Abstract: A method for frequency transposition in a communication device Or a hearing device, respectively, is disclosed by transforming an acoustical signal into an electrical signal (s) and by transforming the electrical signal from time domain into frequency domain to obtain a spectrum (S). A frequency transposition is being applied to the spectrum (S) in order to obtain a transposed spectrum (S′), whereby the frequency transposition is being defined by a nonlinear frequency transposition function. Thereby, it is possible to transpose lower frequencies almost linearly, while higher frequencies are transposed more strongly. As a result thereof, harmonic relationships are not distorted in the lower frequency range, and at the same time, higher frequencies can be moved to a lower frequency range, namely to an audible frequency range of the hearing impaired person.

Abstract: Sounds input via a sound input device are corrected in accordance with age-based hearing characteristic data read from a memory or correction values for reference hearing characteristics for individual hearing characteristics-measured by an individual hearing characteristics measurement device, and the sounds corresponding to the hearing characteristics are measured and displayed by the display.

Abstract: System for enhancing the hearing of certain sounds, the system including an electro-acoustic transducer for producing sounds in the Vicinity of an ear, according to signals provided thereto, and a compensatory signal generator coupled with the electro-acoustic transducer, the compensatory signal generator producing a compensatory signal, according to at least a portion of a compensatory waveform, the compensatory waveform being determined according to ear otoacoustic emissions, the compensatory signal being employed to enhance the hearing, the compensatory signal generator providing the compensatory signal to the electro-acoustic transducer.

Abstract: A method and system for improving auditory processing and increasing spatial awareness, wherein the voices of a musical composition are arranged, recorded and produced in particular combinations and at particular frequency levels according to a predetermined program designed to exercise auditory muscles, imprint the frequency map, and stimulate dendritic growth and synaptic organization.

Abstract: A method for producing an approximated partial transfer function can be used in an electroacoustic appliance for producing an environment correction transfer function that matches an appliance transfer function for the electroacoustic appliance to an acoustic environment, by a) providing a number of basic functions, which each have one basic characteristic of a spectral profile of partial transfer functions, b) providing the approximated partial transfer function by combination of the basic functions weighted by weighting factors, in that at the weighting factor is in each case determined for each basic function such that operation of the electroacoustic appliance is matched to an acoustic environment taking into account the approximated partial transfer function which is formed by the weighting factors and the basic functions, and c) storing the approximated partial transfer function in the electroacoustic appliance for use during operation.

Abstract: A multichannel hearing aid (20) comprises at least one frequency channel having a compressor (38) with a compression threshold at an output level below the hearing threshold and an attack time above 0.5 seconds whereby hearing of a sudden sound in a stationary sound environment is facilitated. With this compressor, the amplification of low signal levels may be increased compared to the prior art, as the compressor kicks in to generally suppress steady noises. The gain may generally be increased as high as feasible in view of the microphone baseline noise, which should preferably be kept below the hearing threshold. Thus the user of the hearing aid will generally have the option of a higher gain of low level sounds than generally feasible with prior art hearing aids.

Abstract: A signal processing system, such as a hearing aid system, adapted to enhance binaural input signals is provided. The signal processing system is essentially a system with a first signal channel having a first filter and a second signal channel having a second filter for processing first and second channel inputs and producing first and second channel outputs, respectively. Filter coefficients of at least one of the first and second filters are adjusted to minimize the difference between the first channel input and the second channel input in producing the first and second channel outputs. The resultant signal match processing of the signal processing system gives broader regions of signal suppression than using the Wiener filters alone for frequency regions where the interaural correlation is low, and may be more effective in reducing the effects of interference on the desired speech signal.

Abstract: In a method and a device for the signal processing in a hearing aid, in which coefficients of a filter for the frequency-dependent amplitude adaptation of an input signal are adapted in accordance with this input signal, the following steps are carried out:

Abstract: The present invention is a hearing aid for amplifying an acoustic signals comprising: a controller for determining in real time a frequency band at the highest level of the acoustic signals through frequency analysis of the acoustic signals that vary over time, and for generating a control signal to raise a gain for signals of a higher frequency range than the frequency band at the highest level (such as an amplifier Q3, or a band-pass filter group 2 and a diode matrix 3 and a comparator 4, or a digital signal processor 13, or the like); and a first amplifier, in which the control signal from said controller is inputted so that the frequency characteristics are varied, for amplifying the acoustic signals by increasing the gain for signals of the higher frequency range than the frequency band at the highest level (such as an amplifier system consisting of amplifiers Q1 and Q2, or a parametric equalizer 5, or a digital signal processor 13, or the like).

Abstract: In a hearing aid having a microphone and an earphone with a signal processing unit connected therebetween, and a method for operating such a hearing aid, a filter element in the signal processing unit splits the signal from the microphone into a number of partial signals, and each of the partial signals is analyzed, to determine whether they respectively contain speech information, by comparing the magnitude of the envelope of the partial signal to the average value of the partial signal. For any partial signal having an envelope magnitude which exceeds the average value, the analysis unit operates an amplification stage to amplify that partial signal before it proceeds to the earphone.

Abstract: For improving the automatic switching and control of hearing aid devices in view of respective auditory situation, the acoustic signal picked up by the hearing aid device is analyzed for noise signals. The control of the hearing aid device then ensues on the basis of the analyzed noise signals. Individual transmission parameters or entire hearing programs of the hearing aid device can be controlled or switched.

Abstract: A tinnitus method and device for providing relief to a person suffering from the disturbing effects of tinnitus is described. The method can be implemented entirely in software to spectrally modify an audio signal in accordance with a predetermined masking algorithm which modifies the intensity of the audio signal at selected frequencies. A predetermined masking algorithm is described which provides intermittent masking of the tinnitus wherein, at a comfortable listening level, during peaks of the audio signal the tinnitus is completely obscured, whereas during troughs the perception of the tinnitus occasionally emerges. In practice it has been found that such intermittent masking provides an immediate sense of relief, control and relaxation for the person, whilst enabling sufficient perception of the tinnitus for habituation and long term treatment to occur. Advantageously the predetermined masking algorithm is specifically tailored to the audiometric configuration of the person.

Abstract: A hearing aid with a directional characteristic, including at least two spaced apart input transducers and wherein transducer signal type, such as transducer noise, wind noise, sound emitted from a sound source located in the surroundings of the hearing aid, distorted signals, such as clipped signals, slew rate limited signals, etc, is determined, and wherein signal processing in the hearing aid, such as transducer matching, filtering, signal combination, etc, is adapted according to the determined signal type. For example, the directional characteristic may be switched to an omnidirectional characteristic when at least one of the input transducer signals is dominated by noise or distortion, and/or adaptive matching of input transducers may be put on hold while at least one of the input transducer signals is dominated by noise or distortion.

Abstract: A system and method upper auditory frequency range hearing. A speech signal is filtered, and then modulated to an upper auditory frequency range. The modulated signal is then provided to a transducer, which causes a vibration in the ear canal, the head or the neck of a user, which is received in an inner ear of the user. That vibration is translated as a signal to the brain, which interprets that signal as intelligible speech.

Abstract: In one embodiment an apparatus for processing sound includes a means (401) for analyzing a sound signal into a number frequency bands and a means (403) for applying variable gain to each frequency band independently. Gain is applied under control of a number of gain comparator means (409) each of which generates a number of statistical estimates in respect of each signal and compares those estimates to predetermined hearing response parameters stored in memory (411). The numerous gain compensated frequency bands are then combined (415) in order to generate a single sound signal. The apparatus may be implemented in dedicated hardware embodiment or by software running on a microprocessor.

Abstract: The hearing instrument comprises a circuit (12) for transforming an audio signal (11) into an output signal (13). This circuit (12) comprises emphasis means for emphasising in the output signal (13) substantial intensity changes of the audio signal (11). The emphasis means are embodied so as to emphasise the output signal (13) during a period of about 10 ms when an intensity change rate of the audio signal (11) exceeds a predetermined value. In this way, the time domain response of a normal hearing person is approximated.

Abstract: The present invention is a hearing aid for amplifying an acoustic signals comprising: a controller for determining in real time a frequency band at the highest level of the acoustic signals through frequency analysis of the acoustic signals that vary over time, and for generating a control signal to raise a gain for signals of a higher frequency range than the frequency band at the highest level (such as an amplifier Q3, or a band-pass filter group 2 and a diode matrix 3 and a comparator 4, or a digital signal processor 13, or the like); and a first amplifier, in which the control signal from said controller is inputted so that the frequency characteristics are varied, for amplifying the acoustic signals by increasing the gain for signals of the higher frequency range than the frequency band at the highest level (such as an amplifier system consisting of amplifiers Q1 and Q2, or a parametric equalizer 5, or a digital signal processor 13, or the like).

Abstract: A simple hearing enhancement device that takes the normally adequately loud sound levels and optimizes selective frequency gain of the patient's ear passage to improve speech comprehension.

Abstract: A hearing aid includes a sound replay capability. The hearing aid can operate in a normal mode, augmenting sound as the sound occurs, or the hearing aid can operate in a replay mode (typically in response to input from the wearer of the hearing aid), replaying sound beginning up to a specified duration of time prior to the current time.

Abstract: Hearing assistive apparatus can be implemented to include sound replay capability, the hearing assistive apparatus enabling operation in a replay mode (in which the hearing assistive apparatus replays sound that occurred prior to the current time), as well as in a current sound mode (in which the hearing assistive apparatus reproduces—and, typically, enhances—sound as the sound occurs) and/or an off mode (in which the hearing assistive apparatus does not produce sound). The hearing assistive apparatus can also, additionally or alternatively, be implemented so that part of the hearing assistive apparatus is spatially separated from another part of the hearing assistive apparatus (in particular, so that part of the hearing assistive apparatus is spatially separated from part of the hearing assistive apparatus that is ear-mounted).

Abstract: A signal processing circuit and method for increasing speech intelligibility. The invention comprises a receiving circuit for receiving an audio signal detectable by a human. A gain amplifying circuit provides gain amplification of the audio signal. A shaping filter modifies the audio signal to be in phase with a second audio signal present at the receiving circuit and which is detected by the human unprocessed by the signal processing circuit. The shaping filter further differentially amplifies first and second speech formant frequencies to restore a normal loudness relationship between them. A feedback circuit controls the gain amplification in the gain amplifying circuit for enabling the signal processing circuit to substantially prevent regenerative oscillation of the amplified audio signal. Additionally, a signal tone may be injected into the signal processing circuit for automatically controlling the gain amplifying circuit.

Abstract: A system for adjusting audio output including a transmitter unit adapted to be carried by a user and a sound generating system. The transmitter unit includes a memory and a signal transmitter. The sound generating system includes a sound generator, a signal receiver, means for altering a sound signal from the signal generator based upon a signal transmitted by the transmitter to the receiver, and speakers connected to the altering means.

Abstract: An apparatus and a method of voice detection and discrimination apparatus for controlling a voice operated system. A protective ear terminal element (1,2) in the apparatus protects the ear by providing acoustic attenuation. An inner electroacoustic transducer element (M2) on an inner side of the ear terminal element (1,2) detects a first acoustic field and provides a first electronic signal representing the first acoustic field. An outer electroacoustic transducer element (M1) on an outer side of the ear terminal element (1,2) detects a second acoustic field and provides a second electronic signal representing said second acoustic field. An electronics unit (11,E3) is connected with said electroacoustic transducer elements (M1,M2) and comprises first comparison means (27) for comparison of said first and second electronic signals to obtain the difference between said two electronic signals.

Abstract: A hearing prosthesis that automatically adjusts itself to a surrounding listening environment by applying Hidden Markov Models is provided. In one aspect, classification results are utilized to support automatic parameter adjustment of a parameter or parameters of a predetermined signal processing algorithm executed by processing means of the hearing prosthesis. According to another aspect, features vectors extracted from a digital input signal of the hearing prosthesis and processed by the Hidden Markov Models represent substantially level and/or absolute spectrum shape independent signal features of the digital input signal. This level independent property of the extracted features vectors provides robust classification results in real-life acoustic environments.

Abstract: The suppression of interfering noise in an input signal (7) takes place in a main signal path (5), which contains neither a transformation in the frequency range nor a splitting-up into partial band signals, but only a suppression filter (4). The transmission function of the suppression filter (4) is periodically newly determined on the basis of attenuation factors, which are determined in a signal analysis path (6), which lies parallel to the main signal path (5). The attenuation factors are utilized for the attenuation of signal components in frequency bands with a significant proportion of interfering noise. The suppression filter (4) is implemented as transversal filter, the impulse response of which is periodically newly calculated as a weighted sum of the impulse responses of transversal band-pass filters.

Abstract: Apparatus and methods for audio compression and frequency shifting retain the spectral shape of an audio input signal while compressing and shifting its frequency. The fast Fourier transform of the input signal is generated, to allow processing in the frequency domain. The input audio signal is divided into small time segments, and each is subjected to frequency analysis. Frequency processing includes compression and optional frequency shifting. The inverse fast Fourier transform function is performed on the compressed and frequency shifted spectrum, to compose an output audio signal, equal in duration to the original signal. The output signal is then provided to the listener with appropriate amplification to insure audible speech across the usable frequency range.

Abstract: A programmable digital hearing aid circuit and method for operating and programming same are disclosed. The device provides a flexible means to compensate for undesirable frequency response distortion normally due to the electro-acoustical characteristics of the microphone, receiver, and sound coupling mechanisms employed in hearing aid design. Parameters of the programmable hearing aid circuit may also be set to tailor the hearing aid response characteristics for the frequency-dependent hearing loss of an individual hearing aid user. The device is intended to make available a significant improvement in audio fidelity to users of hearing aid devices.

Abstract: A frequency-warped processing system using either sample-by-sample or block processing is provided. Such a system can be used, for example, in a hearing aid to increase the dynamic-range contrast in the speech spectrum, thus improving ease of listening and possibly speech intelligibility. The processing system is comprised of a cascade of all-pass filters that provide the frequency warping. The power spectrum is computed from the warped sequence and then compression gains are computed from the warped power spectrum for the auditory analysis bands. Spectral enhancement gains are also computed in the warped sequence allowing a net compression-plus-enhancement gain function to be produced. The gain versus frequency function is a set of pure real numbers, so the inverse frequency domain transform gives a set of time-domain filter coefficients. The speech segment is convolved with the enhancement filter in the warped time-domain to give the processed output signal.

Abstract: In a binaural hearing aid with signal processors in each unit, each signal processor includes a first processor part for hearing compensation processing of signals representing the sound received at that unit, and a second processor part for hearing compensation processing of signals received from the other unit via a communications link.

Abstract: A method for equalizing output signals from a plurality of signal paths is disclosed. The method comprises steps of identifying a transfer function for each of signal paths, determining a filtering function for each signal path such that a product of the transfer function, and the filtering function is a selected function and applying the filtering function to the corresponding signal path, thereby correcting the transfer function of the signal path to the selected function to equalize the output signals from the signal paths. The step of applying the filtering function comprises steps of providing an equalization filter to the signal path and applying the filtering function to the equalization filter of its corresponding signal path, thereby equalizing output signals from the filter of the signal paths.

Abstract: A self-powered medical device, which is operated independently of the public utility network, has at least one voltage source, a signal input for accepting an analog input signal and a signal processing unit. The sampling rate for the input signal or the clock frequency of at least one digital component can be varied. Therefore, the energy demand can be reduced according to the requirements of the signal processing or when the discharge state of the voltage source requires a reduction.

Abstract: A multichannel hearing aid is provided with at least one frequency channel having a compressor with a compression threshold with an output level below the hearing threshold and an attack time above 0.5 seconds whereby hearing of a sudden sound in a stationary sound environment is facilitated.

Abstract: A system and method for characterizing the contents of an input audio signal and for suppressing noise components of the input audio signal are described. The audio signal is divided into a number of frequency domain input signals. Each frequency domain input signal can be processed separately to determine its intensity change, modulation frequency, and time duration characteristics to characterize the frequency domain input signal as containing a desirable signal or as a type of noise. An index signal is calculated based on a combination of the determined characteristics and signals identified as noise are suppressed in comparison to signals identified as desirable to produce a set of frequency domain output signals with reduced noise. The frequency domain output signals are combined to provide an output audio signal corresponding to the input audio signal but having suppressed noise components and comparatively enhanced desirable signal components.

Abstract: Improved approaches to assist those having hearing loss are disclosed. One approach pertains to providing customization of personal audio systems for a hearing impaired individual. Another approach pertains to designing and producing an audio product that includes components or software that can be customized for hearing impaired individuals. The customization provided by either approach can be performed on-line or off-line.

Abstract: A hearing aid is set forth that includes one or more hearing aid components that introduce an undesired undamped peak into the frequency response of the hearing aid. An electronic damping filter is utilized to compensate for the undamped peak. The electronic damping filter has a notch filter response that includes an inverse peak across the frequency range of the undamped peak thereby electronically damping the frequency response so that the hearing aid output is generally unaffected by the undesired characteristics of the inverse peak. The electronic damping filter may be programmable to vary the magnitude and/or shift the frequency of the inverse peak. A method is set forth that exploits this programmability and allows the same circuit topology to be used in two different hearing aids respectively having two different undamped peaks. A further method allows handling two or more peaks.

Abstract: A signal processing circuit and method for increasing speech intelligibility. The invention comprises a receiving circuit for receiving an audio signal detectable by a human. A gain amplifying circuit provides gain amplification of the audio signal. A shaping filter modifies the audio signal to be in phase with a second audio signal present at the receiving circuit and which is detected by the human unprocessed by the signal processing circuit. The shaping filter further differentially amplifies first and second speech formant frequencies to restore a normal loudness relationship between them. A feedback circuit controls the gain amplification in the gain amplifying circuit for enabling the signal processing circuit to substantially prevent regenerative oscillation of the amplified audio signal. Additionally, a signal tone may be injected into the signal processing circuit for automatically controlling the gain amplifying circuit.

Abstract: A signal processing circuit and method for increasing speech intelligibility. The invention comprises a receiving circuit for receiving an audio signal detectable by a human. A gain amplifying circuit provides gain amplification of the audio signal. A shaping filter modifies the audio signal to be in phase with a second audio signal present at the receiving circuit and which is detected by the human unprocessed by the signal processing circuit. The shaping filter further differentially amplifies first and second speech formant frequencies to restore a normal loudness relationship between them. A feedback circuit controls the gain amplification in the gain amplifying circuit for enabling the signal processing circuit to substantially prevent regenerative oscillation of the amplified audio signal. Additionally, a signal tone may be injected into the signal processing circuit for automatically controlling the gain amplifying circuit.

Abstract: Time-limited electrical audio signals are fed to an electromechanical output transducer in addition to the signals from the hearing aid input. Some of the time-limited audio signals are user-defined. The process is implemented in a hearing aid having an electromechanical transducer and a signal processor. An audio signal generator has a user-changeable memory and/or a read/write memory that can be programmed by the user.

Abstract: There is provided a method of controlling feedback in an acoustic system, for example a digital hearing aid, in which there is a potential feedback path between the output and the input. The method comprises making a spectral estimate of the input signal spectrum, and then subjecting the spectral estimate to a psycho-acoustic model to generate a control signal. A noise source is passed through a shaping filter, which is controlled with the control signal, to generate frequency-shaped noise, which is inaudible to someone hearing the output. The frequency-shaped noise is then added to the input signal to form a combined signal, which is processed in a forward path, to generate a first output signal. The first output signal and the frequency-shaped noise signal are analyzed, to determine the presence of feedback at difference frequencies, and the characteristics of the forward path are modified to reduce the gain thereof at frequencies where feedback is detected.

Abstract: An apparatus and method for screening an individual's ability to process acoustic events is provided. The invention provides sequences (or trials) of acoustically processed target and distractor phonemes to a subject for identification. The acoustic processing includes amplitude emphasis of selected frequency envelopes, stretching (in the time domain) of selected portions of phonemes, and phase adjustment of selection portions of phonemes relative to a base frequency. After a number of trials, the method of the present invention develops a profile for an individual that indicates whether the individual's ability to process acoustic events is within a normal range, and if not, what processing can provide the individual with optimal hearing. The individual's profile can then be used by a listening or processing device to particularly emphasize, stretch, or otherwise manipulate an audio stream to provide the individual with an optimal chance of distinguishing between similar acoustic events.

Abstract: A digital hearing aid applies clipping to the processed digital signal after at least part of the interpolation of the signal has occurred. The clipping may be incorporated into the output demodulation stage of the hearing aid. The final stages of interpolation may also be incorporated into the demodulation stage.

Abstract: A hearing aid with different assemblies for picking up, further processing and adjusting an acoustic signal to the hearing ability of a hearing impaired person, wherein a digital signal processing is performed in the hearing aid, has a buffer storage unit which shifts or transposes a defined frequency region into another frequency region which is better detectable by the wearer of the hearing aid. The buffer storage unit has individual storage segments for receiving and storing digital data representing acoustic content of an incoming audio signal, and the read-in and read-out frequency of the buffer storage unit differs by a defined factor.

Abstract: Apparatus and method for shifting the frequency range of an audio frequency range signal using digital frequency shifting and single sideband amplitude modulation techniques. In one embodiment, a frequency shifted single sideband, amplitude modulated signal is formed for application to the human hearing sensory system to provide enhanced hearing for hearing impaired persons. In another embodiment, the audio signal is shifted to an ultrasonic frequency range utilizing digital signal processing techniques in which an audio frequency signal is frequency shifted by modulation to a carrier frequency and in which a single sideband modulated signal is formed. The digital single sideband amplitude modulation techniques of this embodiment are also applicable to digital modulation systems for uses other than in hearing aids. In another embodiment, analog single sideband modulation is utilized for frequency shifting in a hearing aid application.

Abstract: A hearing aid apparatus and method of the type in which audio frequency signals are frequency upshifted to ultrasonic frequency bands and are applied as vibrations to the human body to generate a hearing response. Frequency upshifting of the audio frequency signals to the ultrasonic frequency band is attained in one embodiment by amplitude modulation of an ultrasonic frequency carrier signal with an audio frequency modulating signal to form a modulated signal in which one of the two sidebands is either completely or substantially suppressed and a modulated signal having only one predominant sideband is thus derived for application to the human sensory system to generate a hearing response.

Abstract: A hearing aid has a microphone, a signal processing chain having a series of signal processing levels, a signal output transducer, a memory in which at least one set of parameters allocated to a hearing situation is stored that enables matching of the parameters of the signal processing stages to the hearing impairment of the wearer. Adjustment elements independent of the signal processing stage are provided, and the allocation of the adjustment elements 9 to the individual signal processing stages can be selectively determined. Adjustment of the respective signal processing stages can be accomplished in a non-computerized manner using the adjustment elements, thereby allowing the hearing aid to be used in countries and regions wherein personal computers are not readily available or accessible.

Abstract: A method is provided for programming a digital hearing aid using a program encoded in an audio band (20 Hz-20 kHz) signal, to transmit and verify programs and algorithm parameters. Preferably, this is in a digital hearing aid including filterbanks, filtering the audio signal into different frequency bands. The signal is encoded by the presence and absence of a signal in each frequency band or by other well-known modulation techniques used by computer modems. Special programming signals are provided alternating between the frequency bands in a manner to clearly distinguish the program data from any other interfering or normally present audio signal. The method does not require additional hardware, and offers reduced power consumption, as compared to some known wireless programming interfaces. It enables remote programming over a network using standard multimedia computer hardware.

Abstract: For limiting the signal transmitted to the human ear in dependency on an incoming acoustical signal, there is provided a signal processor, the output of which acts on an output transducer and on a calculator unit which calculates according to a preselected model the psychoacoustical entity loudness of the incoming acoustic signal. The loudness, thus calculated, is compared with a predetermined loudness level (MAL) and according to the result of such comparison, parameters at the processor unit are varied so as to restrict the transmitted loudness on the MAL level.

Abstract: A digital signal processing hearing aid is disclosed having a plurality of digital signal processing means for processing input digital signals, and a selector switch manipulable by a user for choosing which of the processing means to utilize. Each of the digital signal processing means is designed to provide optimal results in a particular listening environment. Since the user is allowed to choose which of the plurality of processing means to invoke, and since each processing means is specifically designed to operate in a particular listening environment, the hearing aid is capable of providing excellent results in a plurality of listening environments.

Abstract: A hearing compensation system for the hearing impaired comprises an input transducer for converting acoustical information at an input to electrical signals at an output, an output transducer for converting electrical signals at an input to acoustical information at an output, a plurality of bandpass filters, each bandpass filter having an input connected to the output of said input transducer, a plurality of AGC circuits, each individual AGC circuit associated with a different one of the bandpass filters and having an input connected to the output of its associated bandpass filter and an output connected to the input of the output transducer. The bandpass filters and AGC circuits may be divided into two processing channels, one for low frequencies and one for high frequencies and may drive separate audio transducers, one configured for maximum efficiency at low frequencies and one configured for maximum efficiency at high frequencies.

Abstract: A hearing aid is set forth that includes one or more hearing aid components that introduce an undesired undamped peak into the frequency response of the hearing aid. An electronic damping filter is utilized to compensate for the undamped peak. The electronic damping filter has a notch filter response that includes an inverse peak across the frequency range of the undamped peak thereby electronically damping the frequency response so that the hearing aid output is generally unaffected by the undesired characteristics of the inverse peak.The electronic damping filter may be programmable to vary the magnitude and/or shift the frequency of the inverse peak. A method is set forth that exploits this programmability and allows the same circuit topology to be used in two different hearing aids respectively having two different undamped peaks. A further method allows handling two or more peaks.