Abstract: Disclosed is an IC chip type hearing aid module for use in a mobile communication terminal having a hearing aid circuit including: a first section having a resistor R1 and an inductor L1 which are connected in series on a first output line Receiver_P between the mobile communication terminal and an ear speaker; a second section having a resistor R4 and an inductor L2 which are connected in series on a second output line Receiver_N between the mobile communication terminal and the ear speaker; a third section having a resistor R2 and a capacitor C2 which are connected in series between an input terminal of the first section and an output terminal of the second section; and a fourth section having a resistor R3 and a capacitor C3 which are connected in series between an input terminal of the second section and an output terminal of the first section.

Abstract: In order to switch between different hearing programs to adjust to a momentary acoustic scene, a method for adjusting a hearing device, in which one of several possible hearing programs can be selected in order to adjust to a momentary acoustic scene, the method comprising the steps of detecting a desired hearing program change, changing parameters (b1, . . . , bm) of a transfer function provided between a microphone (M1) and a receiver of the hearing device in order to adapt it to the detected hearing program change, adjusting the parameters (b1, . . . , bm) to be changed from a momentary value to a desired value in such a manner that a smooth transition is perceived by the hearing device user while changing from a momentary hearing program to the desired hearing program, whereas each of the smooth transition is individually adjustable.

Abstract: A method for modification of a cepstro-temporally smoothed gain function of a gain function resulting in a bias compensated spectral gain function is provided. The cepstro-temporal smoothing increases the quality of an enhanced output signal, as it affects only spectral outliers caused by estimation errors, while the speech characteristics are well preserved. However, due to the cepstral transform, the temporal smoothing is done in the logarithmic domain rather than the linear domain, and hence results in a certain bias. Thus, the method for a general bias compensation for a cepstro-temporal smoothing of spectral filter gain functions that is only dependent on the lower limit of the spectral filter-gain function.

Abstract: The object of the claimed hearing aid device is to improve the speech intelligibility of a speech signal transmitted by the hearing aid device. To this end provision is made to define a maximum gain of the input signal and to determine a target gain at least at a first and second frequency of an input signal. A resulting gain is set in the hearing aid device, which does not exceed the maximum gain. A reduction of the resulting gain compared with the target gain at the first frequency is compensated for according to the invention by an automatic increase in the set resulting gain compared with the target gain at the second frequency, with compensation preferably being achieved by improving the speech intelligibility of a speech signal transmitted with the aid of the hearing aid device.

Abstract: An audio signal received at a mobile device is processed using a first set of equalization parameters in an acoustic coupling mode of operation, and is processed using a second set of equalization parameters in an inductive coupling mode of operation. The processed audio signal is transmitted to a speaker associated with the mobile device for acoustic coupling to a human ear and/or a hearing aid microphone, or for inductive coupling to a hearing aid telecoil.

Abstract: A hearing aid (22) having a microphone (1), a processor (53) and an output transducer (12), is adapted for obtaining an estimate of a sound environment, determining an estimate of the speech intelligibility according to the sound environment estimate, and for adapting the transfer function of the hearing aid processor in order to enhance the speech intelligibility estimate. The method according to the invention achieves an adaptation of the processor transfer function suitable for optimizing the speech intelligibility in a particular sound environment. Means for obtaining the sound environment estimate and for determining the speech intelligibility estimate may be incorporated in the hearing aid processor, or they may be wholly or partially implemented in an external processing means (56), adapted for communicating data to the hearing aid processor via an appropriate link.

Abstract: Hearing aid wearers can find themselves in situations in which the necessary replacement and/or recharging of a discharged voltage source is not immediately possible. In such a situation, reducing the low-frequency signal elements in the acoustic output signal allows a type of emergency mode to be maintained for a certain period. The majority of important acoustic information thus remains comprehensible to the hearing aid wearer.

Abstract: The aim of the invention is to control the bass reproduction of audio signals in electroacoustic transducers based on the psychoacoustic principle denoted by the term “virtual pitch” or “residual hearing (hearing of missing fundamental)”, in such a way that the perception of the virtual bass reproduction of the audio signals is improved in relation to prior art. To this end, the reproduction of the low pitched frequencies or basses released in the electroacoustic transducer is controlled by the amplification of the harmonic waves already contained in the audio signals, in the form of a simulation, in such a way that the listener experiences or perceives an improved bass reproduction.

Abstract: A sound processing process is disclosed, with particular application to auditory prostheses. After input sound signals are processed into channels, an algorithm is applied to selectively increase the modulation depth of the envelope signals.

Abstract: The invention relates to a hearing aid system comprising an input transducer for converting an input sound signal comprising an information signal part of a known waveform and a background noise part to an electrical analogue input signal, optionally an A/D converter for converting the electrical input signal to a digital input signal. The invention further relates to a method of making a critical gain measurement. The object of the present invention is to improve the signal-to-noise ratio of a signal to be measured or detected in a hearing instrument compared to prior art solutions. The problem is solved in that a matched filter receiving said analogue or digital input signal and optimized to improve the identification of the information signal part from the noisy input signal. An advantage of the invention is that it provides an alternative scheme for improving signal to noise ratio of a hearing aid. The invention may e.g.

Abstract: According to the invention, in a hearing instrument it is once or repeatedly checked whether a second hearing instrument is present and active. If a second hearing instrument is active on the contralateral side, a first gain (corresponding to the gain for binaural fitting) is applied. If, however, no further hearing instrument is found to be active, a second—increased—gain is applied. The second gain may simply correspond to the first gain increased by a certain dB value. It may as an alternative be a specifically adapted gain characteristic of a monaural fitting for the user.

Abstract: Provided is a hearing device and method for compensating for a hearing loss. To compensate for the hearing loss, the hearing device and method utilize the gain during feedback in the forward path of a compressive system which, after having reached its steady state in “closed loop” operation, is equal to the feedback threshold gain. The steady state is reached soon after having applied a low input signal level to the hearing device, which input signal level is below 55 dB SPL (Sound Pressure Level), for example, and would result, for the open loop compressive system, in a larger gain than the feedback threshold gain of the closed loop system, respectively, would result in the maximum possible hearing device gain if maximum possible hearing device gain is below feedback threshold gain. The signal feedback gain is assessed in this steady state.

Abstract: It is intended that the switching on and off of a hearing aid (I) should be made more user-friendly. To this end the hearing aid (I) is fitted with a temperature sensor (TS) to detect the body heat of the hearing aid wearer and therefore that the hearing aid is inserted in the auditory canal. The hearing aid is then switched on and off again based on the temperature signal. As an alternative to the temperature sensor, it is possible to use a pressure sensor to detect contact pressure of the hearing aid housing on the auditory canal, a resistance sensor to detect an electrical load resistance as a function of volume or an acoustic sensor to detect a sound level. It is also possible to switch the hearing aid (I) on and off wirelessly using a remote control.

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: [Problems] To easily associate the parameters representing the acoustic characteristic of a hearing aid with the audibility of the hearing aid user, shorten the time for adjusting the hearing aid, and improve the accuracy of the adjustment of the parameter. [Means for Solving Problems] By applying a two-dimensional matrix for changing the acoustic parameters of the hearing aid, a person adjusting the hearing aid can easily change the acoustic parameters depending on the audibility of the hearing aid user, the accuracy of the adjustment of the acoustic parameters is improved, and the period of time for adjusting the hearing aid is shortened.

Abstract: A hybrid serial/parallel bus interface has a data block demultiplexing device. The data block demultiplexing device has an input configured to receive a data block and demultiplexes the data block into a plurality of nibbles. For each nibble, a parallel to serial converter converts the nibble into serial data. A line transfers each nibble's serial data. A serial to parallel converter converts each nibble's serial data to recover that nibble. A data block reconstruction device combines the recovered nibbles into the data block. The data block is employed by a gain controller.

Abstract: A hearing aid (200) comprises at least one microphone (210), a signal processing means (220) and an output transducer (230). The signal processing means is adapted to receive an input signal from the microphone. The signal processing means is adapted to apply a hearing aid gain to the input signal to produce an output signal to be output by the output transducer, and the signal processing means comprises means for adjusting the hearing aid gain by a direct transmission gain calculated for the hearing aid. The invention further provides a method and a system for reducing noise, as well as a computer program product.

Abstract: A hearing aid comprises at least one microphone, a signal processing means and an output transducer, the signal processing means being adaptive to receive an input signal from the microphone, wherein the signal processing means is adapted to apply a hearing aid gain to the input signal to produce an output signal to be output by the output transducer, and wherein the signal processing means further comprises means for adjusting the hearing aid gain if the hearing aid gain would be below a direct transmission gain calculated for the hearing aid.

Abstract: In a method and hearing aid (200) for processing sound signals for hearing impaired persons by providing multi-band compression processing an input sound signal is filtered by a band split filter (202) into a number of frequency bands to obtain band split signals. A signal level for each of the band split signals is determined and the frequency bands are arranged into a number of groups. Based on the signal levels in each of the groups, a compressor input level for a number of band split compressors each associated to one of the frequency bands is calculated. A compressor gain for each band split compressor is determined based on the corresponding compressor input signal and the band split signals are amplified with the corresponding compressor gain and summed in a summing unit (208) to produce an output sound signal.

Abstract: In one embodiment, an apparatus for processing sound includes a means (401) for analysing a sound signal into a number of frequency bands and a means (403) for applying variable gain to each frequency band independently. Gain in applied under control of a number of gain comparator means (409) each of which generates a number ot statistical distribution estimates in respect of each signal and compares those estimates to predetermined hearing presponse 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 system for implementing an acoustical front end customization are disclosed. The customization is implemented to optimize the sound level for each individual cochlear implant user. A known audio signal is generated using a sound source and captured by a microphone system. The captured sound signal is sampled at one or more locations along the signal processing pathway, and a spectrum is determined for the sampled signal and the known signal. A ratio of the two spectrums is related to the undesired transformation of the sampled signal, and a digital filter is designed based on the ratio to filter out the undesired transformation.

Abstract: A method of amplification includes amplifying an input signal with a predetermined or user adjustable maximum gain to produce a second signal and compressing the dynamic range of the second signal with an automatic gain control to produce an output signal. The compression ratio of the automatic gain control is greater than one for output signals below a predetermined threshold output signal level and is essentially one for output signals above the predetermined threshold output signal level. The compression ratio may be predetermined or the method may include determining the compression ratio of the automatic gain control. The compression ratio may be determined according to the amplitude of the output signal or according to an input-output curve. The input-output curve my be predetermined and may be selected according to the auditory needs of a listener. Preferably the input-output curve is determined according to the predetermined or user adjustable maximum gain.

Abstract: A hearing prosthesis that automatically adjusts itself to a surrounding listening environment is provided. In one aspect, the automatic adjustment is achieved by controlling one or several algorithm parameters of a predetermined signal processing algorithm. In another aspect, the signal input to the hearing prosthesis is continuously and automatically classified as belonging to one of several everyday listening environments, the results of the classification being communicated to the processing means thus allowing the processing means to control the algorithm parameters.

Abstract: The invention provides a method for dynamic determination of time constants to be used in a detection of the signal level of an input signal of unknown level in an electric circuit, comprising the following steps:—feed the input signal through an auxiliary level detection means that is reacting faster to changes in the input sound signal level than the detection of the signal level as a whole,—feed either the input signal or the output of the auxiliary level detection means through a guided level detection means, which is arranged with a guided time constant, and where the guided level detection means outputs an estimate of the level of the input signal,—analyze the outputs of the auxiliary and the guided level detector means, determine the time constant of the guided level detection means based on this analysis.

Abstract: A method of processing an acoustic input signal into an output signal in a hearing instrument includes converting the acoustic input signal into a converted input signal, and applying a gain to the converted input signal to obtain the output signal. According to the invention, the gain is calculated using a room impulse attenuation value being a measure of a maximum negative slope of the a converted input signal power on a logarithmic scale. The calculation of the gain may include evaluating a signal power development value being a measure of the actual converted input signal power attenuation or signal power increase, evaluating a signal-to-reverberation-noise ratio from the signal power development value and the room impulse attenuation value, and calculating, based on a gain rule, said gain from said signal-to-reverberation-noise ratio.

Abstract: It should be possible to quickly and effectively adjust a hearing aid even in the high-frequency range above 8 kHz. For this purpose it is provided that an open-loop gain measurement is carried out in the upper frequency range and a maximum amplification or a frequency-dependent maximum amplification curve is fixed. This maximum amplification in the high-frequency range should not be exceeded. The hearing aid wearer can then optionally select one of a plurality of amplification curves located there below. In the low-frequency range a conventional amplification adjustment is carried out for example by a prescriptive audiogram-based formula. A hybrid adjustment procedure that is easy to carry out is thereby provided for the entire frequency range.

Abstract: Various methods and apparatus for processing audio signals are disclosed herein. The assembly may be attached, adhered, or otherwise embedded into or upon a removable oral appliance to form a hearing aid assembly. Such an oral appliance may be a custom-made device which can enhance and/or optimize received audio signals for vibrational conduction to the user. Received audio signals may be processed to cancel acoustic echo such that undesired sounds received by one or more intra-buccal and/or extra-buccal microphones are eliminated or mitigated. Additionally, a multiband actuation system may be used where two or more transducers each deliver sounds within certain frequencies. Also, the assembly may also utilize the sensation of directionality via the conducted vibrations to emulate directional perception of audio signals received by the user. Another feature may include the ability to vibrationally conduct ancillary audio signals to the user along with primary audio signals.

Abstract: The present invention relates to a hearing aid with a compressor having a low and gain independent delay and low power consumption. The hearing aid comprises a multi-channel compressor for compensation of dynamic range loss and with a digital input for inputting a digital sound signal, and an output connected to an amplifier with a selectable gain as a function of frequency for compensation of frequency dependent hearing loss, and connected to an output for outputting the processed digital sound signal.

Abstract: A method and apparatus for automatically adjusting the volume of a headset is described herein. The headset includes a speaker and a pressure transducer. The speaker projects audible signals into the ear canal, while the pressure transducer measures a sound pressure level in the ear canal. Based on the measured sound pressure level, a control system controls the volume of the audible sound projected from the speaker.

Abstract: A hearing aid comprises a microphone that receives incident sound waves from one or more sources external to the hearing aid, and converts the sound waves into electronic signals; a circuit that amplifies the electronic signals; a receiver that converts the amplified electronic signals into amplified sound waves; and a tuned resonant cavity between the microphone and the at least one external sound source. At least one parameter of the tuned resonant cavity is selected to modify the frequency response of the incident sound waves that are received by the microphone. In particular, the geometry of one or more openings through which sound waves enter the chamber, the geometry of the chamber itself, and/or the geometry of one or more openings through which sound waves exit the chamber, are selected to condition the incident sound waves by modifying the frequency response of the audio signal prior to the signal being received at the microphone.

Abstract: A hearing aid device includes a signal transforming unit for converting an analog sound signal received by a signal receiving unit to a digital signal; a memory unit for temporarily storing the digital signal; a sound processing module for reprocessing the sound signal stored in the memory unit via a microprocessing unit according to parameters set by a user; a sound control module for receiving the sound signal processed by the sound processing module and adjusting an output volume via the microprocessing unit; and a signal output unit for receiving and outputting the sound signal adjusted by the sound control module. By such arrangement, a speed, tone and/or volume of an output sound can be controlled using the hearing aid device to thereby help the user to receive and recognize the sound signal.

Abstract: This invention relates to a method for the individualized adaptation of a hearing aid to a person. The method consists basically of the measurement and quantification by parameters of the loudness perception of the individual, weighted by a first factor. Also weighted is a standardized normal loudness perception and its parameters by a second factor. Finally, the weighted loudness perceptions and their parameters are used for determining the optimal settings of the hearing aid for the individual concerned. The advantage of the method according to this invention lies in the fact that it permits significantly better adaptation of the hearing aid to the individual person.

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 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: 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: A input signal for a hearing device signal processing unit is generated with the aid of an electro-acoustic converter. An output signal is generated by applying signal amplification. The signal amplification is adapted by determining a reference level of the input signal using statistical calculation methods. A minimum input signal level is determined by subtracting a predefined difference value from the reference level of the input signal. An amplification function is applied to the input signal when the signal level is less than the minimum input signal level. This amplification function is different from an amplification function applied when the signal level is greater than the minimum input signal level.

Abstract: At a beamformer with at least two acoustical/electrical converters (M1, M2) the outputs (A1, A2) thereof are operationally connected to a beamformer unit (12). There signals dependent on signals (S1, S2) arising at said outputs (A1, A2) are co-processed to result in a beamformer output signal (Sa) dependent on both output signals of the converters. Frequency roll-off of the output signal (Sa) is counteracted by establishing a gain mismatch (10) of the two gains between the acoustical input signal (A) on one hand and the inputs to unit 12.

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 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 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: 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: 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: An apparatus for and method of employing an electronic hearing aid device to assist a hearing impaired patient. The resonance curve of the outer auditory canal of the patient is determined. A device in accordance with the present invention is tuned to a frequency response curve which matches the measured resonance curve. The device is tuned by adjusting the overshoot of a low pass filter stage which is interposed between the microphone input and a class D output stage.

Abstract: A hearing aid device providing instantaneous gain compression for sound signals and adaptive control of nonlinear waveform distortion, the device comprising: (a) at least one bandpass nonlinearity (BPNL) amplifier comprising a first bandpass filter, a second bandpass filter, and a memoryless nonlinear (MNL) compressive audio amplifier configured to receive a sound signal from the first bandpass filter and provide an MNL compressive audio amplifier output to the second bandpass filter, wherein the MNL compressive audio amplifier is configured to produce the MNL compressive audio amplifier output by providing memoryless gain compression directly on a sound signal that is (1) received from the first bandpass filter and (2) exhibits instantaneous amplitudes greater than a compression threshold, the BPNL amplifier thereby producing a desired gain compression on the received sound signal at an output of the second bandpass filter, and (b) a controller in communication with the BPNL amplifier, the controller being c

Abstract: The invention provides a canal hearing device and method in which the device is implemented with a mode of operation that provides acoustic transparency as well as a power-saving function, particularly useful to permit the user to wear the device in the ear canal during periods of sleep or inactivity without substantial loss of normal unaided response. The transparent mode has an in-situ acoustic transfer function that compensates for the insertion loss caused by the presence of a hearing device in the ear canal. While the device is in this transparent mode, its acoustic transfer function gives the user a perception of unaided hearing, as though the device were removed, when it is actually being worn continuously in the ear canal. Current drain of the device is significantly reduced as the transparent mode serves to shut off or reduce bias currents of at least one circuit element within the device circuitry.

Abstract: An improved hearing aid, and processes for adaptively processing signals therein to improve the perception of desired sounds by a user thereof. In one broad aspect, the present invention relates to a process in which one or more signal processing methods are applied to frequency band signals derived from an input digital signal. The level of each frequency band signal is computed and compared to at least one plurality of threshold values to determine which signal processing schemes are to be applied. In one embodiment of the invention, each plurality of threshold values to which levels of the frequency band signals are compared, is derived from a speech-shaped spectrum. Additional measures such as amplitude modulation or a signal index may also be employed and compared to corresponding threshold values in the determination.

Abstract: The invention relates first of all to a method for identifying a transient acoustic scene, said method including the extraction, during an extraction phase, of characteristic features from an acoustic signal captured by at least one microphone (2a, 2b), and the identification, during an identification phase, of the transient acoustic scene on the basis of the extracted characteristics. According to the invention, at least auditory-based characteristics are identified in the extraction phase. Also specified are an application of the method per this invention and a 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 hearing compensation system for the hearing impaired comprises a plurality of bandpass filters having an input connected to an input transducer and each bandpass filter having an output connected to the input of one of a plurality of multiplicative AGC circuits whose outputs are summed together and connected to the input of an output transducer. The multiplicative AGC circuits attenuate acoustic signals having a constant background level without the loss of speech intelligibility. The identification of the background noise portion of the acoustic signal is made by the constancy of the envelope of the input signal in each of the several frequency bands. The background noise that will be suppressed includes multi-talker speech babble, fan noise, feedback whistle, florescent light hum, and white noise.

Abstract: For the automatic choice of a hearing program, a hearing aid (1) detects whether it is located in the immediate vicinity of an external transmitter (10, 10?). The transmitter (10, 10?) generates a transmitter-specific signal so that an assignment of different transmitters can be made. In addition, in the case of the hearing aid (1), the current time of day and the day of the week can also influence the choice of the active hearing program.