Abstract: Disclosed are devices, systems, apparatus, methods, products, and other implementations, including a method comprising obtaining, by a device, a combined sound signal for signals combined from multiple sound sources in an area in which a person is located, and applying, by the device, speech-separation processing (e.g., deep attractor network (DAN) processing, online DAN processing, LSTM-TasNet processing, Conv-TasNet processing), to the combined sound signal from the multiple sound sources to derive a plurality of separated signals that each contains signals corresponding to different groups of the multiple sound sources. The method further includes obtaining, by the device, neural signals for the person, the neural signals being indicative of one or more of the multiple sound sources the person is attentive to, and selecting one of the plurality of separated signals based on the obtained neural signals. The selected signal may then be processed (amplified, attenuated).

Abstract: An electronic device configured with a microphone, a voice interaction wake-up method executed by an electronic device equipped with a microphone, and a computer-readable medium, the electronic device comprising a memory and a central processing unit, wherein the memory stores computer-executable instructions, and when executed by the central processing unit, the computer-executable instructions perform the following operations: analyzing a sound signal collected by a microphone, identifying whether the sound signal contains speech spoken by a person and whether it contains wind noise sounds generated by airflows hitting the microphone as a result of the speech spoken by the person, and in response to determining that the sound signal contains sound spoken by the person and contains wind noise sounds generated by airflows hitting the microphone as a result of the speech spoken by the user, processing the sound signal as speech input by the user.

Abstract: Various implementations include a computational architecture for a personal active noise reduction (ANR) device. The device includes a communication interface that receives an audio stream, a driver, a microphone system and an ANR processing platform. The platform includes a first DSP configured to: receive the audio stream and signals from the microphone system, perform ANR on the audio stream according to a set of parameters in the first DSP, and output a processed audio stream. The platform includes a second DSP configured to: generate state data in response to an analysis of the source audio stream, signals from the microphone system, and the processed audio stream; and alter the operational parameters on the first DSP. The platform includes a general purpose processor configured to characterize a malfunction in response to instability or error condition events detected by the second DSP.

Abstract: The present disclosure relates to a pair of smart glasses. The pair of smart glasses may include a frame, one or more lenses, and one or more temples. The pair of smart glasses may further include at least one microphone array, a noise reduction device, at least one low-frequency acoustic driver, at least one high-frequency acoustic driver, and a controller. The at least one microphone array may be configured to generate sub-band voice signals. The noise reduction device may be configured to generate a noise correction signal based on each of the sub-band voice signal and generate a sub-band correction signal based on the noise correction signal and the sub-band voice signal. The controller may be configured to direct the low-frequency acoustic driver to output the sounds in a first frequency range and direct the high-frequency acoustic driver to output the sounds in a second frequency range.

Abstract: Disclosed are devices, systems, apparatus, methods, products, and other implementations, including a method comprising obtaining, by a device, a combined sound signal for signals combined from multiple sound sources in an area in which a person is located, and applying, by the device, speech-separation processing (e.g., deep attractor network (DAN) processing, online DAN processing, LSTM-TasNet processing, Conv-TasNet processing), to the combined sound signal from the multiple sound sources to derive a plurality of separated signals that each contains signals corresponding to different groups of the multiple sound sources. The method further includes obtaining, by the device, neural signals for the person, the neural signals being indicative of one or more of the multiple sound sources the person is attentive to, and selecting one of the plurality of separated signals based on the obtained neural signals. The selected signal may then be processed (amplified, attenuated).

Abstract: Disclosed herein are systems and methods for eardrum acoustic pressure estimation for hearing device applications. A receiver signal is produced through a receiver of a hearing device, or a physical extension of the receiver, placed inside an ear canal of a user. A microphone signal is sensed, in response to the receiver signal, using a microphone placed outside the ear canal. Based on the receiver signal and the microphone signal, a reliability signal of the microphone signal and a feedback-path signal are computed. A spectral representation signal of the feedback-path signal is computed, and a real-ear response signal is computed using a mathematical combination of the spectral-representation signal and the reliability signal. The computed real-ear response signal is used to modify an operational characteristic of the hearing device, or to make a recommendation for modification of an operational characteristic or a physical characteristic of the hearing device.

Abstract: An in-ear device occludes an ear canal of an ear of a user. The in-ear device is configured to be calibrated such that the user perceives audio content as though the in-ear device is not occluding the ear canal. A transducer of the in-ear device presents audio content, and an inner microphone of the in-ear device detects sound pressure data within the ear canal. A controller of the in-ear device determines a blocked sound pressure at the entrance to the ear canal based on sound pressure data from an outer microphone. The controller generates sound filters custom to the user based in part on the detected sound pressure within the ear canal and the blocked sound pressure at the entrance to the ear canal. The controller adjusts audio content using the sound filter, and the transducer presents the adjusted audio content to the user.

Abstract: Various implementations include a method for implementing a computational architecture for a personal active noise reduction (ANR) device. A method includes receiving a source audio stream with a first DSP and performing ANR on the source audio stream utilizing operational parameters stored in the first DSP; outputting a processed audio stream from the first DSP; generating state data with a second DSP in response to an analysis of at least one of the source audio stream, microphone inputs and the processed audio stream, and communicating signals to the first DSP over a common bus coupled to the first and second DSPs to alter the operational parameters in the first DSP; and utilizing a general purpose processor coupled to both the first DSP and the second DSP to communicate control signals with a communication interface, process state data from the second DSP, and alter the operational parameters in the first DSP.

Abstract: A method and device for processing information are provided. The method for processing information includes: determining a current output voltage of a power supply in a terminal device; determining a target parameter of an audio processing circuit in the terminal device according to the current output voltage; and configuring the audio processing circuit for processing an audio signal according to the target parameter. With the embodiments of the present disclosure, an impact of a change of the output voltage of the power supply on output volume of a loudspeaker can be reduced, thereby improving user experience.

Abstract: A binaural beamformer comprising two beamforming filters may be communicatively coupled to a microphone array to generates two beamforming outputs, one for the left ear and the other for the right ear. The beamforming filters may be configured in such a way that they are orthogonal to each other to make white noise components in the binaural outputs substantially uncorrelated and desired signal components in the binaural outputs highly correlated. As a result, the human auditory system may better separate the desired signal from white noise and intelligibility of the desired signal may be improved.

Abstract: Presented herein are techniques for enhancing a hearing prosthesis recipient's perception of multiple frequencies present in received sound signals. The hearing prosthesis is configured to extract a plurality of frequencies from the received sound signals and to use the plurality of frequencies to modulate the amplitudes of different stimulation pulse sequences that are to be delivered to the recipient via different stimulation channels. The hearing prosthesis may also adapt a stimulation resolution of the stimulation pulse sequences when delivering the modulated stimulation pulses sequences to the recipient.

Abstract: Various implementations include a computational architecture for a personal active noise reduction (ANR) device. The device includes a communication interface that receives an audio stream, a driver, a microphone system and an ANR processing platform. The platform includes a first DSP configured to: receive the audio stream and signals from the microphone system, perform ANR on the audio stream according to a set of parameters in the first DSP, and output a processed audio stream. The platform includes a second DSP configured to: generate state data in response to an analysis of the source audio stream, signals from the microphone system, and the processed audio stream; and alter the operational parameters on the first DSP. The platform includes a general purpose processor configured to: communicate control signals with the communication interface, process state data from the second DSP, and alter the parameters on the first DSP.

Abstract: Method for estimating a pure tone hearing threshold or a pure tone audiogram of a person with a non-calibrated audio-system, comprising the steps of performing a supra-threshold test at least over a portion of the audible frequency spectrum, wherein the audible frequency spectrum ranges particularly from 16 Hz to 20.

Abstract: A filter for a microphone system of a miniature electronic device. The filter includes a sound inlet and a sound outlet, wherein the filter is adapted to be mounted on a surface of a printed circuit board.

Abstract: A system that incorporates teachings of the present disclosure may include, for example, a computing device having a controller to modify sound produced by a gaming application according to an otological profile of a user. Additional embodiments are disclosed.

Abstract: A method for operating a hearing apparatus system with at least one hearing apparatus uses a first sensor of the hearing apparatus to capture a first measured variable characteristic of an actual wearing situation. If the first measured variable and/or a variation over time derived therefrom satisfies at least one prescribed first criterion, it is inferred that the hearing apparatus has had a tumble. If the tumble is detected to have taken place, at least one further measured variable and/or a variation over time derived therefrom is captured, and the at least one further measured variable is used to ascertain whether the hearing apparatus is arranged on the head of a hearing device wearer after the tumble.

Abstract: A system to optically measure an ear includes a controller with logic that when executed by the controller causes the system to perform operations. Operations may include capturing the one or more images of the ear using the one or more image sensors, and generating image data from the one or more images. 3D keypoints of the ear are calculated from the image data, and a 3D model of the ear is generated using the 3D keypoints.

Abstract: Earpieces and methods for acute sound detection and reproduction are provided. A method can include measuring an external ambient sound level (xASL), monitoring a change in the xASL for detecting an acute sound, estimating a proximity of the acute sound, and upon detecting the acute sound and its proximity, reproducing the acute sound within an ear canal, where the ear canal is at least partially occluded by an earpiece. Other embodiments are disclosed.

Abstract: The present application discloses am electrical stimulation device for electrically stimulating at least one target zone of an organism. The electrical stimulation device comprises a control unit and an electrical stimulation unit. The electrical stimulation unit includes a frequency synthesizer, an amplifier, a variable resistor, at least one first electrode and at least one second electrode. The frequency synthesizer is coupled to the control unit and generates a frequency signal. The amplifier is coupled to the frequency synthesizer. The variable resistor comprises a resistance and is coupled to the control unit and the amplifier. The first electrode and the second electrode are coupled to the amplifier. The amplifier outputs an electrical stimulation signal according to the frequency signal of the frequency synthesizer and the resistance of the variable resistor to impel the first electrode and the second electrode to generate an electric field.

Abstract: Switching circuits controllable to force an input into a circuit and to sense a responsively produced output in multiple ways to produce different combinations of positive and negative polarities of a desired signal and of sources of offsets and 1/f noise. The switching circuits are controlled in a non-ordered time sequence of different combinations of positive and negative polarities of the sources of the offsets and 1/f noise that spreads their energy to a frequency range above the desired signal frequency band. The non-ordered time sequence leaves the polarity of the desired signal unchanged. Uncorrelated delta-sigma modulators may generate the control signal. A DSP processes a resulting spectrum of a digital domain version of the sensed output to measure residual offsets and 1/f noise and adds to an input present at the DSMs a signal equal in magnitude and opposite in sign to the measured residual offsets and 1/f noise.

Abstract: Cochlear implant systems can include a cochlear electrode, a stimulator in electrical communication with the cochlear electrode, and a signal processor in communication with the stimulator. The signal processor can include circuitry and a can surrounding and housing the circuitry. The signal processor can further include a first impedance between the circuitry and the can in order to reduce unintended electrical communication between the cochlear electrode and the circuitry of the signal processor. Other components can similarly include impedance between their respective cans and circuitry to prevent such undesired communication. Components can communicate with one another via bidirectional communication. Communication can include communication of signals and inverted signals to reduce net charge flow through the wearer's body.

Abstract: A hearing assistance system obtains a first input audio signal that is based on sound received by a first set of microphones. The system also obtains a second input audio signal that is based on sound received by a second, different set of microphones. A first adaptive beamformer generates a first output audio signal based on the first input audio signal, the second input audio signal, and a value of a first parameter. A second adaptive beamformer generates a second output audio signal based on the first input audio signal, the second input audio signal, and a value of a second parameter. The value of the first parameter and the value of the second parameter are determined such that a magnitude squared coherence (MSC) of the first output audio signal and the second output audio signal is less than or equal to the coherence threshold.

Abstract: A sound field device is disclosed that comprises an elevation cue estimator, a low-frequency filter estimator, and a high-frequency filter estimator. The elevation cue-estimator is configured to estimate an elevation cue of a head-related transfer function (HRTF) of at least one listener. The low-frequency filter estimator is configured to estimate one or more low-frequency filter elements based on the elevation cue. The high-frequency filter estimator is configured to estimate one or more high-frequency filter elements based on the elevation cue. An estimation method of the low-frequency filter estimator is different from an estimation method of the high-frequency filter estimator. The one or more low-frequency filter elements and the one or more high-frequency filter elements are for driving an array of loudspeakers to generate an elevated sound impression at a bright zone.

Abstract: According to an embodiment, a method for producing stimulation pulses in a bilateral cochlear implant (CI) is disclosed. The method includes receiving a sound at a first microphone or a first microphone array positioned at or in the vicinity of a first ear of a user of the bilateral CI and receiving the sound at a second microphone or a second microphone array positioned at or in the vicinity of a second ear of the user of the bilateral CI. Furthermore, generating, using the first microphone or first microphone array, a first microphone signal in response to the sound received at the first microphone or first microphone array and generating, using the second microphone or second microphone array, a second microphone signal in response to the sound received at the second microphone or second microphone array.

Abstract: An approach to audio processing aims to improve intelligibility by amplifying time segments of an input signal when the level of the signal falls below a long-term average level of the input signal, for instance, introducing a time-varying gain such that the signal level of the amplified segment matches the long-term average level.

Abstract: Application of a hearing aid alters the distribution of sound pressure at the tympanic membrane in the ear. Measurements of sound pressure of a free ear and with a hearing aid are performed to evaluate the alteration produced and the in situ frequency response of the hearing aid itself. The following steps are performed: calculating the REUG; calculating a magnitude; calculating one of two components and designing of the filter for the compensation for such component; calculating the REAG; calculating one of two components. The sum of two components is the Insertion Gain; designing the compensating filter of the POST component of the Insertion Gain; testing and verifying, on the subject wearing the hearing aid, the compensating filter of REIG, and possible refinement procedure to correct the compensation of the Insertion Gain obtained in the previous step; tuning the hearing aid for deafness taking into account the compensating filter.

Abstract: A mobile communication device (50) receives an audio stream as input and delivers a processed audio stream as output. The mobile communication device has a data connection providing access to the Internet, and a short range data connection for delivering a processed audio stream as output to a specific hearing aid (10). The mobile communication device acquires a data set containing hearing aid settings for said specific hearing aid from a remote server (71), and adjusts the emulation software application by means of the data set containing hearing aid settings for said specific hearing aid (10). The mobile communication device transmits the control signals and a processed audio stream to the specific hearing aid via said short range data connection and the specific hearing aid outputs the audio signal to the user without additional amplification. The invention also provides a method of signal processing in a mobile communication device.

Abstract: An input sound signal is processed to generate band pass signals that each represent an associated band of audio frequencies. Stimulation timing signals are generated for each band pass signal, including for one or more selected band pass signals using a timing function defined to determine a mapped instantaneous frequency by adjusting instantaneous frequency based on a frequency relation factor representing a relationship between: (1) the range of audio frequencies for the band pass channel, and (2) a patient-specific, channel-specific range of stimulation rates defined based on patient-specific pitch perception measurements to enhance patient pitch perception within the range of audio frequencies for the band pass channel, and iii. generate the stimulation timing signal for the selected band pass signal at the mapped instantaneous frequency of the selected band pass signal.

Abstract: A signal processing arrangement is described for signal processing in a bilateral hearing implant system having left side and right side hearing implants. An interaural coherence analysis module analyzes system input signals to produce an interaural coherence signal output characterizing reverberation-related similarity of the input signals.

Abstract: An arrangement is described for speech signal processing. An input microphone signal is received that includes a speech signal component and a noise component. The microphone signal is transformed into a frequency domain set of short-term spectra signals. Then speech formant components within the spectra signals are estimated based on detecting regions of high energy density in the spectra signals. One or more dynamically adjusted gain factors are applied to the spectra signals to enhance the speech formant components.

Abstract: A signal processing system is described for a bilateral hearing implant system having left side and right side hearing implants. An interaural coherence analysis module receives input signals from each hearing implant including sensing microphone signals and band pass signals, and analyzes the input signals to produce an interaural coherence signal output characterizing reverberation-related similarity of the input signals. A pulse timing and coding module for each hearing implant then processes the band pass signals to develop stimulation timing signals, wherein for one or more selected band pass signals, wherein the processing includes using an envelope gating function developed from the interaural coherence signal.

Abstract: A method performs signal processing in a binaural hearing device that has first and second hearing aids with first and second microphones producing first and second signals and with first and second sound generators. The first and second signals ascertain a direction of a main sound source. A deviation in the direction from a frontal direction prompts the hearing aid that is closer to the main sound source to be defined as the local hearing aid and the hearing aid that is more remote from the main sound source to be a remote hearing aid. The local hearing aid, in one frequency band, filters the first signal using an angle-dependent first filter factor, and thus produces a first filtered signal. The first signal, the second signal and/or the direction of the main sound source is used for determining an adaptation coefficient, a first adapted signal and a local directional characteristic.

Abstract: A method, and a corresponding apparatus, for processing an input signal comprise filtering the input signal to separate a passband frequency component of the input signal from a stopband frequency component of the input signal, and adjusting relative signal power values of the passband frequency component and the stopband frequency component of the input signal based at least in part on signal values of a number of samples associated with the input signal. In the case of audio signals, for example, such processing is used for spectral expansion of the input signal by enhancing the power of the stopband, or low and high frequencies, component with respect to the power of the passband component of the input signal. As a result, a better audio quality is achieved.

Abstract: A device for a hearing impaired person being subjected to a tinnitus is disclosed. The device includes an input transducer for providing an electric input signal comprising audio, a controllable filter for filtering the electric input signal and configured to output a filtered electric input signal such that signal energy of the electric input signal is reduced in at least a part of a frequency range, a signal processor, connected to the controllable filter, for processing the filtered electrical input signal according to a processing algorithm and to output a processed electric signal, an output transducer, connected to the signal processor, for converting the processed electric signal to an acoustic output signal to be presented to the hearing impaired person wearing the listening device, and an activator, coupled to the controllable filter, for activating and deactivating the controllable filter in dependence on a detection signal and a timer signal.

Abstract: The invention relates to a speaker enclosure comprising at least two channels, wherein a first and second channel are respectively dedicated to separate and adjacent first and second frequency bands, each of said channels including a filtering stage supplied by the control signal and a speaker unit, wherein said speaker enclosure is characterized in that said channels also include means for simultaneously: disconnecting the first channel from the control signal; and modifying the filtering stage of the second channel so as to modify the frequency response thereof.

Abstract: A hearing aid includes at least one electrode located at a surface of a housing of the hearing aid to allow the electrodes to contact the skin of a user during use of the hearing aid, at least one electrode being adapted to pick up a low voltage signal from the user's brain, an amplifier unit operationally connected to the electrode(s) and adapted for amplifying the low voltage signal(s) to provide amplified brain signal(s), and a signal processing unit adapted to process the amplified brain signal(s) to provide a processed brain signal as well as to apply a time and frequency dependent gain to an input audio signal and to provide a processed audio output signal. Also a method of operating a hearing aid and a hearing aid system.

Abstract: Technology presented herein increases a user's enjoyment of sound by personalizing an audio signal so that the user perceives the audio signal as if the user had ideal hearing and/or desired hearing. In one embodiment, headphones on a user's head include a sensor and a speaker. While the speaker plays an audio signal to the user, the sensor records the user's response to the audio signal. The sensor can be a microphone, a brainwave sensor, an EEG sensor, etc. The user's response can be the audio response inside the user's ear, the brainwave response associated with the user, electrical skin response associated with the user, etc. Based on the measured response, and based on the knowledge of how other people perceive sound, the audio signal is modified to compensate for the difference between the user's hearing and the ideal hearing and/or desired hearing, therefore increasing the user's enjoyment of sound.

Abstract: Broadly speaking, the technology relates to using wave field synthesis theory to simulate one or more idealized virtual point sources in a multi-speaker system. The speaker transfer function of each speaker is modeled, and the values and directional gradient of the combined speaker transfer function at test points in a convexly-bounded listening region are compared to the desired values and directional gradient for the idealized transfer function of the idealized virtual point source(s) at the test points to determine filter coefficient sets for each filter. The determined filter coefficients are those which minimize the total difference between the values and directional gradient of the combined speaker transfer function and the values and directional gradient of the idealized transfer function of the idealized virtual point source across all the test points for a plurality of frequency bins.

Abstract: Mechanisms for obtaining an audio signal and information relating to the recording conditions of the audio signal are provided, the method comprising: obtaining an audio signal, wherein the audio signal is a voltage signal representation of sound, the voltage signal having been converted from a pressure signal; obtaining first information indicative of at least one objective feature and/or at least one perceptual feature associated with the conversion of the pressure signal into the audio signal in the form of a voltage signal representation; storing the audio signal, the first information, and second information identifying a relationship between the audio signal and the first information; determining an error adjustment for adjusting the audio signal based on the obtained information; applying the error adjustment to the audio signal to create an error-adjusted audio signal; and obtaining an audio signal and information relating to the recording conditions of the audio signal.

Abstract: A system for updating a hearing aid by providing an update to a hearing aid to configure the hearing aid for an acoustic environment with a sound profile different than a physical environment a user is currently located in with an acoustic sample representative of the acoustic environment.

Abstract: Methods, systems, and apparatus, including computer programs encoded on computer storage media, for training a neural network.

Abstract: A listening device for a hearing impaired person being subjected to a tinnitus at a tinnitus frequency range including a tinnitus frequency is disclosed. The device includes an input transducer for providing an electric input signal comprising audio and a controllable filter for filtering the electric input signal received from the input transducer. The filter outputs a filtered electric input signal such that signal energy of the electric input signal immediately surrounding the tinnitus frequency remain substantially unchanged and signal energy of the electric input signal at a distance to the tinnitus frequency is substantially reduced. The device further includes a processor connected downstream of the filter and processes the filtered electrical input signal and outputs a processed electric signal, and an output transducer connected downstream of the processor and converts the processed electric signal to an acoustic output signal to be presented to a hearing impaired person.

Abstract: An apparatus comprising: a transducer suitable for generating an acoustic sound pressure wave being configured to: magnetically couple the apparatus to a further apparatus transducer for transferring data.

Abstract: A method is described for generating electrical stimulation signals for stimulation contacts in an auditory implant system. Characteristic feature periods are determined for frequency components in an input audio signal. For selected feature periods that meet a period selection criteria, adjusted feature periods are determined that correspond to a nearest integer multiple of a language-specific fundamental period. A corresponding stimulation rate frequency is determined for each adjusted feature period, and each stimulation rate frequency is assigned to one or more stimulation contacts. The stimulation signals are then generated for the stimulation contacts at their respective stimulation rate frequencies.

Abstract: A system, methods, and devices that allow a listener to compensate for variability in the physical configuration and characteristics of individual ears to optimize the sound quality and listening experience for sound reproduction or amplification through an earphone, earbud, hearing aid, or the like. The present invention comprises one or more earphones, earbuds, hearing aid, dongle, or similar devices with an equalizer or equalizing circuitry or processing program contained therein, or in a circuit with the earphone, earbud, hearing aid or similar device. The equalizer or equalizing circuitry or processing program may be located in a separate device attached or in electronic communication, wired or wirelessly, to the earphone, earbuds or hearing aid, such as a remote. It also may be located in the sound-providing or music-providing device. The equalizer settings can be adjusted manually or automatically for each ear. An amplifier, or amplifying circuitry, also may be present.

Abstract: A hearing device, an acoustic apparatus and a sound processing method thereof are provided. The method includes detecting audio signals, determining whether the audio signals include a voice signal of a user, and controlling amplification of the audio signals according to a result of the determination.

Abstract: A method of treating tinnitus comprising measuring a patient's hearing, determining the patient's hearing loss and the patient's tinnitus frequency using the measurements of the patient's hearing, programming a clinical controller with the measurements of the patient's hearing, selecting a plurality of therapeutic tones, where the therapeutic tones are selected to be at least a half-octave above or below of the patient's tinnitus frequency, setting an appropriate volume for each of the plurality of tones, repetitively playing each of the plurality of therapeutic tones, and pairing a vagus nerve stimulation pulse train with each playing of a therapeutic tone, thereby reducing the patient's perception of tinnitus.

Abstract: A computer tool is presented. The computer tool has, for example, a working memory to receive input data, an auxiliary memory to store several approximation dictionaries, a reducer arranged for the iterative calculation of the approximation data of the input data. The successive iterations are of the “matching pursuit” type. The computer tool also has, for example, a selector arranged to choose an approximation dictionary in the auxiliary memory based on a selection ruled based on at least some of the preceding iterations, in which the reducer calls the selector in each iteration, and uses the dictionary thus selected in the adaptive matching.

Abstract: A hearing aid (60A) configured to be worn by a hearing-impaired user has a speech detector (10A) and a speech enhancer (40A) for enhancing speech being present in an input signal of the hearing aid (60A). The speech detector (10A) has means (11, 12) for independently detecting the presence of voiced and unvoiced speech in order to allow for the speech enhancer (40A) to increase the gain of speech signals suitably fast to incorporate the speech signals themselves. The hearing aid (60A) has means (49A, 50A) for communicating information regarding the detected speech signals wirelessly to a similar hearing aid (60B) worn contralaterally by the user for the purpose of mutually enhancing speech signals in the two hearing aids (60A, 60B) when speech is detected to be originating from the front of the user, and means (52B) for suppressing speech enhancement in the contralateral hearing aid (60B) when speech is detected to be originating from the ipse-lateral side of the user.

Abstract: A device includes a sound input element for receiving a sound signal and converting the sound signal into an electrical signal. The device also includes an actuator and a sound processor for processing the electrical signal to generate a stimulation signal. The sound processor is configured to apply frequency shifting on the stimulation signal to generate a frequency shifted stimulation signal and to apply the frequency shifted stimulation signal as an output from the device. The frequency shifting can be triggered by detecting that the stimulation signal is associated with frequencies above an output limit of the device for a recipient and/or the stimulation signal can be applied as vibration.