Abstract: An audio amplification device such as a hearing aid equipped with a user-operable volume control that provides users with the facility to return the volume of the device to a specific level after having been adjusted to a different level by the user. The device includes an electronic module adapted to detect when the volume control reaches a pre-determined volume reserve position from a second volume position, and to emit a reference signal when the volume reserve position is reached. The reference signal may be an audible tone, or another type of distinct signal. The present invention may be adapted for use with binaural hearing aid fittings.

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: To improve the speech comprehensibility given treatment with a hearing aid device, during the operation of the hearing aid device speech signal levels and noise signal levels are determined in a plurality of frequency bands of an input signal. An automatic adjustment of the amplification follows, dependent on the determined signal level and the signal frequency. The determination of amplification parameters thereby ensues under inclusion of a loudness model and a speech comprehensibility model.

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: Apparatus and methods for enabling and disabling an amplifier is described. Amplifiers can be enabled and disabled for the sake of power and energy savings and reduction of dissipated heat. The circuit is based on a gain stage and threshold circuitry. A signal associated with an amplifier is introduced into the gain stage. The gain stage amplifies the signal level and provides the amplified signal level to threshold circuitry. The threshold circuitry references the amplified signal level and determines signal states. The control circuitry interprets these signal states to determine when to enable or disable an amplifier. The output of the threshold circuitry can be used to adjust the amplified signal level to oppose drift in the gain stage.

Abstract: A hearing assist device (10) for a person (P). The device comprises a speaker device (SP1) for presenting sound to an ear canal of the person and circuitry for identifying a specified area relative to the person. The device further comprises a first microphone (M1) for providing a first sound signal in response to a first sound source located inside the area and in response to a second sound source located outside the area. Further, the device comprises a second microphone (M2) for providing a second sound signal in response to the first sound source and the second sound source. Still further, the device comprises circuitry (16) for determining a position of the first sound source and the second sound source in response to the specified area, the first sound signal and the second signal. Finally, the device comprises circuitry (16) for outputting a processed signal in response to the position.

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: 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: In a hearing aid and a method for processing microphone signals in a hearing aid, a signal processing unit is provided in order to amplify and/or attenuate signal parts of at least two microphone signals in a directionally dependent manner. The hearing aid has a signal analysis unit that is capable of modifying at least one property of the direction-dependent amplification and/or attenuation thereby achieving high transmission quality and noise suppression in a multitude of auditory situations.

Abstract: The invention relates to a hearing aid, preferably a programmable hearing aid, having at least one microphone (1), at least one signal processor with at least one channel, an output amplifier (9) and an output transducer (10), at least one of the channels containing a signal processing circuit (4) with at least one percentile estimator (6) for the continuous determination or calculation of at least one percentile value of the input signal from a continuous analysis and evaluation of the frequency and/or amplitude distribution of the input signal, whereby the percentile value(s) serve either directly or indirectly as control signals for controlling the gain and/or the frequency response of the electronic processing circuit, the percentile estimator (6) consisting essentially of a comparator stage (12) with two inputs and two outputs, the first input being directly or indirectly connected to the input of the hearing aid, its two outputs controlling a first control stage (13) the output signals of which control

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: 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: In a hearing aid signal processor with AGC in at least three processing channels (a, B, C) for different frequency bands and with noise squelching capability to affect the gain control in a t least a lowest frequency band (A) and one intermediate frequency band (B) speech signals components in the intermediate frequency band of an input signal including background noise are intensified by estimation of the content of speech signal components in at least the highest frequency band (C) and modification of the gain adjustment cause by noise squelching in the intermediate frequency band (B) to reduce the noise spending.

Abstract: An improved hearing aid is disclosed having a switched release automatic gain control. In one embodiment, an input signal is presented to a comparator with a reference typically around, for example, 65 dB SPL instantaneous. The output of the comparator controls a switch in a release circuit of an automatic gain control. While the instantaneous input level exceeds the threshold, the release circuit is enabled and the automatic gain control behaves normally. While the instantaneous input level does not exceed the threshold, the release circuit is disabled and the automatic gain control maintains its current gain setting, essentially indefinitely. Because speech, even soft speech, has a very high positive peak content, this circuit will recover any needed gain in the presence of speech (assuming an appropriate threshold) but not recover gain to background noise whose positive peak content is below the threshold.

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: A method and apparatus for matching or balancing average signal levels between at least two input signal channels and their respective microphone elements so as to allow a hearing aid or instrument to maintain optimum directional characteristics over time are provided. The hearing aid may comprise an analogue signal processor or a digital signal processor adapted for controlling characteristics, e.g., gain and/or frequency response, of one or more of the input signal channels.

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: 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: The present invention provides a sound processing apparatus capable of outputting clearly perceivable sound. The sound processing apparatus comprises high frequency range enhancement unit for a sound signal amplified by an amplifier, gain-varying unit for varying the gain of a sound signal whose high frequency component is enhanced on the basis of a control signal presented from amplitude sensing unit, addition unit for adding a sound signal that is output from the gain-varying unit and a sound signal from the amplifier, frequency characteristic adding unit whereby the amplitude of the sound signal that is input from the amplifier is increased with increasing frequency in the specified frequency range, and amplitude sensing unit for sensing the magnitude of the amplitude of the sound signal from frequency characteristic adding unit and generating a voltage signal as a control signal substantially proportional to the amplitude thereof.

Abstract: For reducing the power consumption of a hearing aid system, an internal or external hearing aid signal source with a signal line for the transmission of a signal to a hearing aid amplifier and a control device for the connection and disconnection of the hearing aid signal source is provided in the hearing aid. A monitoring logic monitors the signal line and supplies a switch signal to the control device, so that the hearing aid signal source can be connected and disconnected on the basis of the switch signal. Each signal source can thus be individually connected and disconnected without employing additional connections from the hearing aid system to the signal source.

Abstract: The method for configuring the functional properties of an audiological device in the form of a hearing aid initially provides a hearing aid with an IC that can be differently configured in view of its properties, permitting configuration upgrade information to be employed that is either distributed to middlemen via a separate data carrier or transmitted on-line from a data store of the manufacturer to a programming station of the middleman. The middleman has the possibility of himself upgrading hearing aids initially present as basic hearing aids in customized fashion, the configuration information being used for this purpose and the hearing manufacturer being paid for this.

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: Automatic gain control in a hearing aid is effected by detecting an input sound level and/or an output sound level and adapting the output sound level supplied by the hearing aid in response to the detected sound level by controlling the gain of the hearing aid towards an actual desired value of the output sound level. The gain control is effected at increases and decreases, respectively, of the input sound level by adjusting the gain towards the actual desired value with an attack time and a release time, respectively, which are adjusted in response to the detected sound level to a relatively short duration providing fast gain adjustment at high input and/or output sound levels and to a relatively long duration providing slow gain adjustment at low input and/or output sound levels.

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: Hearing aid for improving the hearing ability of the hard of hearing, comprising an array of microphones, the electrical output signals of which are fed to at least one transmission path belonging to an ear. Means are provided for deriving two array output signals from the output signals of the microphones, the array having two main sensitivity directions running at an angle with respect to one another and each of which is associated to an array output signal. Each array output signal is fed to its own transmission path belonging to one ear of a person who is hard of hearing.

Abstract: A hearing improvement device using a multi-coil coupling system and methods for operating such a device are disclosed. An embodiment of the present invention may use an array microphone to provide highly directional reception. The received audio signal may be filtered, amplified, and converted into a magnetic field for coupling to the telecoil in a conventional hearing aid. Multiple transmit inductors may be used to effectively couple to both in-the-ear and behind-the-ear type hearing aids, and an additional embodiment is disclosed which may be used with an earphone, for users not requiring a hearing aid.

Abstract: A hearing aid for converting input acoustical energy into an electrical signal, processing the signal and converting the processed electrical signal into acoustical energy is disclosed. The hearing aid includes a digitally programmable gain amplifier. The digitally programmable gain amplifier comprises circuitry for decoding a digital control input signal (referred to as the serial data input), and based upon the digital control signal it comprises a means for changing the amplifier gain.

Abstract: A method for fitting a hearing compensation device comprises selecting a plurality of loudness levels for a plurality of frequencies and comparing each loudness level for each frequency for perceived sameness. The loudness levels may then be adjusted as needed to achieve perceived sameness across the frequency spectrum. A gain curve for each frequency is calculated from the selected plurality of loudness levels.

Abstract: An improved analog-to-digital converter device suitable for use in digital hearing aids, and methods for operating such a device are disclosed. Aspects of the present invention may provide a full 16-18 bits (96-108 dB) of dynamic range in a digital hearing aid circuit using 14-bit analog-to-digital converters typically available for hearing aid use. An embodiment of the present invention may use two analog-to-digital converters that are clocked by the same sampling trigger. One of the converters may be preceded by an amplifier having, for example, 24 dB of gain, while the other converter may be preceded by an amplifier having 0 dB of gain. The output of the first converter or the output of the second converter may be used, or a combination of the two outputs may be used to produce audio for the hearing aid user, depending upon input signal level.

Abstract: An auditory prosthesis, and method, which is able to adapt better to filter out a selected unwanted portion of the auditory input signal by relying on a human activation, such as activation by the user, who knows by listening when the auditory environment contains only, or mostly only, the selected unwanted portion of the auditory input signal. This person may then activate the adaptive filter of the auditory prosthesis. The adaptive filter then utilizes the then current auditory environment as a noise reference on which to adapt. A transducer is adapted to receive the environmental sound and convert the environmental sound into an electrical input signal. The electrical input signal contains a selected electrical component corresponding to the selected auditory component in the environmental sound. An adaptive filter receives the electrical input signal and provides a filtered signal. The adaptive filter has adaptable filtering characteristics based upon a reference.

Abstract: In a method and apparatus for producing constant sound pressure level in a hearing aid, a microphone, a receiver and a signal processing unit are provided. A switching unit produces constant sound pressure levels. The switching unit has an envelope generator for producing an envelope value of a signal value to be processed and a divider for calculating the ratio of the signal value and the envelope value as the signal to be output or further processed.

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 digital quasi-RMS detector is provided that approximates the time-varying RMS energy of a signal. The digital quasi-RMS detector rectifies the signal and compares the rectified signal with an estimated present energy value of the audio signal. If the difference between the rectified signal and the estimated present energy value is not greater than zero, then the digital quasi-RMS detector multiplies the rectified signal by a first time constant to generate a first filtered signal and sums the first filtered signal with the estimated present energy value to determine the approximate RMS energy. If the difference between the rectified signal and the estimated present energy is greater than zero, however, then the digital quasi-RMS detector multiplies the rectified signal by a second time constant to generate a second filtered signal and sums the second filtered signal with the estimated present energy value to determine the approximate RMS energy.

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 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: The present invention is a digital dynamic compression or automatic gain control (AGC) (10) adapted for use in high quality audio and hearing aids applications. An efficient digital AGC design employs two compact ROM-based tables (ROM_CSD, ROM_SPL) in addition to two comparators (COMP_A, COMP_B) and several registers (REG_A, REG_B, ADDR_A, ADDR_B). While one ROM stores the values of discrete input signal levels, the other contains gain codes based on a canonical signed digit (CSD) coding approach that leads to a very simple gain multiplier (20). In many cases an extremely compact table for gain values can be achieved by reusing a single small-size ROM that behaves like one that is several time larger. Two design examples are shown to expound the insights of the new digital AGC design. For the less-than-half-dB-gain-step cases only two adders are required for the multiplier whereas just three adders are needed in the situations with less than quarter-dB gain steps.

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 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: Systems, devices, and methods are provided to inhibit apparent amplitude modulation in non-linear processing that causes distortion in a processed signal. One aspect of the invention includes a hearing aid. The hearing aid includes a microphone to receive an input signal, a speaker to reproduce the input signal, and a processor. The processor processes the input signal using a gain. The processor includes an inhibitor, which inhibits distortions, and an adjuster, which adjusts the gain. The inhibitor acts to smooth an envelope of the input signal to inhibit undesired modulation. The adjuster adjusts the gain if the envelope is either above or below a threshold. The hearing aid further includes a compression recapture system to supply the compressed portion of the input signal to more closely reproduce the actual input signal.

Abstract: Systems, devices, and methods are provided to inhibit apparent amplitude modulation in non-linear processing that causes distortion in a processed signal. One aspect of the invention includes a hearing aid. The hearing aid includes a microphone to receive an input signal, a speaker to reproduce the input signal, and a processor. The processor processes the input signal using a gain. The processor includes an inhibitor, which inhibits distortions, and an adjuster, which adjusts the gain. The inhibitor acts to smooth an envelope of the input signal to inhibit undesired modulation. The adjuster adjusts the gain if the envelope is either above or below a threshold.

Abstract: In a hearing amplification device adapted to receive a sound signal, the hearing amplification device having at least one channel configured to receive an input representative of said sound signal, the improvement comprising the channel being further configured to provide (1) linear gain for an input representative of a portion of the sound signal having a sound level less than a compression threshold, (2) rapid compressive gain for an input representative of a portion of the sound signal having a sound level greater than the compression threshold, wherein the rapid compressive gain is less than the linear gain, and (3) adaptive control of the compression threshold. Preferably the rapid compressive gain is instantaneous. Adaptive compression threshold control may be achieved in response to a user input and/or to sound signal changes. By adaptively controlling the compression threshold, performance of the device can be optimized to match its environment.

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: An open ear canal hearing aid system comprises an ear canal tube sized for positioning in an ear canal of a user so that the ear canal is at least partially open for directly receiving ambient sounds. The open ear canal hearing aid system further comprises a sound processor for amplifying received ambient sounds included within a predetermined frequency to produce processed sounds and for supplying said processed sounds to said ear canal tube.

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: An apparatus for a hearing aid provides an application specific integrated circuit (ASIC) for filtering of input signals and a programmable digital signal processor connected to it, for control of filterbank gains. This provides a compromise between the conflicting goals of size, re-programmability and power consumption. The fixed portion of the processing, i.e. filtering is implemented in hardware in the ASIC, and the variable or adjustable portion of the processing is implemented by the programmable digital signal processor. The filterbank has an adjustable number of channels and means for shifting the center frequencies of the bands in unison to one of two discrete sets of center frequencies. A wide range of hearing loss compensation schemes can be implemented. Software programs can be executed on the programmable digital signal processor.

Abstract: A method for fitting a hearing compensation device comprises selecting a plurality of loudness levels for a plurality of frequencies and comparing each loudness level for each frequency for perceived sameness. The loudness levels may then be adjusted as needed to achieve perceived sameness across the frequency spectrum. A gain curve for each frequency is calculated from the selected plurality of loudness levels.

Abstract: In a method and circuit for the amplification of input signals of a hearing aid, a compression of the signals picked up by the hearing aid ensues in a AGC circuit dependent on the acquirable signal level. For assuring a dynamics compression, the method and circuit implement a signal analysis for the recognition of the acoustic situation in addition to the acquisition of the signal level of the input signal, and the behavior of the dynamics compression is adaptively varied on the basis of the result of the signal analysis. A fixed input-output characteristic that is usually defined on the basis of stationary signals forms the basis of parameter settings of a dynamics compression has been used in conventional calculations. The resulting time behavior for the compensation of input signals by a hearing aid exhibits an effectively different compression given time-variable, modulated signals, particularly given speech. The result is that the compression parameters are not optimally set for all signals.

Abstract: In a hearing aid having feedback cancellation, a voltage regulator is provided to regulate the voltage from the hearing aid battery to the class D output. Feedback cancellation processing dynamically models the entire acoustic feedback path from the hearing aid processing to microphone, including the gain of the output stage. The gain of the output stage varies with the voltage of the battery, and the battery voltage depresses when a loud signal is processed by hearing aid, so the voltage regulator prevents the output stage gain from varying with time.

Abstract: A "true" hearing aid transfer function, including feedback, is derived from measurements taken with the hearing aid fitted in a patient's ear canal. Closed loop transfer functions are calculated at several hearing aid gains without opening the internal circuitry of the hearing aid using a time domain Weiner optimal filter model. The combined open loop transfer function of the hearing aid and feedback path is then calculated. Once the open loop transfer function is known, potentially unstable frequencies are identified and maximum hearing aid gain settings are determined. The hearing aid transfer function and transfer function of feedback path are also calculated from the closed loop transfer function measurements.

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.