Abstract: A hearing aid includes a hearing aid housing enclosing a microphone for converting sound into an audio signal, first feedback compensation means for providing a first feedback compensation signal of signals picked up by the microphone by modelling an internal mechanical feedback signal path of the hearing aid, second feedback compensation means for providing a second feedback compensation signal by modelling an external feedback signal path of the hearing aid, subtracting means for subtracting the first and second feedback compensation signals from the audio signal to form a compensated audio signal, processing means, connected to an output of the subtracting means, for processing the compensated audio signal, and a receiver, connected to an output of the processing means, for converting the processed compensated audio signal into a sound signal.

Abstract: An acoustic module for a hearing aid device has at least one microphone and at least one earphone (acoustic output transducer) combined together as a unit, the unit including a signal-processing unit for feedback suppression that is connected with the microphone and the earphone. Feedback signal portions are removed from the microphone signal by the signal-processing unit. The acoustic module places acousto-mechanical components within a unitary component, with acousto-mechanical characteristics that can be measured and included in the signal processing for feedback suppression.

Abstract: A hearing assistance system having adjustable bulk delay for cancellation of a time varying acoustic feedback path. The hearing assistance system including an FIR filter, coefficient update module, and delay rules module for programmable adaptive filtering. The hearing assistance system adjustable for continuous bulk delay adjustments. The hearing assistance system providing a number of coefficient update routines, including, but not limited to an LMS coefficient update process and a normalized LMS coefficient update process.

Abstract: A method of signal processing an input signal in a hearing aid to avoid entrainment, the hearing aid including a receiver and a microphone, the method comprising using an adaptive filter to measure an acoustic feedback path from the receiver to the microphone and adjusting an adaptation rate of the adaptive filter using an output from a filter having an autoregressive portion, the output derived at least in part from a ratio of a predictive estimate of the input signal to a difference of the predictive estimate and the input signal.

Abstract: Method and apparatus for signal processing an input signal in a hearing assistance device to avoid entrainment, the hearing assistance device including a receiver and a microphone, the system comprising using a gradient adaptive lattice filter including one or more reflection coefficients to measure an acoustic feedback path from the receiver to the microphone of the hearing assistance device.

Abstract: In a feedback compensation method and a feedback compensator in an acoustic amplification system such as a hearing aid, an adaptive feedback compensation filter generates a compensation signal from the amplified output signal, and one or more filters restrict the frequency range in which the compensation signal is generated. These filters are adaptable with regard to their filter function during the operation of the feedback compensator. The adaptation ensues with an analysis and control unit that checks the frequency range affected by the feedback and adapts the filter functions of the filters to it. The checking ensues, for example, by a comparison of the filter function of the feedback compensation filter with the filter functions of the filters to restrict the frequency range, or with the use of an oscillation detector.

Abstract: A system of signal processing an input signal in a hearing assistance device to avoid entrainment wherein the hearing assistance device including a receiver and a microphone, the method comprising using an adaptive filter to estimate an acoustic feedback path from the receiver to the microphone, generating one or more estimated future pole positions of a transfer function of the adaptive filter, analyzing stability of the one or more estimated pole positions for an indication of entrainment and adjusting the adaptation of the adaptive filter based on the stability.

Abstract: A system of signal processing an input signal in a hearing aid to avoid entrainment, the hearing aid including a receiver and a microphone, the method comprising using a transform domain adaptive filter including two or more eigenvalues to measure an acoustic feedback path from the receiver to the microphone, analyzing a measure of eigenvalue spread against a predetermined threshold for indication of entrainment of the transform domain adaptive feedback cancellation filter, and upon indication of entrainment of the transform domain adaptive feedback cancellation filter, modulating the adaptation of the transform domain adaptive feedback cancellation filter.

Abstract: A mobile wireless communications device includes a housing and circuit board carried by the housing. Radio frequency (RF) circuitry carried by the circuit board forms a transmitter and receiver. A processor is carried by the circuit board and operative with the RF circuitry. A keyboard connector is mounted on the circuit board. A plurality of Key_Out and Key_In connection lines are carried by the circuit board and connect the keyboard connector and processor. A keypad is carried by the housing and operatively connected to the keyboard connector. A filter is operative with each connection line for limiting degradation in receiver sensitivity.

Abstract: Improved approaches of designing and fitting hearing aids to make hearing aids more accessible to people with hearing loss are disclosed. The hearing aids can be capable of being fitted by users themselves or by other non-hearing specialists. In one embodiment a hearing aid can be self-calibrating. In another embodiment, hearing aids can be fitted, i.e., configured, for individuals with hearing loss using a simplified procedure that hearing aid users or non-hearing specialists can easily follow.

Abstract: Acoustical signals impinging on an input converter are converted into a first electric signal by a controllably variable transfer characteristic. The transfer characteristic is dependent on the angle at which the acoustical signals impinge on the input converter. The first electric signal is processed and a resulting signal is applied to an output converter. Feedback to be suppressed is compensated by a feedback compensating signal, which is generated in dependency of the resulting signal and is fed back by a feedback signal path upstream of the processing. The electric feedback compensating signal is fed back to and superimposed upon the first electric signal. An adaptation rate of the converting into a first electric signal by a controllably variable transfer characteristic is controlled in dependency of the loop gain along the feedback signal path.

Abstract: The determination of a suitable step size for controlling the adaptation of a filter of an acoustic feedback compensator especially for hearing aids is to be improved. A step size controller is therefore provided in which an input signal is analyzed in at least two frequency bands. In doing so, any steep signal edges in the individual frequency bands are detected. The adaptation step size of an adaptive filter is then controlled depending on the number of frequency bands in which steep signal edges are detected, wherein the magnitude of the signal edges can be included in the decision.

Abstract: A hearing device is activated by operating the hearing device in an operating mode in a first phase after supplying energy to internal circuitry of the hearing device, by operating the hearing device in a reduced operating mode in a second phase and by operating the hearing device in a selectable operating mode in a third phase. Thereby, the hearing device user can examine a correct functioning of the hearing device before its insertion into the ear. In addition, the hearing device can thereafter be inserted without that a feedback signal is generated.

Abstract: The invention relates to oscillation detection and, more particularly, concerns a method and apparatus for identifying oscillation in a signal due to feedback, permitting appropriate action to be taken to suppress the oscillation. The method involves using an FFT device or similar to convert a signal at each of a series of successive time windows into the frequency domain, calculating, for each of a plurality of frequency bands, the change in signal phase from a time window to a subsequent time window, and comparing, for some or all of said frequency bands, the results of the calculation step to one or more defined criteria to provide a measure of whether oscillation due to feedback is present in the signal. For additional discrimination, the change in signal amplitude from a time window to a subsequent time window may also be calculated for each of the frequency bands, and the result compared with one or more further defined criteria. The invention has particular application in hearing aid devices.

Abstract: To enable hearing apparatus feedback to be reliably detected, it is provided that the hearing apparatus has an analyzer for analyzing the resonant behavior of the overall system as a function of a modification of the signal processing device and for determining from the analysis result a feedback variable constituting a measure of the feedback. On the basis of the feedback variable, an adaptive compensation filter, for example, can then be step-size-controlled to compensate the feedback.

Abstract: 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.

Abstract: Fine customization of a hearing aid to the individual hearing environments and habits of a user are to be simplified and improved. To this end, when the hearing aid is turned on or when the hearing aid is switched to a particular operating mode, neither the value most recently valid for the parameter prior to turning off or switching to a different operating mode nor the parameter value transferred to the hearing aid at the beginning of programming is set. Rather, from the changes to the value of the parameter taking place during operation of the hearing aid a new start value is ascertained and stored, which value is then set automatically after turning on or switching mode.

Abstract: The microphones used in hearing devices normally possess different characteristic lines that are to be adapted to one another. For this purpose, the amplitude of an output signal of a first microphone and the amplitude of an output signal of a second microphone are measured. The output signal of the first microphone is subsequently filtered dependent on both measured amplitudes, such that the difference between the two output signals is reduced. One of the two microphones hereby serves as a reference, and an absolute normalization can be foregone.

Abstract: A digital hearing aid system includes a signal path with an input transducer, a signal processor and an output transducer, a part of the system being intended for delivering sound into an ear canal of a hearing aid user leaving the ear canal with an unobstructed cross-sectional area corresponding to a vent channel with a diameter of at least 3 mm or a total area of at least 7.07 mm2. The signal path is designed to have a signal delay less than 15 ms. The hearing aid signal path furthermore preferably includes means for providing an adaptive feedback compensation. The signal processor is adjusted to provide increased gain in low frequency areas.

Abstract: Improved approaches to adaptively suppress interfering noise in a multi-microphone directional system are disclosed. These approaches operate to adapt the direction null for the multi-microphone directional system in accordance with a dominant noise source.

Abstract: In a hearing aid, as well as in a method for the operation of a hearing aid having a microphone system in which different directional characteristics can be set, the tonal quality is improved, particularly in a quiet hearing environment, by the signal delay for at least one microphone signal being increased so as to increase the transfer function in the frequency response of the microphone system, thus also improving the signal-to-noise ratio, by decreasing the proportion of the microphone noise in the microphone output signal.

Abstract: Feedback reduction in hearing aids is simplified and improved in an amplification device that has two separate compensation paths, an acoustic input and an inductive input. In the first compensation path, a first filter device acts to compensate acoustic feedback, and in the second compensation path, a second filter device acts to compensate inductive feedback. The two very different feedbacks can be individually compensated without increased outlay.

Abstract: A system and method to compensate for changes in the frequency response of a microphone caused by factors interfering with the receipt of acoustic sound in the microphone. The system includes at least a microphone and a signal processor. The signal processor is operational to process at least one feedback frequency response from the microphone to generate at least one test parameter. The signal processor uses the at least one test parameter to determine at least one operational characteristic of the microphone. The feedback frequency response is generated by the microphone in response to acoustic feedback. The acoustic feedback is generated by actuation of a transducer in response to at least one test signal that is provided to the transducer. The signal processor uses the at least one test parameter to process acoustic frequency responses from the microphone to compensate for changes in the acoustic frequency responses of the microphone.

Abstract: A multi-channel digital hearing instrument is provided that includes a microphone, an analog-to-digital (A/D) converter, a sound processor, a digital-to-analog (D/A) converter and a speaker. The microphone receives an acoustical signal and generates an analog audio signal. The A/D converter converts the analog audio signal into a digital audio signal. The sound processor includes channel processing circuitry that filters the digital audio signal into a plurality of frequency band-limited audio signals and that provides an automatic gain control function that permits quieter sounds to be amplified at a higher gain than louder sounds and may be configured to the dynamic hearing range of a particular hearing instrument user. The D/A converter converts the output from the sound processor into an analog audio output signal. The speaker converts the analog audio output signal into an acoustical output signal that is directed into the ear canal of the hearing instrument user.

Abstract: A feedback compensator for a hearing aid device has a filter arrangement that splits a signal path, to implement an adaptive feedback compensation with only one buffer memory, two splitting nodes, and two addition nodes. The feedback compensation ensues only in the feedback-susceptible frequency range of the input signal. In addition to the filtering of the input signal, it is advantageous to feed to the adaptive feedback compensation filter a bandwidth-limited signal that is taken from the amplified output signal.

Abstract: The invention relates to a method for reducing feedback caused by electromagnetic interference between an induction coil and an output transducer, a voltage supply or an amplifier or a combination thereof, the method comprising: producing by means of a filter an equivalent of an electromagnetic feedback path in a system with a pickup coil and subtraction of the equivalent of the feedback signal from the input signal in order to obtain feedback reduction. The invention further relates to a device for implementing the method, the device comprising an induction coil and an output transducer, voltage supply or an amplifier or a combination thereof. The device further comprises filter means for generating an equivalent of an electromagnetic feedgack signal occurring between the induction coil and an output transducer, a voltage supply or an amplifier or a combination thereof and means for subtracting the equivalent from an input signal in order to obtain feedback reduction.

Abstract: A method for canceling feedback in an acoustic system including a microphone, a signal path, a speaker and means for detecting presence of feedback between the speaker and the microphone, the method including providing an LMS algorithm for processing the signal; where the LMS algorithm operates with a predetermined adaptation speed when feedback is not present; where the LMS algorithm operates an adaptation speed faster than the predetermined adaptation speed when feedback is present, and where the means for detecting the presence of feedback is used to control the adaptation speed selection o the LMS algorithm, where the feedback detection means comprises bandwidth detection means for determining the presence of a feedback signal.

Abstract: A system and method for adaptively removing feedback in audio systems and, more particularly, hearing aid systems. The audio system comprises an analysis unit, for receiving an input signal and providing N bandpass input signals, and an adaptive feedback cancellation unit for removing a feedback condition in one or more of the N bandpass input signals. The adaptive feedback cancellation unit comprises N sub-units with at least one of the sub-units including: (i) a feedback detector for indicating the presence of the feedback condition in one of the bandpass input signals, (ii) an adaptive feedback canceller for providing an adaptive gain modification factor for adjusting gain when the feedback condition is detected within one of the bandpass input signals, and (iii) a multiplier coupled to the adaptive feedback canceller and the analysis unit for providing a bandpass output signal based on one of the bandpass input signals and the corresponding adaptive gain modification factor.

Abstract: A method for operating a digital, programmable hearing aid is provided that uses both a transmission characteristic of a normal amplification as well as a transmission characteristic of a maximum amplification of an audio signal over a frequency range that can be nearly freely configured. Given a modification of the amplification by settings at the hearing aid as well as using parameters that result from the signal processing, the gain for the overall system is always calculated utilizing all parameters and is potentially limited to the maximum amplification at the respective frequency if this would otherwise be exceeded.

Abstract: A precision, low jitter oscillator circuit is provided that is particularly well-suited for generating a clock signal in miniature digital systems, such as digital hearing aids. The oscillator includes a plurality of differential inverters configured in a feedback loop to generate an oscillating clock signal. The differential inverters include a capacitive trimming network for adjusting the frequency of the oscillating clock signal and employ resistive loads for minimizing jitter in the clock signal. The components of the oscillator are fabricated in a common silicon process to minimize the size of the oscillator.

Abstract: A new subband feedback cancellation scheme is proposed, capable of providing additional stable gain without introducing audible artifacts. The subband feedback cancellation scheme employs a cascade of two narrow-band filters Ai(Z) and Bi(Z) along with a fixed delay, instead of a single filter Wi(Z) and a delay to represent the feedback path in each subband. The first filter, Ai(Z), is called the training filter, and models the static portion of the feedback path in ith subband, including microphone, receiver, ear canal resonance, and other relatively static parameters. The training filter can be implemented as a FIR filter or as an IIR filter. The second filter, BI(Z), is called a tracking filter and is typically implemented as a FIR filter with fewer taps than the training filter. This second filter tracks the variations of the feedback path in the ith subband caused by jaw movement or objects close to the ears of the user.

Abstract: For reducing feedback-conditioned oscillations in a hearing aid device, microphone signals of a first microphone and of a distanced, second microphone are compared to one another. When oscillations are detected at the same frequency in both microphone signals, these oscillations are determined to be useful (non-feedback) tonal signals. Oscillations that are only present in one of the microphone signals, in contrast, are feedback-conditioned and are suppressed using suitable measures.

Abstract: A feedback threshold in a hearing device is determined by feeding an input signal to the hearing device while it is inserted in an ear canal of a user. The input signal results in a higher amplification than a supposed, that is, not yet known, feedback threshold. An amplification in the forward path of the device is measured. The feedback threshold is equal to the measured amplification.

Abstract: A feedback compensator for a hearing aid device has a filter arrangement that splits a signal path, to implement an adaptive feedback compensation with only one buffer memory, two splitting nodes, and two addition nodes. The feedback compensation ensues only in the feedback-susceptible frequency range of the input signal. In addition to the filtering of the input signal, it is advantageous to feed to the adaptive feedback compensation filter a bandwidth-limited signal that is taken from the amplified output signal.

Abstract: In the case of a hearing aid device which can be worn in the ear or a hearing aid device with a otoplastic which can be worn in the ear, the penetration of direct sound through a ventilation channel of the hearing aid device or of the otoplastic is prevented. An acoustic signal is picked up from the ventilation channel via a second microphone in a first region of the ventilation channel and phase-shifting it in a filter device in such a way that the direct sound is at least largely eliminated after the phase-shifted signal is emitted into the ventilation channel via a second earphone. The invention offers the advantage that an enlargement of the cross section of the ventilation channel is made possible as a result, even to provide an open supply, without disturbing direct sound getting into the auditory canal of the person wearing the hearing device.

Abstract: The present invention relates to a hearing aid with an adaptive filter for suppression of acoustic feedback in the hearing aid. The hearing aid further comprises a controller that is adapted to compensate for acoustic feedback by determination of a first parameter of an acoustic feedback loop of the hearing aid and adjustment of a second parameter of the hearing aid in response to the first parameter whereby generation of undesired sounds is substantially avoided. Hereby a gain safety margin requirement is significantly reduced.

Abstract: A method for operating a hearing device (1) including the extraction, during an extraction phase, of characteristic features from an acoustical signal captured by at least one microphone (2a, 2b), and the processing, during an identification phase and with the aid of Hidden Markov Models, of the characteristic features especially for the determination of a momentary acoustic scene or of sounds and/or for voice and word recognition. A hearing device is also specified.

Abstract: A noise reduction system for a sound reproduction system, in particular for hearing aids, comprises a primary and a secondary microphone for producing input signals in response to sound in which a noise component is present. The system has a first signal processing section comprising a fixed filter and a summing function, wherein the first signal processing section has means for receiving signals from the microphones and producing a speech reference signal and a noise reference signal. A second signal processing section comprises an adaptive filter and an additional summing function, and the second signal processing section has means for receiving the speech and noise reference signals and producing an output signal with an improved signal-to-noise ratio.

Abstract: In a method for operating a hearing aid or hearing aid system, and a hearing aid or hearing aid system, wind noises are detected by analyzing the output signals of at least two microphones. If wind noises are present, the signal processing unit of the hearing aid or hearing aid system and/or the signal paths of microphones are adapted in order to reduce such noises.

Abstract: 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.

Abstract: The invention relates to oscillation suppression and, more particularly, concerns a method and apparatus for suppressing oscillation in a signal identified as or suspected of containing an oscillation due to feedback.

Abstract: A feedback cancellation system with reduced sensitivity to low-frequency tonal inputs is provided. Such a system can be used, for example, in a hearing aid to prevent cancellation of the desired tonal inputs to the hearing aid, thus improving the gain at high frequencies of the hearing aid while simultaneously preserving the desired tonal inputs at low frequencies. The feedback cancellation system comprises a first adaptive filter block for adaptively filtering an error signal to remove the low-frequency tonal components from the error signal. The first adaptive filter block is constrained so that only low-frequency tones in the error signal are cancelled, thus enabling the feedback cancellation system to still cancel “whistling” at high frequencies due to the temporary instability of the hearing aid. A second adaptive filter block adaptively filters a feedback path signal to produce an adaptively filtered feedback path signal.

Abstract: The invention relates to oscillation detection and, more particularly, concerns a method and apparatus for identifying oscillation in a signal due to feedback, permitting appropriate action to be taken to suppress the oscillation.

Abstract: An electronic earplug for wideband control of pressures at the tympanic membrane is presented. A unique methodology of determining effective component placement inside an earplug that provides acoustic isolation between the ambient noise and tympanic membrane is explained. Methods for providing accurate dosimetry and improved active control result from the unique earplug design process, leading to very wideband active noise reduction at the tympanic membrane.

Abstract: A method and apparatus for increasing the intelligibility of speech signals received by mobile stations via adapting some of the acoustic parameters of speech in the frequency domain. The mobile station may have the capability to modify speech signal parameters to conform the speech signal to a listener's hearing profile. In an alternative embodiment, mobile station processing and battery power are conserved by the communication network having the capability to modify speech signal parameters to conform the speech signal to a listener's hearing profile. To conserve mobile station memory, the user's hearing profile may be stored on the network.

Abstract: A hearing aid (1) with microphone (2), an electrical or electronic circuit (3,6,12) for the processing of the signal from the microphone, and a sound transducer (13) for the generation of an acoustic output signal, and with an arrangement for switching the hearing aid on and off and an external storage unit (14) coupled together via a data bus (5,17), further comprising means which generate a special acoustic signal during or at least in part of a comfort delay period.

Abstract: The invention relates to a method of managing acoustic feedback in a hearing aid, characterised by using a multi (two or more) channel type of filter hereby splitting the frequency range up in two or more separate parts; determining the maximum allowable gain in at least one frequency range before feedback occurs; monitoring the volume control in such a way, that if the maximum allowable gain before feedback occurs is reached, the gain is manipulated for the offending frequency range. The determination of maximum allowable gain and the monitoring of the volume control is applied for several channels. The invention further relates to hearing aid implementing the method according to the method according to the invention.

Abstract: The present invention relates to a hearing aid with an adaptive filter for suppression of acoustic feedback in the hearing aid. The hearing aid further comprises a controller that is adapted to compensate for acoustic feedback by determination of a first parameter of an acoustic feedback loop of the hearing aid and adjustment of a second parameter of the hearing aid in response to the first parameter whereby generation of undesired sounds is substantially avoided. Hereby a gain safety margin requirement is significantly reduced.

Abstract: A new subband feedback cancellation scheme is proposed, capable of providing additional stable gain without introducing audible artifacts. The subband feedback cancellation scheme employs a cascade of two narrow-band filters Ai(Z) and Bi(Z) along with a fixed delay, instead of a single filter Wi(Z) and a delay to represent the feedback path in each subband. The first filter, Ai(Z), is called the training filter, and models the static portion of the feedback path in ith subband, including microphone, receiver, ear canal resonance, and other relatively static parameters. The training filter can be implemented as a FIR filter or as an IIR filter. The second filter, BI(Z), is called a tracking filter and is typically implemented as a FIR filter with fewer taps than the training filter. This second filter tracks the variations of the feedback path in the ith subband caused by jaw movement or objects close to the ears of the user.

Abstract: A feedback cancellation system for a hearing instrument is described. The feedback cancellation system includes both a constrained adaptive filter, the constrained adaptive filter can be constrained by initiation coefficients, as well as an adaptive gain modification. The adaptive gain modification allows the feedback cancellation to respond to substantially different environments from the test environment used to derive the initiation coefficients for the constrained adaptive filter.