Abstract: One embodiment of the present disclosure sets forth a technique for delivering audio content to a user. The technique includes analyzing audio content to determine a frequency at which to vibrate an effector. The technique further includes configuring one or more haptic output devices to generate haptic output that generates a haptic trap proximate to the effector. The technique further includes configuring the one or more haptic output devices to modify the haptic output to cause the haptic trap to vibrate the effector at the determined frequency. The effector vibrates to transmit the audio content. In one embodiment, the effector vibrates a skull of a user.

Abstract: A method for detecting ambient noise to change the playing voice frequency and a sound playing device thereof are disclosed. The method includes the following steps: obtaining an input voice; detecting an ambient noise, and analyzing a noise frequency range of the ambient noise; determining whether a consonant frequency range of the input voice falls within the noise frequency range; if yes, adjusting the frequency of the consonant of the input voice to avoid the noise frequency range to form a modified voice; and playing an output voice, wherein the output voice includes the modified voice.

Abstract: A method of operating a hearing aid system comprising a hearing aid (300) wherein a digital filter (302) is adapted to be of minimum phase and to provide a frequency dependent target gain.

Abstract: A hearing system for stimulating an auditory system for sound perception by activating a particular region of a cochlea of a user using ultrasound signals, the particular region corresponding to a target frequency range, the system including: an ultrasonic transducer configured to deliver an ultrasound signal via an interface medium; and a processor communicatively coupled to the ultrasonic transducer, the processor to: obtain an audio signal, extract at least one of a temporal feature or a spectral feature from the audio signal, transpose the audio signal to the target frequency range based on extracting the at least one of the temporal feature or the spectral feature from the audio signal, generate a modulated ultrasound signal based on modifying a carrier signal having at least one frequency between 100 kHz and 4 MHz by the transposed audio signal, and provide the modulated ultrasound signal to the ultrasonic transducer for delivery via an interface medium.

Abstract: The present invention provides a method implementing a speech strategy based on zero crossing behavior of speech time waveforms; the zero crossing containing both spectral and temporal speech information. This method uses temporal information of speech to activate electrodes instead of spectral information; maps temporal segment durations to spatial durations along the basilar membrane inside the cochlea; and provides instantaneous, continuous information about speech to electrodes that stimulate the auditory nerve. Timing of oval window mechanical motion is represented by zero crossings which are used to activate electrodes implanted inside the cochlea. Motion of the tympanic membrane, and the oval and round windows, follow the speech signal temporal waveform. Positive segments of the temporal waveform cause inward displacement of the oval membrane from its stationary position and negative segments causes outward retraction of the membranes.

Abstract: A hearing aid system (400) including a hearing aid (412) and an external device (413), wherein the hearing aid (412) includes a network adapted to provide a desired processed output, wherein a multitude of weights determines the processing of the network and wherein the values of the multitude of weights are determined by the external device (413).

Abstract: The invention discloses an algorithm-based audio optimization method, an intelligent terminal and a storage device. The method includes steps of converting an original audio file on time domain into an audio file on frequency domain through Fourier transform; before extracting and matching frequency range and amplitude information of an audio signal with those of different types of existing audio test standard sound sources to determine type of the audio signal; matching corresponding frequency mapping function by function library for transform to obtain a processed audio file, obtaining an optimized audio file by inverse Fourier transform. The present invention searches a similar type of sound source by comparing and matching original audio with standard sound sources, determines the frequency mapping function for transform, maps the frequency, compresses or expands relevant frequency, and automatically performs tuning to achieve effect of improving sound quality.

Abstract: In one embodiment, an audio system can replace a portion of an audio signal within a first range of frequencies, with an amplitude modulated noise signal comprising frequencies within the first range of frequencies and having a volume envelope corresponding to a volume envelope of the audio signal within a second range of frequencies.

Abstract: A hearing system for stimulating an auditory system for sound perception by activating a particular region of a cochlea of a user using ultrasound signals, the particular region corresponding to a target frequency range, the system including: an ultrasonic transducer configured to deliver an ultrasound signal via an interface medium; and a processor communicatively coupled to the ultrasonic transducer, the processor to: obtain an audio signal, extract at least one of a temporal feature or a spectral feature from the audio signal, transpose the audio signal to the target frequency range based on extracting the at least one of the temporal feature or the spectral feature from the audio signal, generate a modulated ultrasound signal based on modifying a carrier signal having at least one frequency between 100 kHz and 4 MHz by the transposed audio signal, and provide the modulated ultrasound signal to the ultrasonic transducer for delivery via an interface medium.

Abstract: Disclosed are technologies for processing a surround sound audio signal. The center channel is processed to generate a processed center channel. For each pair of corresponding left and right audio channels, mid and side channel signals are generated by taking the sum and difference of the pair, respectively. The side channel is outputted to a first signal pathway that features a dynamic range compressor. The mid channel is outputted to a second signal pathway and spectrally decomposed into a plurality of sub-band signals. Each sub-band signal is compressed by a dynamic range compressor and transmitted to a gain stage. The compressed sub-band signals are outputted from the gain stage and recombined. The outputs of the first and second signal pathways are then recombined and decoded to produce processed left and right audio signals. Each pair or processed left and right audio signals are output along with the processed center channel.

Abstract: In one embodiment, an audio system can generate a parametrically formulated noise signal which can be mixed with an audio signal or acoustic signal. According to an embodiment, a parametrically formulated noise signal can be configured to have a power spectrum amplitude that is a function of frequency. According to an embodiment, a parametrically formulated noise signal can have a power spectrum amplitude across a range of frequencies that is a function of an individual's hearing thresholds across the range of frequencies.

Abstract: In one embodiment, an audio system can replace a portion of an audio signal within a first range of frequencies, with an amplitude modulated noise signal comprising frequencies within the first range of frequencies and having a volume envelope corresponding to a volume envelope of the audio signal within a second range of frequencies.

Abstract: According to an embodiment, an auditory device for generating an electrode stimulation signal is disclosed. The device includes a filter bank configured to filter an incoming audio signal into a plurality of band pass limited audio signals, each band pass limited audio signal relating to an audio frequency band. The device further includes a processor configured to extract an onset time for firing an electrode in accordance with rate of change of a band pass limited audio signal and/or a derived band pass limited audio signal satisfying a predetermined criterion, and a signal generator for generating the electrode stimulation signal for the electrode corresponding to the band pass limited audio signal and/or derived band pass limited audio signal in accordance with the determined onset time. The device includes a plurality of implantable stimulation electrodes configured to deliver the electrode stimulation signal to a region of cochlea.

Abstract: Synthetically generated signals are reproduced via a binaural hearing system, which includes two hearing devices. After reproducing a first signal via the first hearing device a reproduction of a second signal via the second hearing device is also carried out delayed by a defined time interval. The additional time delay is to be quantified such that the impression of a sound amplification is produced with a hearing system wearer.

Abstract: An electronic device and a sound signal processing method for improving sound perception of a hearing-impaired user are provided. The electronic device of the present disclosure includes a sound input unit comprising sound input circuitry configured to detect a sound and to convert the sound into a first sound signal and a processor which is electrically connected to the sound input unit, the processor configured to receive the first sound signal and to perform a predetermined signal processing on the first sound signal to generate a second sound signal, wherein the signal processing includes detecting a frequency band with a level equal to or greater than a predetermined value in a first frequency band above a predetermined cutoff frequency of the first sound signal, generating harmonic signals including a plurality of frequency bins that are identical in level with a signal in the detected frequency band, and overlapping the harmonic signals with the first sound signal.

Abstract: A hearing prosthesis, the hearing prosthesis including a plurality of sound capture devices and a determinator configured to generate a parameter indicative of an orientation of the plurality of sound capture devices relative to a reference, wherein the hearing prosthesis is configured to adjust a direction of focus of the hearing prosthesis based on at least the parameter.

Abstract: The application relates to a hearing aid system comprising a hearing aid device comprising a configurable signal processing unit, a programming device configured to program the configurable signal processing unit, and a communication link allowing the exchange of data between the hearing aid device and the programming device. The application further relates to a method of programming configurable signal processing unit to provide an increased speech intelligibility by frequency lowering. The hearing aid system is configured to program the configurable signal processing unit of the hearing device according the claimed method. The invention may e.g. be used in hearing aids.

Abstract: In various embodiments, a circuit arrangement is provided. The circuit arrangement includes a sensor set up to provide an analogue signal, an analogue/digital converter set up to receive the analogue signal and to provide a first signal, and a first filter set up to receive a signal based on the first signal and to provide a second signal. The first filter is set up in such a manner that the second signal is allowed through without amplification or substantially without amplification in a frequency range of approximately 20 Hz to approximately 10 kHz, and the second signal has a gain of greater than 0 dB at least above a predefined frequency which is greater than approximately 20 kHz.

Abstract: In general, in one aspect, a method performed by one or more processes executing on a computer systems includes receiving an audio signal comprising a range of audio frequencies including high frequencies and low frequencies, converting a first portion of the range of audio frequencies into haptic data, shifting a second portion of the range of audio frequencies down to lower audio frequencies, and presenting at least one of the converted first portion and the shifted second portion to a human user. Other implementations of this aspect include corresponding systems, apparatus, and computer program products.

Abstract: A method includes programming a sound processor to apply frequency shifting on a stimulation signal to generate a frequency shifted stimulation signal. The frequency shifting depends on one or more of a decibel level of a received sound signal, a hearing loss level associated with generating the stimulation signal, attenuation of an output based on the frequency shifted stimulation signal, or operating a hearing prosthesis in a single sided mode or a bilateral mode. The method includes receiving a sound signal, generating the stimulation signal from the sound signal, applying the frequency shifting to the stimulation signal to generate the frequency shifted stimulation signal, and generating, by an actuator of the hearing prosthesis, the output based on the frequency shifted stimulation signal, wherein the output is configured to be perceived as sound.

Abstract: Methods, systems, and devices for determining a stimulus based on a masking model of human hearing are disclosed. An example method includes receiving a signal that includes information indicative of one or more spectral components of a sound. The example method also includes determining a masking curve that includes information indicative of one or more masking spectral components. The masking curve is determined according to a masking model of human hearing that accounts for at least an effect due to temporal masking. The example method further includes generating a stimulus based on a difference between the one or more spectral components and the one or more masking spectral components. The stimulation allows a recipient to perceive at least a portion of the sound.

Abstract: Disclosed herein, among other things, are methods and apparatuses for hearing assistance low profile offset spouts. One aspect of the present subject matter relates to a hearing assistance spout consisting of a one-piece spout with a curved path to an off-axis (e.g., offset) port. The spout may be manufactured using metal injection molding to provide for a reflow solder attachment to the hearing assistance microphone. One aspect of the present subject matter relates to placing a full-sized spout off-axis while maintaining an open state that does not restrict the acoustic path, which may allow the offset spout to include a radial seal independent of the microphone body itself. One aspect of the present subject matter relates to an offset spout including a sealed acoustic barrier wall to protect the acoustic inlet of the microphone from debris or moisture.

Abstract: A method of processing a signal in a hearing aid (50), the method comprising the steps of determining a gain vector applied in a number of individual frequency bands, determining a gradient of a speech intelligibility measure as a function of the gain vector, modifying the value of the gradient and using the modified value of the gradient to optimize both speech intelligibility and listening comfort. The invention also provides a hearing aid (50).

Abstract: A fitting device for fitting a hearing device to compensate for a hearing loss of a user is disclosed, the hearing device comprising a receiver and a microphone, wherein a feedback path exists between the receiver and the microphone, wherein the hearing device further comprises an adaptive feedback canceller configured to reduce feedback, the adaptive feedback canceller comprising a fixed filter corresponding to an invariant portion of the feedback path, and an adaptive filter corresponding to a variant portion of the feedback path, the fitting device comprising a processor configured to determine the invariant portion of the feedback path, wherein the processor is configured to provide the fixed filter with information relating to the invariant portion of the feedback path independently of the user using the hearing device.

Abstract: Methods, systems, and devices for processing an auditory signal to generate auditory prosthesis electrode stimuli in response to an incoming acoustic signal are disclosed. An example method includes dividing the incoming acoustic signal into one or more frequency envelopes. The example method also includes determining which auditory prosthesis electrodes to stimulate. The example method additionally includes determining a temporal reference point to which auditory prosthesis electrode stimuli are referenced. The example method further includes determining a delay for each of the auditory prosthesis electrode stimuli from the temporal reference point. The example method yet further includes determining amplitudes of the auditory prosthesis electrode stimuli. The example method also includes determining a wait period length before each successive temporal reference point.

Abstract: A method of processing a voice segment includes checking whether a voice segment is a vowel segment. If the voice segment is not a vowel segment, then the process checks whether the voice segment is a high frequency consonant or a low frequency consonant. If the voice segment is a high frequency consonant, then the voice segment will be processed to lower its frequency.

Abstract: Disclosed is an auditory comprehension and audibility device that assists users in hearing and comprehending auditory signals. The auditory comprehension and audibility device includes a magnetostrictive pad and a multi-turn coil that is disposed on the magnetostrictive pad. The multi-turn coil is electrically isolated from other devices, such as a speaker. An optional permanent magnet can also be placed on the magnetostrictive pad, which may help increase audibility of said auditory signals. The auditory comprehension and audibility device can be used in mobile phone covers, head phones, hearing aids and other devices that are used for hearing.

Abstract: In accordance with an embodiment, a system for processing an incoming signal for audio application includes a parameter extraction module coupled for receiving an incoming signal having a first frequency that is configured to extract at least one parameter from the incoming signal. The system includes at least one oscillator and at least one processing engine processing that receives a clock signal from the at least one oscillator. In accordance with another embodiment, a method is provided that performs a first processing with the clock signal and generates an oscillator control signal for adjusting the frequency of the clock from the oscillator adaptively depending on parameters of the incoming signal extracted or analyzed in the first processing.

Abstract: A method of processing a signal in a hearing aid (50) comprising the steps of determining a gradient of a speech intelligibility measure using a closed form expression of the gradient and modifying the frequency dependent hearing aid gains in dependence on the calculated gradient. The invention also provides a hearing aid (50).

Abstract: An active medical device, including an input receiver configured to receive a frequency-varying input signal, and a functional component that reacts to the input signal and consumes power at a rate dependant on the frequency of portions of the input signal to which the functional component reacts, wherein the active medical device is configured to adjust one or more portions of the input signal corresponding to portions of the input signal where the functional component consumes power at a rate that is greater than that of other portions of the input signal.

Abstract: Disclosed herein, among other things, are apparatus and methods for a threshold-derived fitting rationale using frequency translation for hearing assistance devices. In various method embodiments, a first audiogram is received for a first hearing assistance device for a wearer, and a second audiogram is received for a second hearing assistance device for the wearer. The first audiogram and the second audiogram are compared to audiometric thresholds to determine if frequency translation should be enabled.

Abstract: A hearing aid comprising a frequency shifter (20) has means (22) for detecting a first frequency and a second frequency in an input signal. The frequency shifter (20) transposes a first frequency range of the input signal to a second frequency range of the input signal based on the presence of a fixed relationship between the first and the second detected frequency. The means (34, 35, 36) for detecting the fixed relationship between the first and the second frequency is used for controlling the frequency transposer (20). A speech detector (26) configured for detecting the presence of voiced and unvoiced speech is provided for suppressing the transposition of voiced-speech signals in order to preserve the speech formants. The purpose of transposing frequency bands in this way in a hearing aid is to render inaudible frequencies audible to a user of the hearing aid while maintaining the original envelope, harmonic coherence and speech intelligibility of the signal.

Abstract: The present invention is directed to a method and a hearing device for extending a usable frequency range of an analog input signal (i) being processed by a hearing device, the method comprising the steps of converting the analog input signal (i) to a first output signal (o1) and to an intermediate signal (om), the first output signal (o1) having a final sampling rate and the intermediate signal (om) having an intermediate sampling rate that is greater than the final sampling rate, applying a band-pass filter unit (31) to the intermediate signal (om) in order to obtain a filtered intermediate signal (omf), a lower cut-off frequency of the band-pass filter unit (31) being above half the final sampling rate, an upper cut-off frequency of the band-pass filter unit (31) being below half the intermediate sampling rate, and shifting a spectrum of the filtered intermediate signal (omf) to a frequency range being below the final sampling rate to obtain an intermediate output signal (om2).

Abstract: In general, in one aspect, a method performed by one or more processes executing on a computer systems includes receiving an audio signal comprising a range of audio frequencies including high frequencies and low frequencies, converting a first portion of the range of audio frequencies into haptic data, shifting a second portion of the range of audio frequencies to a different range of audio frequencies, and presenting at least one of the converted first portion and the shifted second portion to a human user. Other implementations of this aspect include corresponding systems, apparatus, and computer program products.

Abstract: A system for remapping an audio range to a human perceivable range includes an audio transducer configured to output audio and a processing circuit. The processing circuit is configured to receive the audio from an audio input, analyze the audio to determine a first audio range, a second audio range, and a third audio range. The processing circuit is further configured to use frequency compression on the first audio range based on the second audio range and third audio range to create a first open frequency range, move the second audio range into the first open frequency range to create a second open frequency range, move the third audio range into the second open frequency range, and provide audio output including the compressed first audio range, the moved second audio range, and the moved third audio range.

Abstract: The present invention relates to a new method for effective estimation of signal processing parameters in a hearing aid. It is based on an interactive estimation process that incorporates—possibly inconsistent—user feedback. In particular, the present invention relates to optimization of hearing aid signal processing parameters based on Bayesian incremental preference elicitation.

Abstract: A personal communication device provides hearing support, and includes an input receiving an audio signal, a programmable processor, and an output outputting the processed signal. The programmable processor performs a filtering operation on a digital version of the audio signal in a first signal path based on parameter settings and provides control logic that determines the parameter settings based on information on user preferences, on audiological information and on information on the listening situation to provide hearing loss compensation. The programmable processor has a second signal path in parallel with the first signal path that receives the digital version of the audio signal and parameter settings and determining the filter coefficients of the filtering operation based on the parameter settings and the digital version of the audio signal.

Abstract: A method of pre-processing a sound signal including music for an auditory prosthesis is provided. An input sound signal is received. The sound signal is processed using music pre-processing software to produce a music pre-processed signal. The music pre-processed signal is presented for further processing, so as to produce a corresponding stimuli signal.

Abstract: The invention describes a hearing improvement device including components for training the brain to connect and stimulate the auditory and visual corteces and strengthen pathways between them, in order to overcome the shortcomings of the damaged auditory channel.

Abstract: The present invention relates to a listening device for a hearing impaired person. The present invention furthermore relates to a corresponding operating method of operating a listening device and to a corresponding computer program. In particular, the present invention relates to a listening device that comprises a signal processing unit that is controlled by a controller configured to implement a combined feed-forward and feed-back control in order to ensure that both an electric input signal and a processed electric output signal have at least almost identical modulation index values. Thereby, speech intelligibility is increased, in particular for a hearing impaired person being capable of perceiving sound pressure levels in a substantially decreased dynamic range.

Abstract: A method of processes audio signals picked up from a sound field by a microphone system of a listening device adapted for being worn at a particular one of the left or right ear of a user, the sound field comprising sound signals from one or more sound sources, the sound signals impinging on the user from one or more directions relative to the user. Information about a user's Ear, Head, and Torso Geometry and the user's hearing ability in combination with knowledge of the spectral profile and location of current sound sources provide the means for deciding upon which frequency bands that, at a given time, contribute most to the BEE seen by the listener or the Hearing Instrument. For a given sound source, a number of donor frequency bands is determined at a given time, where an SNR-measure for the selected signal is above a predefined threshold.

Abstract: A method and device for frequency compression of audio signals to reduce the occurrence of artifacts. A component of the audio signal having a plurality of frequency channels is shifted from a first frequency channel into a second frequency channel. A dominant instantaneous frequency is determined in the first frequency channel. During shifting or mapping, first the entire first frequency channel, including the dominant instantaneous frequency, is shifted or mapped into the second frequency channel, wherein the dominant instantaneous frequency obtains an intermediate frequency position. A final frequency position for the dominant instantaneous frequency is determined using a predefined compression characteristic in the second frequency channel, starting from the frequency position of the dominant instantaneous frequency in the first frequency channel. Finally, the dominant instantaneous frequency is shifted or mapped from the intermediate frequency position to the final frequency position.

Abstract: An frequency shift keying (FSK) transceiver for use in a hearing aid is capable of automatic tuning of the transceiver frequency. The automatic tuning makes use of a phase difference detector and a tuning manager adapted to control the frequency of a resonant circuit of the FSK transceiver according to information received from the phase difference detector. The phase difference detector measures the phase difference across a component of the resonant circuit of the FSK transceiver. If the phase difference of a binary FSK symbol is equal to the phase difference of the opposite binary FSK symbol, the resonant circuit is tuned, otherwise the tuning manager performs a tuning procedure in order to minimize the difference in phase differences, and adjusts the frequency of the resonant circuit accordingly.

Abstract: Systems, apparatus, and techniques are described to perform operations including receiving speech that includes audio signals from a talker. A fundamental frequency, unique to the talker, is extracted from the audio signals. A tone in frequency with the extracted fundamental frequency of the received speech is modulated and the modulated tone is delivered to an audio transducing device.

Abstract: A method of processing sound in a hearing aid (100, 300, 400) comprises separating the input transducer signal into a periodic and aperiodic signal for further processing in the hearing aid (100, 300, 400). The invention also provides a hearing aid (100, 300, 400) adapted for carrying out such a method of sound processing.

Abstract: A communication system comprises a transmitter unit (10, 60) arranged to be carried by or located near a user, and an earpiece including a receiver unit (30, 70), the system operating with a carrier frequency of less than 100 kHz, preferably less than 30 kHz, most preferably in the range 14 to 20 kHz. The system employs vestigial sideband modulation by cutting off part of an upper sideband. The received signal is digitized and then sampled to an envelope-tracking sampled-data root-filter square detector (40) configured as a digital signal processor.

Abstract: A method of analyzing an audio signal is disclosed. A digital representation of an audio signal is received and a first output function is generated based on a response of a physiological model to the digital representation. At least one property of the first output function may be determined. One or more values are determined for use in analyzing the audio signal, based on the determined property of the first output function.

Abstract: A hearing aid device (2) is disclosed. The hearing aid device (2) is configured to be sealingly inserted into the bony region (20) of the ear canal (16). The hearing aid device (2) comprises a receiver (12) and at least one microphone (4, 6) and means for processing sound signals detected by the at least one microphone (4, 6). The hearing aid device (2) comprises means for carrying out electronic feedback suppression by applying a frequency shift (SF) and/or a time delay to the detected sound signals. A method for providing electronic feedback reduction in a hearing aid device (2) is also disclosed.

Abstract: Aspect of the present invention are generally directed to synchronization between a first auditory prosthesis and a second auditory prosthesis of a bilateral auditory prosthesis system. In this system, a primary wireless communications channel, usable for synchronizing the first and second prostheses, may be disabled to, for example, conserve power. Then, upon detection of a trigger in sound received by one or more of the prostheses, the primary wireless communication channel is enabled and the prostheses synchronized using the primary wireless communication channel.

Abstract: A method and apparatus for adding audible noise with time varying volume to audio devices are disclosed which makes the time varying volume envelope of the added audible noise proportional to the time varying volume envelope of sound for frequencies where an individual has a restricted range of perception. The method and apparatus are used to improve the audibility, speech intelligibility, and word recognition characteristics in audio devices.