Abstract: A system and method is provided for confirming the identity of a user, where the confirmation is made using biometric data. The system includes an internal speaker having an audio port, which is positioned within the ear of the user, and is adapted for producing a signal having one or more frequencies. The system further includes an internal microphone having an audio port, which is positioned within the ear of the user, and is adapted for detecting a resulting signal including the signal produced by the internal speaker and any corresponding signal reflections. A determination is then made as to whether the resulting signal detected by the internal microphone matches the corresponding predetermined expected signal, based upon a prestored hearing profile.

Abstract: There is described a hearing aid comprising input transducer means for converting input acoustic signals into electrical input signals, signal processor means and output transducer means. The signal processor means is operable to divide said electrical input signals into a plurality of frequency bands and to perform a contrast enhancement operation in each said frequency band, to increase the difference between the amplitudes of those frequency components of the input electrical signals having a relatively high amplitude and those frequency components thereof having a relatively low amplitude to produce output electrical signals in each said respective frequency bands. The output transducer means is operable to produce an output acoustic signal corresponding to a combination of said output electrical signals.

Abstract: The stimulation provided in the electrically stimulated cochlea is modulated in accordance with the amplitude of a received acoustic signal and the onset of a sound in a received acoustic signal to provide increased sound perception. An onset time that corresponds to the onset of a sound is detected in an acoustic signal associated with a frequency band. A forcing voltage and a transmitting factor are determined, wherein the forcing voltage and the transmitting factor are associated with the frequency band at the detected onset time. The acoustic signal is modulated as a function of the forcing voltage and the transmitting factor to generate an output signal. The generated output signal can be used to stimulate the cochlea. The modulation strategy can be used in conjunction with sound processing strategies that employ frequency modulation, amplitude modulation, or a combination of frequency and amplitude modulation.

Abstract: An ear sensor apparatus includes a microphone positionable with respect to an ear canal to detect sound signals from the ear canal and to generate a corresponding output signal. The apparatus may include a seal for substantially blocking ambient sound signals from reaching the microphone and an interface to convey the output signal to a remote speech recognition apparatus. The output signal may be normalized so that the normalized output signal is more representative of speech from a user's mouth than speech detected from the user's ear.

Abstract: The present invention provides a hearing aid in which at least one portion of an input sound signal is divided into a frequency band signal, and a single or a plurality of the frequency band signals are subjected to noise to generate a Noise-Vocoded Speech Sound signal, which a user can hear. Such a hearing aid facilitates activation of the brain and is expected to provide an effect on treatment or training of people with a neural disorder. Such a hearing aid lets a Noise-Vocoded Speech Sound signal recognized by utilizing normal portions of the brain to the maximum level and the Noise-Vocoded Speech Sound signal is compensated for by other normal portions of the brain, in order to let a person with hearing difficulty understand the meaning of the input sound.

Abstract: A method and apparatus configure a handheld audio device to communicate audio information by identifying an ear being used with the handheld audio device and by configuring the handheld audio device in response to the ear identification to communicate audio information to the ear. The identification may use sonic or visual techniques to identify the ear.

Abstract: An adaptive place-pitch ranking procedure for use with a cochlear implant or other neural stimulation system provides a systematic method for quantifying the magnitude and direction of errors along the place-pitch continuum. The method may be conducted and completed in a relatively short period of time. In use, the implant user or listener is asked to rank the percepts obtained after a sequential presentation of monopolar stimulation pulses are applied to a selected spatially-defined electrode pair. Should the patient's judgment of pitch order be correct for all applied interrogations, then no further testing involving the tested electrode pair (two electrode contacts) is undertaken. However, should there be errors in the place-pitch ranking, which errors evidence perceptual place-confusions, then a search is undertaken for the spread of the perceptual confusion by separating the target channel and competing channel by one electrode contact at a time.

Abstract: A method for reducing oscillation of a feedback signal in a hearing aid and hearing aid configured according to the present method is provided. The method includes the steps of determining the phase of the feedback signal over a feedback path of the hearing aid and shifting only the phase of the feedback signal a predetermined amount, without modification of other signal characteristics, to achieve a non-zero net phase of the feedback signal over the feedback path such that oscillation of the signal is prevented. In one embodiment of the present method, the step of determining the phase may be performed at the time of fitting of the hearing aid to a patient. In another embodiment of the present method, the method includes the step of periodically determining the phase of the feedback signal over the feedback path such that the phase shifting may be performed based on the periodically determined phase.

Abstract: A method and implementation for detecting and characterizing audible transients in noise includes placing a microphone in a desired location, producing a microphone signal wherein the microphone signal is indicative of the acoustic environment, processing the microphone signal to estimate the acoustic activity that takes place in the human auditory system in response to the acoustic environment, producing an excitation signal indicative of the estimated acoustic activity, processing the excitation signal to identify each impulsive sound frequency-dependent activity as a function of time, producing a detection signal indicative of audible impulse sounds, processing the detection signal to identify an audible impulsive sound, and characterizing each impulsive sound.

Abstract: A hearing enhancement system that enhances a user's hearing based on a directional speaker is disclosed. In one embodiment, the system can include an interface unit that has a directional speaker and a microphone. The microphone captures input audio signals that are transformed into ultrasonic signals. The speaker transmits the ultrasonic signals, which are transformed into output audio signals by interaction with air. At least a portion of the output audio signals are modified to enhance the hearing of the user. Based on the system, the user's ear remains free from any inserted objects and thus is free from annoying occlusion effects. Compared to existing hearing aids, the system is relatively inexpensive. In another embodiment, the system can also be used as a phone. In yet another embodiment, the system can also access audio signals from other portable or non-portable instruments, wired or wirelessly, such as from home entertainment units, phones, microphones at a conference or speakers at a movie theater.

Abstract: A hearing device wearer is to have improved possibilities of being able to perform a fine adjustment of his/her hearing device according to a basic setting. To this end, each processing value of a multi-channel processing system is standardized, in particular a filter bank, to a respectively associated basic setting value. A fine adjustment of the processing values to the hearing device wearer can now be carried out in relation to the standardized processing values starting from a standardized base line. It is thus possible for the hearing device wearer to implement a standardization of the setting values at any point in time and based hereupon to intuitively perform his/her setting requests.

Abstract: A wireless hearing assist device including a microphone unit and a receiver unit. The microphone unit includes a microphone that receives audio signals, a frequency profile adjustment control circuit that adjusts frequency profile of the received audio signals based on input by a user, and a transmitter that wirelessly transmits the adjusted audio signals. The receiver unit is adapted to be worn by the user and includes a receiver that wirelessly receives the adjusted audio signals transmitted by the transmitter, and a speaker that generates sound based on the adjusted audio signals. In at least one embodiment, the microphone unit includes a Bluetooth enabled RF communication link for wireless communication with an external device other than the receiver unit.

Abstract: In accordance with an embodiment, the invention provides a spectral enhancement system that includes a plurality of distributed filters, a plurality of energy distribution units, and a weighted-averaging unit. At least one of the distributed filters receives a multi-frequency input signal. Each of the plurality of energy-detection units is coupled to an output of at least one filter and provides an energy-detection output signal. The weighted-averaging unit is coupled to each of the energy-detection units and provides a weighted-averaging signal to each of the filters responsive to the energy-detection output signals from each of the energy-detection units to implement distributed gain control. In an embodiment, the energy detection units are coupled to the outputs of the filters via a plurality of differentiator units.

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: A hearing enhancement system that enhances a user's hearing based on a directional speaker is disclosed. In one embodiment, the system can include an interface unit that has a directional speaker and a microphone. The microphone captures input audio signals that are transformed into ultrasonic signals. The speaker transmits the ultrasonic signals, which are transformed into output audio signals by interaction with air. At least a portion of the output audio signals are modified to enhance the hearing of the user. Based on the system, the user's ear remains free from any inserted objects and thus is free from annoying occlusion effects. Compared to existing hearing aids, the system is relatively inexpensive. In another embodiment, the system can also be used as a phone. In yet another embodiment, the system can also access audio signals from other portable or non-portable instruments, wired or wirelessly, such as from home entertainment units, phones, microphones at a conference or speakers at a movie theater.

Abstract: For hearing apparatus and especially for hearing devices a tone balance unit is to be able to be used even under difficult general conditions. Therefore a hearing apparatus with a signal processing device, through which signals are able to be processed in a number of frequency channels, and a control device for controlling the level of the individual frequency channels are provided. The setting device features an unsymmetrical tone balance unit related to a predetermined frequency with which at least one lower and at least one upper channel of the number of frequency channels related to a predetermined frequency is able to be simultaneously unsymmetrically adjusted relative to the predetermined frequency. Thus for example a tone balance unit can also be used with a low feedback limit in the upper frequency range.

Abstract: A hearing aid (50) comprises means (55, 56, 57, 58) for reproducing frequencies above the upper frequency limit of a hearing impaired user. The hearing aid (50) according to the invention comprises means (55, 57) for transposing higher bands of frequencies from outside the upper frequency limit of a hearing impaired user down in frequency based on a detected frequency in order to coincide with a lower band of frequencies within the frequency range perceivable by the hearing impaired user. The transposing means (55, 57) comprise an adaptive notch filter (15) for detecting a dominant frequency in the lower band of frequencies, adaptation means (16) controlled by the adaptive notch filter (15), an oscillator (3) controlled by the adaptation means (16), and a multiplier (4) for performing the actual frequency transposition of the signal. The invention further provides a method for processing a signal in a hearing aid.

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: It is an object of the present invention to provide a sound processing apparatus which can allow a user to hear a sound with improved intelligibility even if the sound is hard to hear. The analyzing means 15 is adapted to analyze the input signal from the A/D converter 12, to detect, on the basis of an analysis of the input signal, a frequency band corresponding to a masking sound and a frequency band corresponding to a masked sound, and to change the cutoff frequencies of the lowpass and highpass filters 13 and 14 on the basis of the analysis of the input signal to ensure that the frequency band corresponding to a masking sound is included in a signal from one of the first and second sound output means 18 and 19, and the frequency band corresponding to a masked sound is included in a signal from the other of the first and second sound output means 18 and 19.

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: 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 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: Determining coefficients of a compression amplification gm, which describe a frequency-dependent adaptation of the input signal in accordance with frequency-dependent signal levels of the input signal, determining coefficients of a noise suppression am, which describe a frequency-dependent adaptation of the input signal in accordance with interference noises detected in the input signal, and the calculation of the coefficients of the filter (6) cm out of the coefficients of the compression amplification gm and the coefficients am of the noise suppression.

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

Abstract: An audio signal processor includes a tone control (23). The tone control comprises two low-pass filters (221, 222) operating in current-mode and a subtractor (223) which subtracts the output currents of the filters to produce a band-pass characteristic. Each filter is a tuneable log-domain current-mode filter comprising MOS transistors operating in weak inversion. The tone control is useful in audio signal processors, hearing aids and single-channel and multi-channel Cochlear implants.

Abstract: This invention relates to a hearing aid system (100, 200, 300) for generating auditory spatial cues. The hearing aid system (100, 200, 300) comprises a first microphone unit (306) adapted to convert sound received at a first microphone (102) and received at a second microphone (104), a first delay unit (106) connected to the first microphone (102) delaying the signal from the first microphone (102), a first calculation unit (108) for summing the delayed signal of the first microphone (102) and signal of the second microphone (104), a processor unit (110) processing the summed signal, and a speaker converting the processed signal to a processed sound. The first and second microphones (102, 104) are separated by a predetermined first distance and the first delay unit (106) provides a predetermined first delay thereby generating a first auditory spatial cue representing a first spatial dimension in the summed signal.

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

Abstract: A 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: A method is described to operate a hearing device with an input transducer (1), a signal processing unit (2) and an output transducer (4). The method comprises the steps of converting an acoustic input signal into a converted input signal, processing the converted input signal in a main signal path in order to obtain a main output signal, and supplying the main output signal to an output transducer. By processing the converted input signal in a side signal path to obtain a side path output signal, and by superimposing the side path output signal on the main output signal, wherein a group delay of a signal traveling through the side signal path is smaller than a group delay of a signal traveling through the main signal path, the localization problems are eliminated. At the same time, the hearing device according to the present invention can still have a very high performance. In short terms, a “zero-delay-high-performance” hearing device has been created by the present invention.

Abstract: A system and method for processing an acoustic input signal and providing at least one output acoustic signal to a user of a hearing-aid system. The hearing-aid system includes first and second channels with one of the channels having an adaptive delay. The first channel includes a directional unit for receiving the acoustic input signal and providing a directional signal; a correlative unit for receiving the directional signal and providing a noise reduced signal by utilizing correlative measures for identifying a speech signal of interest in the directional signal; and, a compensator for receiving the noise reduced signal and providing a compensated signal for compensating for a hearing loss of the user.

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: A method for adjusting a hearing device is disclosed in which one of several hearing programs can be selected, as well as a hearing device system to perform the method. The method comprises the steps of identifying a momentary acoustic surround situation and of arranging the selectable hearing programs according to the identified momentary acoustic surround situation. Therewith, the hearing device user is not obliged to know a predefined sequence of the selectable hearing programs; instead the hearing device user may easily select a hearing program which is regarded as the best one for the momentary acoustic situation. The selection is offered by the hearing device by analyzing the momentary acoustic situation. In addition, the hearing device user may easily select other hearing programs, namely by generating an activation signal, for example by pressing a button at the hearing device itself or at a remote control.

Abstract: An adaptive place-pitch ranking procedure for use with a cochlear implant or other neural stimulation system provides a systematic method for quantifying the magnitude and direction of errors along the place-pitch continuum. The method may be conducted and completed in a relatively short period of time. In use, the implant user or listener is asked to rank the percepts obtained after a sequential presentation of monopolar stimulation pulses are applied to a selected spatially-defined electrode pair. Should the patient's judgment of pitch order be correct for all applied interrogations, then no further testing involving the tested electrode pair (two electrode contacts) is undertaken. However, should there be errors in the place-pitch ranking, which errors evidence perceptual place-confusions, then a search is undertaken for the spread of the perceptual confusion by separating the target channel and competing channel by one electrode contact at a time.

Abstract: A system for processing audio signals comprises a sequence of digital filters each configured to process a selected frequency using a set of coefficients. A filter configured to process a certain frequency shares its coefficients with another filter that processes a frequency that is lower than the first frequency by at least one frequency interval, such as an octave. The first filter samples at a certain sampling rate, and the second filter's sampling rate is determined by multiplying the first sampling rate by the ratio of the second frequency to the first frequency. The filters are evenly grouped into frequency intervals, such as octaves. Filters in an octave are sampled at a sampling frequency that is at least twice as high as the highest frequency processed in that octave.

Abstract: An improved processing approach is disclosed in order to allow for different rates of stimulation to be used for different electrodes in a multi-electrode cochlear implant. When the incoming signal is processed by filter array (35), each channel is processed to determine amplitude (37) and to estimate the period of the signal in that channel (39). The amplitude and period information is used to determine which electrode is stimulated, and the timing of that stimulation.

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: In a hearing aid having an oscillation detector and a method for establishing the presence of oscillations in a hearing aid, sinusoidal input signals of the microphone can be detected, and so oscillations that are present can also be detected. To this end, the number of digitized sample values in consecutive periods of the input signal is determined, and a long-term average value NL and a short-term average value NK are formed from these numbers. When NL and NK are essentially identical, the presence of oscillations is detected.

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 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 speech segment is convolved with the enhancement filter in the warped time-domain to give the processed output signal. Processing artifacts are reduced since the frequency-warped system has no temporal aliasing.

Abstract: A method to control the dynamic range of a hearing device, which comprises at least one acoustic/electric input transducer followed by a signal processing unit which in turn is operationally connected to an electric/mechanical output transducer, by selectively switching an input impedance of the electric/mechanical transducer from one value to another.

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

Abstract: Improved approaches are disclosed to filter and compress sound signals so as to achieve not only speech audibility and intelligibility at low levels but also preserves spectrum contrast at high levels. According to one aspect of the invention, gain amounts for different frequency bands are individually constrained based on signal levels for the frequency bands. Hence, the gain amounts for each of the frequency bands may or may not be constrained depending on the corresponding signal levels. As a result, the most critical information for speech intelligibility, speech clarity, and speech quality can be made available to hearing impaired people over wide range of signal level. The invention is particularly useful for hearing aids or other sound systems for the hearing impaired.

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.

Abstract: Methods and devices for audio amplification suitable for hearing aid, hearing aid fitting, and diagnostic purposes include audio amplification having at least one variable gain channel configured to provide relatively higher gain at low levels, rapid gain compression at intermediate levels converging to linear gain at high signal levels, and slow feedback control of the compressive gain. Several such audio channels may be provided in a hearing aid or diagnostic device, and instantaneous gain compression is preferred. An analog implementation provides nonlinear elements in a feedback path to simulate a multiple feedback band-pass non-linearity cochlear filterbank hearing model (MFBPNL), while a digital implementation uses logarithmic representations of signals to minimize functional components in a multiple band-pass non-linearity cochlear filterbank hearing model (MBPNL). When used as a hearing aid, annoying amplification of weak sounds during brief interruptions of sustained intense sounds is prevented.

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: 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.