Abstract: A method for manufacturing an acoustical device, especially a hearing device. A device casing is provided with an acoustical/electrical input converter arrangement with an electric output. An audio signal processing unit establishes audio signal processing of the device according to individual needs and/or purpose of the device. At least one electrical/mechanical output converter is provided. A filter arrangement with adjustable high-pass characteristic has a control input for the characteristic. The following operational connections are established: between the output of the input converter arrangement and the input of the filter arrangement, between the output of the filter arrangement and the control input, between said output of the filter arrangement and the input of the processing unit, between the output of the processing unit and the input of the at least one output converter.

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: According to the present invention, a three-line bus is used for communication between a first digital unit (1) and two other digital units (3). The three-line bus includes a system clock line (SCL) as well as a data transmission line (SD) through which data is sent from the two other units (3), i.e. transmitting units, to the first unit (1), i.e. receiving unit. The system uses an authorization line (WS) for determining which of the two transmitting units (3) is capable of writing data on the data transmission line (SD) and when it can do so. The communication from the first unit (1), now operating as a transmitter, to the two other units (3), now operating as receivers, is made possible by the fact that data signals are also injected in the authorization line (WS) and transmitted through the line together with the authorization control signals.

Abstract: The invention concerns a method of receiving and processing both audio and non-audio signals for use in a hearing aid or other listening device comprising converting electromagnetic radiation to electrical signals, amplifying said electrical signals to produce amplified signals, and filtering the amplified signals in an audio frequency range. The invention extends to a novel integrated amplified telecoil system for carrying out the method.

Abstract: A fitting method is provided for a multichannel hearing aid with at least one low frequency channel having an individually adjustable compressor. The method comprises the steps of first adjusting the characteristic of the compressor according to the hearing loss to be compensated by the hearing aid, followed by the step of increasing the compression ratio of the characteristic of the compressor in the at least one low frequency band. The at least one low frequency channel may further comprise an offset amplifier adding an offset gain to the compressor characteristic, and the method may further comprise the step of adjusting the offset gain in the range from ?20 dB to 20 dB. After adjustment according to the method, compressors operating at low frequencies enhance low level signals and attenuate high level signals whereby perception of occlusion is suppressed.

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

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

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

Abstract: A hearing instrument system detects the insertion or removal of a hearing instrument into a space and includes first and second acoustic transducers, first and second level detection circuitry, and signal processing circuitry. The first acoustic transducer is configured to receive a first electrical signal and in response radiate acoustic energy, and the second acoustic transducer is configured to receive radiated acoustic energy and in response generate a second electrical signal. The first level detection circuitry is operable to receive the first electrical signal and generate a first intensity signal, and the second level detection circuitry is operable to receive the second electrical signal and generate a second intensity signal. The signal processing circuitry is operable to receive the first and second intensity signals and compare the first and second intensity signals and determine whether the hearing instrument is inserted into the space or removed from the space based on the comparison.

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

Abstract: Scheme for processing an input signal A(t) by N resonators (17), each having parameters characterizing it, to generate N individual output signals. Then each of the N individual output signals is weighted using a corresponding weight to generate N individual weighted output signals which are superposed to obtain M output signals C(t). One of the parameters or the weight depend on a time signal P(t).

Abstract: Acoustical signals from the acoustical surrounding (U) which impinge upon a reception unit 30 are evaluated and direction of arrival (DOA) of such signals is determined. From signals indicative of such direction of arrival (DOA) a histogram is formed in unit 32. The behaviour of such histogram is classified under different aspects or criteria and dependent on classification results in a classifying unit 34 the hearing device and thereby especially its signal transfer characteristic from input acoustical signals to output mechanical signals is controlled or adjusted.

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

Abstract: The present invention is related to a method for frequency transposition in a hearing device 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). According the present invention, 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 into a lower frequency range, namely in an audible range of the hearing impaired.

Abstract: In one aspect, the invention divides an audio signal into auditory events, each of which tends to be perceived as separate and distinct, by calculating the spectral content of successive time blocks of the audio signal (5-1), calculating the difference in spectral content between successive time blocks of the audio signal (5-2), and identifying an auditory event boundary as the boundary between successive time blocks when the difference in the spectral content between such successive time blocks exceeds a threshold (5-3). In another aspect, the invention generates a reduced-information representation of an audio signal by dividing an audio signal into auditory events, each of which tends to be perceived as separate and distinct, and formatting and storing information relating to the auditory events (5-4). Optionally, the invention may also assign a characteristic to one or more of the auditory events (5-5).

Abstract: The present invention provides a cochlear stimulation system and method for capturing and translating fine time structure (“FTS”) in incoming sounds and delivering this information spatially to the cochlea. The system comprises a FTS estimator/analyzer and a current navigator. An embodiment of the method comprises analyzing the incoming sounds within a time frequency band, extracting the slowly varying frequency components and estimating the FTS to obtain a more precise dominant FTS component within a frequency band. After adding the fine structure to the carrier to identify a precise dominant FTS component in each analysis frequency band (or stimulation channel), a stimulation current may be “steered” or directed, using the concept of virtual electrodes, to the precise spatial location (place) on the cochlea that corresponds to the dominant FTS component. This process is simultaneously repeated for each stimulation channel and each FTS component.

Abstract: A hearing aid includes a signal path having an input transducer, a processor and an output transducer, the hearing aid further including a radio frequency receiver, where the receiver includes a single crystal oscillator providing a single oscillator frequency and where means are provided for generating a further number of receiving frequencies by transforming the oscillator frequency to the desired receiving frequencies. The hearing aid and the RF receiver may be separate or integrated. The invention further relates to a separate element comprising a RF receiver, the separate element being adapted for mounting on a hearing aid.

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

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

Abstract: To provide a hearing aid that allows desired sound to be selectively heard and allows unnecessary sound not to be heard, and that can be used with no bars even in a situation like a meeting that requires a conversation with a plurality of persons. The hearing aid includes: a hearing aid body having a hearing aid function and an external microphone signal receiving function; and a separate external microphone, wherein connection and disconnection of communication between the above described hearing aid body and the external microphone are automatically switched, and the above described hearing aid body functions as a normal hearing aid when the communication between the above described hearing aid body and the external microphone is disconnected.

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

Abstract: A hearing aid including a radio frequency receiver, where the receiver includes a single oscillator providing a single crystal oscillator frequency and where means are provided for generating a further number of receiving frequencies by transforming the oscillator frequency to the desired receiving frequencies, where further scanning means are provided for upon activation provide a scanning of the possible frequency area and select a detected frequency superceding a predetermined detector level.

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

Abstract: A hearing aid that compensates for a patient's hearing deficit in a gradually progressing fashion. The hearing aid may be programmed to successively select in a defined sequence a parameter set that defines at least one operating characteristic of the signal processing circuit from a group of such parameter sets. The defined sequence may end in a parameter set that optimally compensates the patient's hearing.

Abstract: An apparatus for reducing hearing aid radio frequency (RF) interference including a directional multi-band and/or single band antenna for use with PWDs such as digital cellphones is disclosed. The apparatus greatly reduces or eliminates the audio noise induced in hearing aids by the PWDs and allows operation of a hearing aid during PWD operation. In operation, the apparatus may be provided on the PWD side away from the user's head. The apparatus may be integrated into the PWB during its manufacture or provided as an after market assembly for a PWD that has a port for connection of an external antenna. The apparatus provides for improved front-to-back ratio as compared to antennas currently in use on PWD's, and therefore also reduces SAR (specific absorption rate), the level of RF energy received into the head by a PWD.

Abstract: The present invention is directed to a communication system wherein ultrasonic signals can be used as carriers to efficiently produce high fidelity, wide audio bandwidth sound. Exemplary embodiments rely on the airborne transport of an inaudible ultrasonic carrier directly into the ear canal of a user, such that the non-linearities within the ear itself can be exploited to demodulate the ultrasonic carrier without producing audible sounds at the input to the user's ear. The non-linearities of the ear itself, in conjunction with the human brain's perception of audible frequencies generated in response to ultrasonic stimulation, are relied upon to detect audio information. The ultrasound-to-audio-sound conversion in the confined volume of the inner ear appears to be constant pressure, as opposed to constant velocity, such that all frequencies of the audio bandwidth (including low frequency bass signals) are produced with comparable sound intensity.

Abstract: The present invention is directed to a communication system wherein bone conducted ultrasonic signals are used as carriers to efficiently produce high fidelity, wide audio bandwidth sound. Exemplary embodiments rely on the use of modulation techniques which can achieve high fidelity audible sound, such as carrier plus single sideband (SSB) modulation. Known non-linearities within the ear itself can be exploited to demodulate the modulated ultrasonic carrier without producing audible sounds at the input to the user's ear. The non-linearities of the ear itself, in conjunction with the human brain's perception of audible frequencies generated in response to modulated ultrasonic stimulation, are relied upon to detect audio information.

Abstract: For level limitation of a time-discrete signal in a digital hearing aid having at least an input transducer, an A/D converter, a signal processing unit and an output transducer, the sampling rate of the time-discrete signal is boosted in the signal path following the A/D conversion and signal values of the time-discrete signal with a boosted sampling rate are limited to a maximum value. Harmonics that arise upon limitation of the signal values that are folded back into the audible frequency range are thus suppressed.

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

Abstract: A filterbank structure is provided which provides a flexible compromise between the conflicting goals of processing delay, filter sharpness, memory usage and band interaction. The filterbank has an adjustable number of bands and a stacking which provides for a selectable shift of band frequencies to one of two discrete sets of center frequencies. The width of the bands and hence the number of the bands is selected depending upon acceptable delay, memory usage, and processing speed required. The flexibility in terms of stacking of the bands provides twice the number of potential band edge placements, which is advantageous for hearing loss fitting especially at low frequencies. The same filter coefficients can be used for analysis and synthesis, to reduce memory usage.

Abstract: The invention sets forth a resonance enhancing device for a microphone. The resonance enhancing device is adapted to direct a user's sound waves to a spherical microphone section of the microphone. The resonance enhancing device comprises a body. The body comprises a first end and a second end. The first end is designed for encircling a portion of the microphone and the second end of the body is designed for defining an opening that is disposed proximate the spherical microphone section of the microphone. Thus, when a user sings into the resonance enhancing device and the sound waves of the user are directed by the resonance enhancing device to the spherical microphone section of the microphone, thereby altering the resonance of the sound waves at the spherical microphone section of the microphone.

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

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

Abstract: A hearing aid device includes a microphone, an amplifier, and a speaker. The microphone has bandpass filters, which separate incoming acoustic signals into a multiple channels. Both the microphone and the speaker have a series of amplifying elements, each with a different amplification level. By choosing which amplifying elements to switch on in the microphone and the speaker, the wearer of the hearing aid device may adjust for different amplification levels in each channel.

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

Abstract: A dynamic range compression system is provided, using either a sample-by-sample or a block processing system. Such a system can be used, for example, in a hearing aid. The system, using a frequency-warped processing system, is comprised of a cascade of all-pass filters with the outputs of the all-pass filters providing the input to the frequency analysis used to compute the filter coefficients. The compression filter is then designed in the frequency domain. Using a compression filter having even symmetry guarantees that the group delay is constant and does not depend on the compression gains at any given time. Additionally, due to the use of all-pass filters, the compression filter group delay more closely matches human auditory latency. An inverse frequency transform back into the warped time domain is used to produce the compression filter coefficients that are convolved with the outputs of the all-pass delay line to give the processed output signal.

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

Abstract: A 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: A hearing aid includes a microphone for providing an electrical signal representative of a received acoustic signal, a means for filtering the electrical signal and a means for automatic switching. Upon detecting a presence of a magnetic field, the means for automatic switching enables the means for filtering to modify a frequency response of the hearing aid to boost a low frequency gain and reduce a high frequency gain. Such filtering can be used with or without a telecoil. A hearing aid including a telecoil is configured with a preamplifier, a inductive element, and a switch control integrated onto a single common circuit board.

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

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

Abstract: A fitting method is provided for a multichannel hearing aid with at least one low frequency channel having an individually adjustable compressor. The method comprises the steps of first adjusting the characteristic of the compressor according to the hearing loss to be compensated by the hearing aid, followed by the step of increasing the compression ratio of the characteristic of the compressor in the at least one low frequency band. The at least one low frequency channel may further comprise an offset amplifier adding an offset gain to the compressor characteristic, and the method may further comprise the step of adjusting the offset gain in the range from −20 dB to 20 dB. After adjustment according to the method, compressors operating at low frequencies enhance low level signals and attenuate high level signals whereby perception of occlusion is suppressed.

Abstract: A filterbank structure is provided which provides a flexible compromise between the conflicting goals of processing delay, filter sharpness, memory usage and band interaction. The filterbank has an adjustable number of bands and a stacking which provides for a selectable shift of band frequencies to one of two discrete sets of center frequencies. The width of the bands and hence the number of the bands is selected depending upon acceptable delay, memory usage, and processing speed required. The flexibility in terms of stacking of the bands provides twice the number of potential band edge placements, which is advantageous for hearing loss fitting especially at low frequencies. The same filter coefficients can be used for analysis and synthesis, to reduce memory usage.

Abstract: The present invention is directed to a hearing aid apparatus which can function in a hearing aid state and a communications state, by automatically switching between the hearing aid state and the communications state. Exemplary embodiments combine the functions of headset operation and hearing aid operation into an apparatus which can connect with communication devices and which can automatically reconfigure itself to function as a hearing aid for addressing a hearing impairment of the user. Exemplary embodiments can be configured small and comfortable to permit wear over extended periods of time.

Abstract: According to one embodiment of the present invention, a digital hearing device is disclosed. The digital hearing aid includes a microphone for receiving sound, which may include an analog signal. The analog signal is converted by a first converter into a digital signal. Filters are provided to divide the digital signal into multiple signal parts. A signal processor may be provided for each signal part, and performs signal processing on its respective signal part. An adder adds the output of the signal processors, which results in a processed digital signal. A second converter converts the processed digital signal back into an analog signal. A speaker then outputs the analog signal. According to another embodiment of the present invention, a method for enhancing sound is provided.