Abstract: A new subband feedback cancellation scheme is proposed, capable of providing additional stable gain without introducing audible artifacts. The subband feedback cancellation scheme employs a cascade of two narrow-band filters Ai(Z) and Bi(Z) along with a fixed delay, instead of a single filter Wi(Z) and a delay to represent the feedback path in each subband. The first filter, Ai(Z), is called the training filter, and models the static portion of the feedback path in ith subband, including microphone, receiver, ear canal resonance, and other relatively static parameters. The training filter can be implemented as a FIR filter or as an IIR filter. The second filter, BI(Z), is called a tracking filter and is typically implemented as a FIR filter with fewer taps than the training filter. This second filter tracks the variations of the feedback path in the ith subband caused by jaw movement or objects close to the ears of the user.

Abstract: A feedback cancellation system for a hearing instrument is described. The feedback cancellation system includes both a constrained adaptive filter, the constrained adaptive filter can be constrained by initiation coefficients, as well as an adaptive gain modification. The adaptive gain modification allows the feedback cancellation to respond to substantially different environments from the test environment used to derive the initiation coefficients for the constrained adaptive filter.

Abstract: In order to improve the feedback compensation in hearing d vices th xtent of compensation is controlled. An estimated signal is acquired with which the intensity of the feedback signal is estimated. The damping of the feedback signal is thus controlled using the estimated signal. With this, it is, among other things, possible to disconnect the feedback compensation given no present feedback such that artifacts can be prevented.

Abstract: A feedback compensator for a hearing aid device has a filter arrangement that splits a signal path, to implement an adaptive feedback compensation with only one buffer memory, two splitting nodes, and two addition nodes. The feedback compensation ensues only in the feedback-susceptible frequency range of the input signal. In addition to the filtering of the input signal, it is advantageous to feed to the adaptive feedback compensation filter a bandwidth-limited signal that is taken from the amplified output signal.

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 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 earset including a housing positionable with respect to an ear of a person, a microphone disposed with respect to the housing for insertion into the ear of a person, the microphone operable to detect a change in air pressure within the ear while the person speaks and to produce an electrical microphone signal corresponding to the internally detected change in air pressure and a speaker disposed with respect to the housing and operable to produce a sound corresponding to an electrical speaker signal. The earset also includes a circuit coupled to receive the microphone signal and the speaker signal and operable to produce a corrected microphone signal having a reduced feedback component of the microphone signal, the feedback component resulting from the detection by the microphone of the sound produced by the speaker to produce a corrected microphone signal.

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

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

Abstract: A bone conduction hearing aid includes a vibrator carried by the insertion end of the hearing aid. When the hearing aid is inserted into the ear canal of a patient, the vibrator is positioned in the ear canal adjacent the mastoid bone. A microphone receives sound waves and outputs a microphone signal to the hearing aid electronics where the microphone signal is amplified and then sent to the vibrator, causing the vibrator to vibrate. Vibrations produced by the vibrator are transferred to the opposite cochlea by way of the mastoid bone, enabling enhanced hearing perception in patients with hearing loss in one ear. Transfer of vibrations to the bones of the middle ear also assists patients with conductive pathology in one ear. The hearing aid may also function to enhance communication in high noise environments. Feedback from the vibrator to the microphone is eliminated electronically. Various alternate forms of feedback elimination are also contemplated by the invention.

Abstract: When a sound signal is received, a sound wave generated on an upper surface of a vibration plate is transmitted to a sound collection portion through a first air passage, and a sound wave generated on a lower surface of the vibration plate is transmitted to the sound collection portion through a second air passage. The sound waves generated above and below the upper and lower surfaces of the vibration plate are about 180 degrees out of phase from each other and substantially cancel each other which minimize or prevent howling.

Abstract: A circuit for adaptive suppression of acoustic feedback forms part of a digital hearing aid, comprising a microphone (1), subtracter (3), hearing correcting means (4), receiver (6), delay element (9), filter (10), updating unit (11), lattice decorrelators (12, 13) and control unit (14). The transmission path is modeled with the feedback characteristic (7) and an adder (8). First decorrelator (12) decorrelates the echo-compensated input signal (en) and second decorrelator (13) decorrelates the delayed output signal (xn) by using coefficients (kn) from first decorrelator (12). The coefficients (kn) of the two filters (12, 13) are calculated by adaptive decorrelation of the echo-compensated input signal (en). This permit maximum convergence rates for minimum distortions. Updating of the filter coefficients mainly takes place where the greatest amplifications occur in the hearing correcting means (4). The fed-back signal components are continuously removed from the input signal.

Abstract: The invention relates to a method for reducing feedback caused by electromagnetic interference between an induction coil and an output transducer, a voltage supply or an amplifier or a combination thereof, the method comprising: producing by means of a filter an equivalent of an electromagnetic feedback path in a system with a pickup coil and subtraction of the equivalent of the feedback signal from the input signal in order to obtain feedback reduction. The invention further relates to a device for implementing the method, the device comprising an induction coil and an output transducer, voltage supply or an amplifier or a combination thereof. The device further comprises filter means for generating an equivalent of an electromagnetic feedgack signal occurring between the induction coil and an output transducer, a voltage supply or an amplifier or a combination thereof and means for subtracting the equivalent from an input signal in order to obtain feedback reduction.

Abstract: A hearing aid includes an image sensing device, a sound input transducer, a sound output transducer, and a processor. The image sensing device detects an image from a selected region of a user of the hearing aid while the sound input transducer receives sound and produces an audio signal representative of the sound. The sound output transducer receives the audio signal and converts the audio signal to a sound wave that is provided to the user. The processor receives the image, analyzes the image to determine an existence of human-generated sound, and provides the audio signal from the sound input transducer to the sound output transducer when human-generated sound is detected. The audio signal is provided at a first level when human-generated sound is detected and is provided at a second level in an absence of human-generated sound.

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

Abstract: A feedback cancellation system with reduced sensitivity to low-frequency tonal inputs is provided. Such a system can be used, for example, in a hearing aid to prevent cancellation of the desired tonal inputs to the hearing aid, thus improving the gain at high frequencies of the hearing aid while simultaneously preserving the desired tonal inputs at low frequencies. The feedback cancellation system comprises a first adaptive filter block for adaptively filtering an error signal to remove the low-frequency tonal components from the error signal. The first adaptive filter block is constrained so that only low-frequency tones in the error signal are cancelled, thus enabling the feedback cancellation system to still cancel “whistling” at high frequencies due to the temporary instability of the hearing aid. A second adaptive filter block adaptively filters a feedback path signal to produce an adaptively filtered feedback path signal.

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: Feedback cancellation apparatus uses a cascade of two filters along with a short bulk delay. The first filter is adapted when the hearing aid is turned on in the ear. This filter adapts quickly using a white noise probe signal, and then the filter coefficients are frozen. The first filter models parts of the hearing-aid feedback path that are essentially constant over the course of the day. The second filter adapts while the hearing aid is in use and does not use a separate probe signal. This filter provides a rapid correction to the feedback path model when the hearing aid goes unstable, and more slowly tracks perturbations in the feedback path that occur in daily use. The delay shifts the filter response to make the most effective use of the limited number of filter coefficients.

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: For reducing feedback-conditioned oscillations in a hearing aid device, microphone signals of a first microphone and of a distanced, second microphone are compared to one another. When oscillations are detected at the same frequency in both microphone signals, these oscillations are determined to be useful (non-feedback) tonal signals. Oscillations that are only present in one of the microphone signals, in contrast, are feedback-conditioned and are suppressed using suitable measures.

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: An earset including a housing positionable with respect to an ear of a person, a microphone disposed with respect to the housing for insertion into the ear of a person, the microphone operable to detect a change in air pressure within the ear while the person speaks and to produce an electrical microphone signal corresponding to the internally detected change in air pressure and a speaker disposed with respect to the housing and operable to produce a sound corresponding to an electrical speaker signal. The earset also includes a circuit coupled to receive the microphone signal and the speaker signal and operable to produce a corrected microphone signal having a reduced feedback component of the microphone signal, the feedback component resulting from the detection by the microphone of the sound produced by the speaker to produce a corrected microphone signal.

Abstract: The present invention combines audio compression and feedback cancellation in an audio system such as a hearing aid. The feedback cancellation element of the present invention uses one or more filters to model the feedback path of the system and thereby subtract the expected feedback from the audio input signal before hearing aid processing occurs. The hearing aid processing includes audio compression, for example multiband compression. The operation of the audio compression element may be responsive to information gleaned from the feedback cancellation element, the feedback cancellation may be responsive to information gleaned from the compression element, or both.

Abstract: The present invention combines audio compression and feedback cancellation in an audio system such as a hearing aid. The feedback cancellation element of the present invention uses one or more filters to model the feedback path of the system and thereby subtract the expected feedback from the audio input signal before hearing aid processing occurs. The hearing aid processing includes audio compression, for example multiband compression. The operation of the audio compression element may be responsive to information gleaned from the feedback cancellation element, the feedback cancellation may be responsive to information gleaned from the compression element, or both.

Abstract: In a method for feedback recognition in a hearing aid and a hearing aid operating according to the method, a frequency band is defined, a first signal level in the frequency band is determined, the signal on a signal transmission path of the hearing aid is attenuated, and a second signal level of the attenuated signal in the frequency band is determined, and feedback is recognized on the basis of the identified first and second signal levels.

Abstract: A feedback cancellation system for a hearing aid or the like adapts a first filter in the feedback path that models the quickly varying portion of the hearing aid feedback path, and adapts a second filter in the feedback path that is used either as a reference filter for constrained adaptation or to model more slowly varying portions of the feedback path. The second filter is updated only when the hearing aid signals indicate that an accurate estimate of the feedback path can be obtained. Changes in the second filter are then monitored to detect changes in the hearing aid feedback path. The first filter is adaptively updated at least when the condition of the signal indicates that an accurate estimate of physical feedback cannot be made. It may be updated on a continuous or frequent basis.

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

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

Abstract: A hearing aid with beam forming properties, having at least two microphone channels (1a, 1b) for at least two microphones (2a, 2b), the microphone channels each comprising an analog to digital converter (3a, 3b) and having at least one programmable or program controlled signal processor (5), as well as a digital to analog converter, and at least one receiver and a battery for power supply. The hearing aid particularly comprises in each microphone channel (1a, 1b) a sigma-delta-type analog to digital converter (3a, 3b) including a digital low pas filter and a decimator 94) for converting a 1 Bit Stream of a high clock frequency into a digital word sequence of a lower clock frequency. At least one of the at least two microphone channels contains a controllable delay device (6) connected to the input side of the respective digital low pass filter and decimator (4) of the channel, the delay device (6) being controllable by the at least one signal processor (5).

Abstract: A bobbin unit includes a base portion and a coil winding portion on which a conductor is wound. A hole is formed in a central part of the base portion to allow an armature to pass through. Coil side terminals to which ends of the conductor are fixed and signal input terminals to which an external signal is entered are provided at opposite ends of the base portion with the hole located in between. The coil side terminals are electrically connected to their corresponding signal input terminals inside the base portion. A coil is formed on the coil winding portion by winding the conductor. The ends of the conductor are wound around the respective coil side terminals and fixed thereto by arc welding. Since the conductor is electrically connected to the signal input terminals when the ends of the conductor are fixed to the coil side terminals, connection of the ends of the conductor constituting the coil to the signal input terminals is made easy.

Abstract: Feedback cancellation apparatus uses a cascade of two filters along with a short bulk delay. The first filter is adapted when the hearing aid is turned on in the ear. This filter adapts quickly using a white noise probe signal, and then the filter coefficients are frozen. The first filter models parts of the hearing-aid feedback path that are essentially constant over the course of the day. The second filter adapts while the hearing aid is in use and does not use a separate probe signal. This filter provides a rapid correction to the feedback path model when the hearing aid goes unstable, and more slowly tracks perturbations in the feedback path that occur in daily use. The delay shifts the filter response to make the most effective use of the limited number of filter coefficients.

Abstract: Feedback cancellation apparatus uses a cascade of two filters along with a short bulk delay. The first filter is adapted when the hearing aid is turned on in the ear. This filter adapts quickly using a white noise probe signal, and then the filter coefficients are frozen. The first filter models parts of the hearing-aid feedback path that are essentially constant over the course of the day. The second filter adapts while the hearing aid is in use and does not use a separate probe signal. This filter provides a rapid correction to the feedback path model when the hearing aid goes unstable, and more slowly tracks perturbations in the feedback path that occur in daily use. The delay shifts the filter response to make the most effective use of the limited number of filter coefficients.

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

Abstract: Again control arrangement for an auditory prosthesis, such as a hearing aid or cochlear prosthesis, is disclosed. The output of a controlled amplifier 11 is processed so as to derive an average noise floor level. This is compared to a predetermined level, and the gain adjusted slowly up or down in response to the comparison. Hardware and software implementations are disclosed.

Abstract: A problem in hearing aids or other audio/acoustic amplifier circuits is that external sources of EM energy may be coupled into the electronics of the hearing aid so as to contribute to the acoustic output. The invention provides a circuit for removing the effects of EM interference. A separate reference generator is used to detect the external EM energy. This is fed into an interference canceller which may be adaptive, which effectively removes the unwanted component in the hearing aid signal, leaving only a signal representative of the desired acoustic output.

Abstract: A hearing aid has a microphone (1), or other electromechanical transducer, for converting an acoustic input signal into an electrical signal, a signal-processing and amplifying signal path (2, 3, 13, 4, 5, 6) and an output converter (7) which converts the amplified electrical signals back into acoustic signals, or in the case of an implanted hearing aid into mechanical signals, and a feedback digital finite impulse response filter (FIR filter) (9) for compensation of unwanted feedback (8) from the output converter to the microphone, in which the filter coefficients of filter (9) are determined by feeding a short pulse into the feedback signal path (5, 6, 7, 1, 2) and directly measuring the impulse response of this signal path.

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

Abstract: A digital hearing aid (10) is provided that includes a microphone (12), a control and modeling circuitry (18), and a receiver (20). The microphone (12) receives an input sound signal x(t) and generates a digital input signal x(n) in response. The control and modeling circuitry (18) filters and amplifies the digital input signal x(n) and performs feedback neutralization and feedback path modeling to generate a digital output signal y(n). The receiver (20) receives the digital output signal y(n) and generates an output sound signal y(t) in response.

Abstract: Feedback cancellation apparatus uses a cascade of two filters along with a short bulk delay. The first filter is adapted when the hearing aid is turned on in the ear. This filter adapts quickly using a white noise probe signal, and then the filter coefficients are frozen. The first filter models parts of the hearing-aid feedback path that are essentially constant over the course of the day. The second filter adapts while the hearing aid is in use and does not use a separate probe signal. This filter provides a rapid correction to the feedback path model when the hearing aid goes unstable, and more slowly tracks perturbations in the feedback path that occur in daily use. The delay shifts the filter response to make the most effective use of the limited number of filter coefficients.

Abstract: The process for controlling a programmable or program-controllable hearing aid for in-situ adjustment of said hearing aid to an optimum target gain in one or more frequency bands by establishing the hearing threshold level of the wearer for one or more frequency bands, determining the target input/output response for the detected hearing loss and generating a corresponding parameter set for an ideal input/output response for the detected hearing loss under feedback-free conditions, by setting the control parameter set of a signal processor initially to an input/output response with a gain equal to the maximum target gain, operating the hearing aid in-situ in accordance with said initial input/output response while monitoring said hearing aid for the occurence of any acoustic feedback, and if no noticeable feedback is detected setting said initial parameter set for said input/output response into said hearing aid, and if noticeable acoustic feedback is detected reducing the gain over at least one of said frequ

Abstract: A hearing aid includes a microphone (1), a signal-transmission unit (2, 3,) for forming or otherwise processing the signal, an output amplifier (4) to which an earphone (10) is connected, and a battery as the power supply. The output amplifier (4) is implemented essentially as a .SIGMA.-.DELTA. amplifier and is connected to a pulse generator (8) which produces a high-frequency pulsed clock signal in the 1 MHz region. A series-connected low-pass filter (15) is also provided. The input signal to the signal converter is a representation, produced by signal processing in the transmission unit, of the low-frequency input signal to the hearing aid, this signal being converted in the signal converter into a binary signal. The output signal (14) thus appears, after passing through the low-pass filter, essentially as an amplified copy of the low-frequency input signal.