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143 results about "Digital hearing aid" patented technology

Automated real speech hearing instrument adjustment system

InactiveUS20080298600A1Limited amplificationUnderstanding of audibility and comfortDeaf aid adaptationSpeech hearingEngineering
A method for fitting a hearing instrument comprises placing the hearing instrument in situ includes receiving an audiogram of the user, determining a target gain for the hearing instrument as a function of the audiogram, exposing a reference sensor located adjacent the hearing instrument to an external speech signal, measuring an external sound pressure level (SPL) via the reference sensor, exposing a probe sensor coupled to the inside of the ear to the output of the hearing instrument while the hearing instrument is in situ, measuring an internal sound pressure level (“SPL”) inside the ear of the user via the probe sensor, determining an offset gain as a function of the external SPL, the target gain and the internal SPL, and automatically adjusting a gain of the hearing instrument according to the offset gain. A system for automatically fitting a hearing impaired person with a digital hearing aid in situ includes a digital hearing aid, a reference volume sensor, a probe sensor, a sound mapping module and a parameter control module.
Owner:POE MICHAEL +3

Method and equipment for full frequency domain digital hearing aid

InactiveCN101593522AProcessing speedSolving Hearing Impairment ProblemsSpeech recognitionDeaf-aid setsVoice frequencyDynamic range
The embodiment of the invention provides a method for full frequency domain digital hearing aid, which comprises the following steps: firstly, acquiring input voice signals of front and back two microphones and performing framing, Fourier transformation and voice scene type recognition; secondly, when voice is mixed with noises, performing noise detection of subframe voice frequency domain signals, beamforming of the two microphones, wind noise processing and inhibition of other noises, compacting the dynamic ranges of frequency domains and inhibiting acoustic feedback; and finally performing the Fourier transformation and overlap-add to obtain output voice signals. The embodiment of the invention also discloses equipment for full frequency domain digital hearing aid. Through the proposal provided by the embodiment of the invention, the problem that the prior digital hearing aid focuses on solving only one aspect of hearing disorder rather than comprehensively take all factors influencing use effect into consideration is solved. Meanwhile, the embodiment of the invention provides a proposal for full frequency domain digital hearing aid. The method, the equipment and proposal have the advantages of quick processing, less resource occupation, low energy consumption and the like.
Owner:TSINGHUA UNIV

Inter-channel communication in a multi-channel digital hearing instrument

A multi-channel digital hearing instrument is provided that includes a microphone, an analog-to-digital (A / D) converter, a sound processor, a digital-to-analog (D / A) converter and a speaker. The microphone receives an acoustical signal and generates an analog audio signal. The A / D converter converts the analog audio signal into a digital audio signal. The sound processor includes channel processing circuitry that filters the digital audio signal into a plurality of frequency band-limited audio signals and that provides an automatic gain control function that permits quieter sounds to be amplified at a higher gain than louder sounds and may be configured to the dynamic hearing range of a particular hearing instrument user. The D / A converter converts the output from the sound processor into an analog audio output signal. The speaker converts the analog audio output signal into an acoustical output signal that is directed into the ear canal of the hearing instrument user.
Owner:HIMPP

Remote Programming System for Programmable Hearing Aids

Bluetooth-enabled programmable digital hearing aids can be adjusted remotely and in real time. The hearing aid patient at home or a location convenient to the patient logs on to a hearing clinic web site to connect with a clinician, e.g. audiologist or hearing aid technician, at a central clinic location. The patient computer is provided with software for this and the patient is provided with a wireless hearing aid adjustment device. The clinician downloads the patient's hearing aid program and other data, and then makes adjustments and reprograms the hearing aids in real time.
Owner:ZACCARIA RICHARD

Method for adapting sound in a hearing aid device by frequency modification and such a device

In a digital hearing aid device (1) frequency modification is employed above a lower spectral bound and in accordance with a compression factor. The frequency modification is dynamically adjusted in dependence on a sound environment analysis (10) or an end-user input (30), by modifying the frequency modification parameters such as a lower spectral bound and a compression factor. The adjustment can be based on an interpolation between predefined parameters. In certain sound environments, such as loud noise, own-voice and telephone conversations, frequency modification is reduced or switched off. The proposed solutions have the advantage that the occurrence of disturbing noise and of distortions of harmonic relationships at the end-user's ear is reduced and signal processing resources as well as battery resources are saved.
Owner:SONOVA AG

Digital hearing-aid unequal-width sub-band automatic gain control method

The invention provides a digital hearing-aid unequal-width sub-band automatic gain control method. The method includes: designing an unequal-width analysis filter bank according to human ear auditory characteristics, conducting segment processing to input acoustical signals, filtering each segment acoustical signal by using the analysis filter bank to obtain signals of various sub-bands, counting input signal sound pressure level inside the various sub-bands, conducting sub-band automatic gain adjustment according to the sub-band signal sound pressure level, finally using a synthesizing filter bank to synthesize adjusted sub-band signals to obtain system output signals. According to the digital hearing-aid unequal-width sub-band automatic gain control method, noise and voice signals are isolated out through the sub-band filter bank, and thus the gains of noise frequency ranges and the voice signals are respectively controlled, and the roles of noise suppression and signal-to-noise ratio improvement are played. Compared with a traditional equal-width sub-band filter bank partition method, an unequal-width filter bank sub-band partition method used by the digital hearing-aid unequal-width sub-band automatic gain control method coincides with the human ear auditory characteristics. On the premise of obtaining the same effect, the amount of sub-bands need parting and the calculated amount are reduced. In addition, the digital hearing-aid unequal-width sub-band automatic gain control method is achieved totally by a software program with no need of adding an additional hardware circuit, thereby saving the cost, weight and size of a product.
Owner:NANJING INST OF TECH

Wind noise suppression method used for dual-microphone digital hearing-aid

InactiveCN102254563AImprove the effect of hearing aidsWind noise reductionSpeech analysisDeaf-aid setsNoise suppressionComputer science
The invention relates to a wind noise suppression method used for a dual-microphone digital hearing-aid. The method comprises the following steps of: respectively carrying out framing processing on respectively accessed signals of a left microphone and a right microphone at the same time interval according to a preset frame length; then respectively calculating signal energy values of a left frame and a corresponding right frame as well as cross correlation coefficients of the left and the right frames in a left group of multiframe input signals and a right group of multiframe input signals; judging whether the current left frame and the current right frame which are calculated currently are voice frames or noise frames according to the preset single-microphone short-time energy threshold value and the preset double-microphone correlation proportion threshold value as well as the calculated respective signal energy values and the cross correlation coefficients of the left frame and the right frame, so as to be taken as the basis of a subsequent left frame and a subsequent right frame; when respective former left frame and respective former right frame of the calculated current left frame and the calculated right frame are voice frames, outputting the current left frame and the current right frame as the voice frames; and otherwise, outputting the current left frame and the current left frame through highpass filtering. Therefore, according to the invention, the wind noise is reduced, and the hearing aid effect is improved.
Owner:SHANGHAI CONGWEI ACOUSTICS TECH

Online Anti-feedback system for a hearing aid

The invention relates to a hearing aid system comprising an input transducer, a forward path, an output transducer and an electrical feedback path, the forward path comprising a signal processing unit for modifying an electrical input signal to a specific hearing profile over a predefined frequency range, wherein the predefined frequency range comprises a number of frequency bands, for which maximum forward gain values IGmax for each band can be stored in a memory, the electrical feedback path comprising an adaptive filter for estimating acoustical feedback from the output to the input transducer. The invention further relates to a method of adapting a hearing aid system to varying acoustical input signals. The object of the present invention is to provide an alternative acoustic feedback compensation scheme. The object is fulfilled in that the hearing aid system further comprises an online feedback manager unit for—with a predefined update frequency—identifying current feedback gain in each frequency band of the feedback path, and for subsequently adapting the maximum forward gain values in each of the frequency bands in dependence thereof in accordance with a predefined scheme. This has the advantage of providing a diminished probability for disturbing feedback improved feedback cancellation. The invention may e.g. be used in digital hearing aids for use in a variety of acoustical environments.
Owner:OTICON

Automatic fitting digital hearing aid and use method thereof

ActiveCN103079160ARealize the function of parameter self-adjustmentReduce volumeDeaf-aid setsOutput deviceAnalog signal
The invention discloses an automatic fitting digital hearing aid which comprises a voice picking device, an A/D (Analogue/Digital) converting device, a microprocessor for carrying out data processing on a first digital signal, a D/A converting device for converting an acquired second digital signal into a second analogue signal as well as a voice output device, wherein the microprocessor comprises a voice gaining and signal noise processing module, a key inputting module, an acoumeter, an audiometry program module, an automatic fitting module and a storage module; the key inputting module is used for determining whether to enter an acoumeter mode for automatically fitting; the acoumeter is used for determining the hearing loss condition of a patient; the audiometry program module is used for obtaining audiogram information of the patient; the automatic fitting module is used for automatically fitting the output result of audiometry program to determine a compression threshold, compression ratio and other parameters of the hearing aid; and the storage module is used for storing relevant data and parameters of the program. According to the automatic fitting digital hearing aid disclosed by the invention, a fitting formula is integrated to a DSP (Digital Signal Processor) chip and is planted into a hearing aid body. The automatic fitting digital hearing aid has the characteristics of convenience in wearing, high distinguishing efficiency, accuracy in fitting, effective protection of residual hearing and the like.
Owner:BEIJING TSINGCREA DEV

Method and apparatus for tuning digital hearing aids

A method for generating digital filters for tuning a hearing aid to enhance hearing ability. Data is provided, for a range for a target response curve representative of enhanced hearing ability of sound level versus frequency. Second digital data is also generated, for an initial response curve or audiogram of an individual's hearing ability. The first digital data is compared with the second digital data and it is determined whether the initial response curve is within the tolerance range. If not, digital audio filters are iteratively generated, and the digital audio filters are applied to digital data representing received sound signals to generate third digital data to enhance hearing. Parameters of the digital audio filters are automatically optimized until the compensated response curve is within the tolerance range or a predetermined number of digital audio filters has been reached.
Owner:TEXAS INSTR INC

Digital hearing aid loudness compensation method based on frequency compression and movement

ActiveCN102638755AImprove hearingImprove speech recognitionSpeech analysisDeaf-aid setsHearing testHearing level
The invention discloses a digital hearing aid loudness compensation method based on frequency compression and movement, which belongs to the technical field of voice signal processing. The digital hearing aid loudness compensation method comprises the following steps of: filtering, and extracting the low-frequency part and the high-frequency part of a signal; compressing and moving the voice high-frequency part according to proportion; simulating a voice signal heard by a patient; designing a four-channel FIR-QMFB (Quadrature Mirror Filter Banks); dividing a frequency range; and carrying out loudness compensation and gain control; and combining voice. The digital hearing aid loudness compensation method is characterized in that the original voice signal is divided into low frequency and high frequency, then, the high-frequency part is compressed and moved to a medium and low frequency range according to proportion, and the overlaid voice signal is subjected to frequency band division and loudness compensation according to a hearing test curve of a hearing impairment patient. Therefore, negative effects brought by carrying out high-gain loudness compensation on the high frequency range in the prior art can be effectively avoided, and the hearing level and the language recognition rate of the patient can be effectively improved.
Owner:HAIMEN MAOFA ART DESIGN CO LTD

Digital deaf-aid frequency response compensation method based on mask curve

InactiveCN1870135AResolve claritySolve the problem of decreased intelligibilitySpeech analysisDeaf-aid setsMasking thresholdFrequency response
A method for compensating frequency response of digital deaf-aid based on masked curve includes techniques of time frequency domain switch-over, critical band division, masking threshold calculation and frequency response compensation to improve hearing threshold rising phenomenon caused by hearing-masking effect.
Owner:北达万坤(北京)科技发展有限公司

Environmentally adaptive neural network noise reduction method and system for digital hearing aid and storage medium

The invention provides an environmentally adaptive neural network noise reduction method for a digital hearing aid. The method comprises the following steps of receiving a voice signal with noise, sampling and framing the voice signal with noise, and then transmitting the voice signal with noise to an acoustic scene recognition module; conducting scene recognition, wherein an acoustic scene recognition module is adopted for recognizing the current acoustic scene, and then the acoustic scene recognition module autonomously selects different neural network models in a neural network noise reduction module for transmission; performing neural network noise reduction. The method has the advantages that the real-time performance of voice processing can be ensured, only forward propagation of a neural network is carried out, and the calculation amount is not high; the current acoustic scene can be recognized, then different neural network models are selected autonomously, targeted noise reduction processing is carried out on noise in different scenes, and higher voice quality and voice intelligibility can be ensured; instantaneous noise can be effectively suppressed.
Owner:HARBIN INST OF TECH SHENZHEN GRADUATE SCHOOL

BP-artificial-neural-network-based intelligent matching algorithm for digital hearing aid

ActiveCN106714062AMature digital hearing aid intelligent fitting algorithmWide coverageHearing aids signal processingPattern recognitionFrequency spectrum
The invention discloses a BP-artificial-neural-network-based intelligent matching algorithm for a digital hearing aid. On the basis of a BP artificial neural network, a network is trained by using lots of training data to obtain a satisfactory mature network; and the network is corrected by using a self-built formula model, thereby obtaining a mature intelligent matching algorithm. On the basis of the genetic algorithm principle, an initial weight value and a threshold of a BP artificial neural network are optimized; the BP neural network is trained by using an existing audiogram and a spectrum gain response as raining data; and the network is corrected based on a matching formula model to obtain a mature BP artificial neural network to replace the existing matching prescription formula, so that parameters like all channel gains, the maximum sound output, the compression rate, and the compression inflection point of the digital hearing aid can be obtained.
Owner:SHENZHEN EARTECH CO LTD

WOLA (Weighted-Overlap Add) filter bank based signal processing method for all-digital hearing aid

InactiveCN102256200AEliminate feedbackOvercome severe high frequency distortionAdaptive networkDeaf-aid setsFourier transform on finite groupsSound pressure
The invention relates to a WOLA (Weighted-Overlap Add) filter bank based signal processing method for an all-digital hearing aid, which comprises the following steps of: firstly, sampling and blocking input signals of microphone access at a preset sampling frequency by using a preset algorithm in a time domain, and performing adaptive filtering; secondly, performing serial parallel conversion, analysis window interception and discrete Fourier transform on the adaptively filtered signal so as to separate each frequency sub-band; thirdly, calculating the target gain according to the preset sound pressure input gain curve, processing the calculated target gain through a 1-order IIR (Infinite Impulse Response) filter to obtain real-time dynamic gain, and compressing each frequency sub-band signal according to the real-time dynamic gain; and finally, transforming the compressed signals from a frequency domain to a time domain, and performing comprehensive window processing and parallel serial conversion on the obtained time domain signals to form serial signals to be output. Therefore, the problems of whistling, serious high-frequency signal distortion and too low sub-band energy compensation range of the traditional digital hearing aid during high magnification can be effectively solved.
Owner:SHANGHAI CONGWEI ACOUSTICS TECH

Chinese voice signal process method for digital deaf-aid

The invention describes a Chinese speech sound signal processing used in the digital hearing aids and belongs to an speech sound signal processing. The invention aims to the feature of containing large amount of languages and semantic information in the tunes of the Chinese speech, treats the signal's feature of the speech data stream in according to several frequency parts through frequency analysis and outputs the digital signal after the D / A. the speech sound signal got from the invention can make the listeners hear the Chinese speech more clearly, increases the recognition rate of the Chinese speech and semantics and can improve the effect of the hearing aid.
Owner:南京国光医疗器械有限公司

Voice enhancement method based on double-ear sound source positioning and deep learning in double-ear hearing aid

A vice enhancement method based on double-ear sound source positioning and deep learning in a double-ear digital hearing aid belongs to the field of voice signal processing. Firstly a two-stage deep neural network is used for accurately positioning a target voice, and noise in a direction which is different from the direction of target voice is eliminated according to spatial filtering. By means of a deep learning model in which a time delay control bidirectional long-short term memory deep neural network and a classifier are combined, an extracted multi-resolution hearing cepstrum coefficientis used as a characteristic input. Through nonlinear processing capability of deep learning, each time frequency unit of the noise-containing voice is classified to a voice time frequency unit or noise time frequency unit. Finally a voice waveform combining algorithm is used for eliminating the noise in the direction which is same with that of the target voice. The algorithm eliminates the noisein the direction which is different from the direction of the target voice and eliminates the noise in the direction that is same with the target voice, and finally obtains the enhanced voice which satisfies speech intelligibility and comfort of a deaf person. All deep learning models utilize offline training, thereby satisfying a requirement for real-time performance.
Owner:BEIJING UNIV OF TECH

Multi-channel wide dynamic range compressing system for digital hearing aid

The invention relates to a multi-channel wide dynamic range compressing system based on an audition perception model. The multi-channel wide dynamic range compressing system comprises an analysis filter group for simulating an audition perception model, a sound pressure level detecting module, a compression amplification gain calculating module, a multiplier and an integrated filter group for simulating the audition perception model, wherein audio digital signals x (n) are divided into K channels after passing through the analysis filter group, the sound pressure level detecting module is used for detecting the sound pressure level of each channel, the compression amplification gain calculating module is used for calculating the specific gain value of each channel, the multiplier can multiply the gain values of the channels with corresponding sub-band signals, and the obtained results of the multiplier are integrated into a path of output signal y (n) through the integrated filter group, the integrated filter group respectively carries out the all-pass transformation and all-pass inverse transformation in an analysis filter group and an integrated filter group of a weighted splice adding structure through the mode of combining the weighted splice adding structure and the all-pass transformation, and can simulate ear audition resolution of a human under the condition of fewer channels.
Owner:INST OF ACOUSTICS CHINESE ACAD OF SCI
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