characteristics and matched retrieval, which is realized by an audio processing system used for processing the input sound to output sound and a method for estimating a feedback transfer function in the audio processing system. The audio processing system comprises the following components: an input transducer; an output transducer; a forward path which comprises a signal processing unit; an electric feedback loop which extends from an output side and an input side and comprises a feedback path estimation unit which is used for estimating a sound feedback transfer function from the output transducer to the input transducer; and an enhancement unit which is used for extracting the characteristic of the electrical signal of the forward path and providing the estimated characteristic output; wherein the feedback path estimation unit is suitable for estimating the sound feedback transfer function with the estimated characteristic output. The invention can minimize the feedback in an audio processing system and may e.g. be used in hearing aids, head sets, mobile phones, wearable / portable communication devices, etc.

An improved method for adaptively cancelling acoustic feedback in hearing aids and other audio amplification devices. Feedback cancellation is limited to a frequency band that encompasses all unstable frequencies. By limiting the bandwidth of the feedback cancellation signal, the distortion due to the adaptive filter is minimized and limited only to the unstable feedback regions. A relatively simple signal processing algorithm is used to produce highly effective results with minimal signal distortion.

A method and system for active noise cancellation is provided. The system employs subband processing, and preferably implements over-sampled filterbank. The system is applicable to adaptive noise cancellation, adaptive echo cancellation for portable listening devices, such as headsets and other similar listening devices.

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.

A hearing aid includes: a feedback monitor connected to the adaptive feedback canceller and configured to monitor a state of feedback and having an output providing an indication of the state of feedback; and a cue controller connected to the feedback monitor and the at least one adaptive cue filter, and configured to control, in response to an output of the feedback monitor, the at least one adaptive cue filter so that the difference between the output of the at least one ITE microphone and the combined output of the at least one adaptive cue filter is reduced.

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. The problem is solved in that the audio processing system comprises an input transducer for converting an input sound to an electric input signal and defining an input side, an output transducer for converting a processed electric output signal to an output sound and defining an output side, a forward path being defined between the input transducer and the output transducer, and comprising a signal processing unit adapted for processing an SPU-input signal originating from the electric input signal and to provide a processed SPU-output signal, and an electric feedback loop from the output side to the input side comprising a feedback path estimation unit for estimating an acoustic feedback transfer function from the output transducer to the input transducer, and a enhancement unit for estimating noise-like signal components in the electric signal of the forward path and providing a noise signal estimate output, wherein the feedback path estimation unit is adapted to use the noise signal estimate output in the estimation of the acoustic feedback transfer function. This has the advantage of providing an adaptive feedback cancellation system which is robust in situations with a high degree of correlation between the output signal and the input signal of an audio processing system, e.g. a listening device. The invention may e.g. be used in public address systems, entertainment systems, hearing aids, head sets, mobile phones, wearable / portable communication devices, etc.

A hearing aid comprises at least one microphone (M) for converting input sound into an input signal, a subtraction node for subtracting a feedback cancellation signal from the input signal thereby generating a processor input signal, a hearing aid processor (G) for producing a processor output signal by applying an amplification gain to the processor input signal, a receiver (R) for converting the processor output signal into output sound, an adaptive feedback cancellation filter for adaptively deriving the feedback cancellation signal from the processor output signal by applying filter coefficients, calculation means for calculating the autocorrelation of a reference signal, and an adaptation means for adjusting the filter coefficients with an adaptation rate, wherein the adaptation rate is controlled in dependency of the autocorrelation of the reference signal.

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.

A hearing aid comprises an input transducer (2), a subtraction node for subtracting a feedback cancellation signal from the electrical input signal thereby generating a processor input signal, a signal processor (3), an output transducer (4), a pair of equalization filters (7a, 7b) for selecting from the processor input and output signals a plurality of frequency band signals, a frequency equalization unit for frequency equalization for the selected frequency band signals, and an adaptive feedback estimation filter (5, 6) for adaptively deriving the feedback cancellation signal from the equalized frequency band signals. The equalization of selected frequency bands of the input signals of the adaptive feedback cancellation filter provides for an improved and in particular a faster adaption of the feedback cancellation. The invention further provides a method of reducing acoustic feedback of a hearing aid, and a hearing aid circuit.

Methods and systems for signal processing an audio signal in a hearing device to detect when feedback path change occurs and controlling an adaptive feedback canceler to remove the feedback are provided. The hearing device includes a receiver and a microphone. An exemplary method includes detecting whether a tonal signal is caused by a feedback path change by estimating a product of a subband error signal and a subband output signal generated in response to a subband audio input signal; estimating a fast metric based on the estimated product and estimating a slow metric; and applying or maintaining an adaptation rate to the adaptive feedback canceler of the hearing device, wherein the adaptation rate applied or maintained is selected based upon a value of the difference between the fast and slow metrics compared to a threshold value.

A personal audio device, such as a wireless telephone, includes noise canceling circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone may also be provided proximate the speaker to measure the output of the transducer in order to control the adaptation of the anti-noise signal and to estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit that performs the adaptive noise canceling (ANC) function also detects frequency-dependent characteristics in and / or direction of the ambient sounds and alters adaptation of the noise canceling circuit in response to the detection.

The invention discloses a sub-band adaptive feedback elimination method and system based on step length control. The method comprises the following steps: acquiring a microphone pickup signal d (n) and a loudspeaker output delay signal x (n); performing pre-emphasis processing, sub-band filtering processing and FFT conversion on the signal d (n) and the signal x (n) in sequence; performing frequency-domain filtering processing to obtain a frequency-domain difference signal E (f, k); updating a frequency-domain filter coefficient; after the frequency-domain difference signal E (f, k) is converted to a time domain through IFFT, obtaining a time-domain signal e (n) through sub-band synthesis filtering processing and signal de-emphasis processing in sequence; and performing frequency shift processing on the time-domain signal e (n) to obtain a loudspeaker output signal y (n). According to the method, a sub-band adaptive algorithm is used, a high effect can be achieved with the minimum filter length, meanwhile, a large amount of calculation is reduced, and the processing efficiency is improved.

A method and an audio processing system determine a system parameter, e.g. step size, in an adaptive algorithm, e.g. an adaptive feedback cancellation algorithm so as to provide an alternative scheme for feedback estimation in a multi-microphone audio processing system. A feedback part of the system's open loop transfer function is estimated and separated in a transient part and a steady state part, which can be used to control the adaptation rate of the adaptive feedback cancellation algorithm by adjusting the system parameter, e.g. step size parameter, of the algorithm when desired system properties, such as a steady state value or a convergence rate of the feedback, are given / desired. The method can be used for different adaptation algorithms such as LMS, NLMS, RLS, etc. in hearing aids, headsets, handsfree telephone systems, teleconferencing systems, public address systems, etc.

The invention relates to the technical field of acoustics, in particular to a method and system for adaptive acoustic feedback elimination based on transform domain. A method and system for adaptive acoustic feedback elimination based on the transform domain of the present invention, through audio signal acquisition, signal time-frequency conversion, self-tuning coefficient calculation, autocorrelation matrix calculation, matrix equation fast solution, frequency domain filtering, feedback The steps of elimination, filter coefficient update, and signal time-frequency inverse transformation realize adaptive acoustic feedback elimination. The invention realizes the elimination of acoustic feedback and can reduce the damage to the target audio signal, and can obtain better feedback elimination effect and maintain higher audio quality under the environment of dynamic change of the feedback path. This technology uses the principle of algorithm to estimate the transfer function of the feedback path to separate the target audio signal from the feedback signal, so as to achieve feedback elimination, and the specific type of feedback path, the number of various feedback paths, the target sound source and the feedback sound source It is independent of the spatial topology of the audio and does not produce large audio impairments.

The invention discloses a hearing aid device and a method for fitting a hearing aid device to compensate a user's hearing loss to reduce feedback. The present invention relates to a hearing aid device comprising a receiver and a microphone; wherein there is a feedback path from the receiver to the microphone; wherein the hearing aid device further comprises an adaptive feedback canceller adapted to reduce at least a portion of the feedback; and wherein the adaptive feedback The canceller includes a fixed filter for modeling the constant part of the feedback path and an adaptive filter for modeling the changing part of the feedback path; The acoustic environment in which the hearing device is put into use will provide a constant part to the fixed filter of the hearing aid. Hereby is achieved a hearing aid comprising a fixed filter comprising a constant part of the feedback path.

A signal processing device comprises: input transducer(s) configured to convert input(s) to an input signal; output transducer(s) configured to convert an output signal to output(s); a signal processing circuit configured to at least subtract a feedback estimation signal from the input signal to produce a feedback compensated signal; and an adaptive feedback estimator. The adaptive feedback estimator comprises processor(s) and machine readable medium(s) collectively comprising instructions configured to cause the processor(s) to: estimate feedback path characteristic(s); construct an adaptive feedback cancellation filter based at least in part on the feedback path characteristic(s); select a value for variable p in a diversity measure norm; compute an update rule for the adaptive feedback cancellation filter, the update rule based on the diversity measure norm; apply the update rule to the adaptive feedback cancellation filter; and generate the feedback estimation signal through employment of the adaptive feedback cancellation filter on the output signal.