143 results about "Digital hearing aid" patented technology

A digital hearing aid according to the present invention is capable of measuring its own performance. The measurement and initialization capability may be entirely integral to the hearing aid, or an external processor may be used to download the measurement program and the run time program, and assist in computing the parameters. The hearing aid includes a test signal generator for feeding a test signal into the hearing aid amplifier. The response to the test signal is acquired at a specific point in the hearing aid, depending upon what aspect of performance is to be measured. Various elements of the hearing aid and / or the hearing aid feedback may be bypassed. The hearing aid further includes the capability of initializing hearing aid parameters based upon the performance measurements.

A hearing compensation system comprises an input transducer for converting acoustical information at an input thereof to electrical signals at an output thereof, a differential analog-to-digital converter sampling the electrical signals output from the input transducer at an input thereof and outputting differential signal samples at an output thereof, a digital signal processing circuit having an input connected to the output of the differential analog-to-digital converter and operating on the differential signal samples to form processed differential signal samples at an output thereof, and an output transducer for converting electrical signals at an input thereof to acoustical information at an output thereof, the processed differential signal samples coupled to the input of the output transducer.

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.

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.

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.

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.

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.

The invention discloses a voice enhancement method for fusing phase estimation and human ear hearing characteristics in a digital hearing aid, comprising steps of obtaining a frequency domain expression mode containing noise through Fourier transformation, adopting a minimum value control recursive average method to obtain a noise power spectrum, obtaining initial enhancement voice and a noise amplitude spectrum, obtaining the initial enhancement voice and noise through correcting the phase of the voice and the noise through improving the phase estimation of the voice distortion under the low signal-to-noise ratio environment, performing filtering processing on the initial enhancement voice and the noise through a gammatone filter assembly which simulates the working mechanism of the artificial cochlea, performing analysis on the time frequency of the gammatone filter assembly to obtain the time frequency expression form consisting of time frequency units, using the hearing characteristics of the human ear to calculate binary mast containing noise in the time frequency domain, and using the mask value to obtain the enhanced voice after synthesis.

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.

Disclosed is a digital hearing aid fitting system capable of changing a frequency band and channel. The digital hearing aid fitting system capable of changing the frequency band and channel may include a digital signal processing (DSP) chip to adjust a fitting parameter including at least one parameter from among a number of bands of a frequency, a bandwidth, and a number of channels of the frequency, and a controlling unit to control the DSP chip based on a hearing test result to adjust the fitting parameter to a person with hearing loss.

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.

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.

The present invention relates to a digital hearing aid, which models the structures of external ear canals, sizes and shape characteristics of which differ between respective persons, obtains resonance gains generated due to the structural characteristics of the external ear canals, and performs digitization and signal processing to allow the resonance gains to be used as the gain factors of the digital hearing aid, and thus applies the gain factors to digital signal processing units. Further, the present invention proposes a gain obtainment unit capable of taking both resonance gains, generated due to the structural characteristics, and gains, obtained through a hearing test, into account, thus reducing the time required for gain fitting and possible errors, and optimizing the performance of the digital hearing aid for each individual.

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.

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.

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.

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.

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.

A mobile communication terminal interacts with a digital hearing aid and a wireless hearing aid uses the mobile communication terminal. When a user having the mobile communication terminal interworking with the digital hearing aid operates the hearing aid, an amplification function suitable for auditory characteristics of the user is performed. That is, when auditory information of the user is sent to the terminal, the terminal adjusts an amplification gain to prevent the acoustic shock by measuring an environmental signal. The terminal computes an environment profile by analyzing the measured environmental signal and auditory information of the user sent from the hearing aid and automatically adjusts a non-linear amplification level according to a user environment by sending the environment profile to the hearing aid.

The invention discloses a wide dynamic compression method of a digital hearing aid based on sound pressure level segmentation, wherein the wide dynamic compression method comprises the following steps: framing a voice signal, and filtering the framed signal through a 16-channelnonuniform spaced 6-order IIR decomposition filter bank; then, calculating the sound pressure levels of the voice signals of the channels, and obtaining a hearing compensation curve of a patient in combination with an audiogram of the patient; carrying out channel-separated hearing compensation on the patient according to the hearing compensation curve, integrating compensated multichannel signals to obtain a compensated useful signal, and providing the useful signal for the patient. The wide dynamic compression method disclosed by the invention has the beneficial effects that, the sound pressure level is thinned in 8 segments to obtain more accurate input / output curves; and the 6-order IIR decomposition synthesis filter bank conforming to human hearing characteristics can be used for obtaining a compensation gain value better conforming to the actual demand of the patient.

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.

A digital hearing aid is disclosed in the present disclosure. The digital hearing aid comprises a body hung behind an ear and a sound transmitting device for receiving and outputting sound that is placed within an auricle and extends into an external auditory canal. The body and the sound transmitting device are connected and fixed together via a connecting device. The connecting device is formed of a material and a structure which are bendable, extendable and capable of maintaining the bending deformation, and the connecting device is preformed into a shape of an ear hook. In this way, the present disclosure can have the advantages of both the behind-the-ear hearing aid and the complete-in-canal hearing aid and is adaptive to the shape of the auricle and the facial structure so as to meet the demands of people in different growth stages or of different people.

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.

The invention discloses a full-digital hearing aid which comprises a digital signal processor. In the digital signal processor, at least two hearing compensation prescription modes undergoing corresponding hearing aid parameter programming or non-linear signal processing are preset according to typical non-linear hearing loss graph types of a target patient suffering from hearing impairment. The patient suffering from hearing impairment does not need special fitting services of a traditional digital hearing aid, can directly wear the hearing aid, chooses one hearing compensation prescription mode suitable for the hearing loss type and the hearing loss degree of the patient suffering from hearing impairment, adjusts volume in the hearing compensation prescription mode, experiences the hearing compensation effect according to self feeling, and can choose the hearing compensation mode and the volume, wherein the special fitting services of the traditional digital hearing aid include but are not limited to hearing tests and hearing aid programming and debugging. The full-digital hearing aid can meet the requirement for fitting of the patient suffering from hearing impairment and is wide in application range. The invention further discloses a hearing aid fitting method.

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.

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.

The invention relates to the technical field of voice signal processing and especially relates to a digital hearing aid voice noise elimination method. The method mainly comprises the following steps: S1) carrying out sub-band division and framing on sampled signals of input voice to obtain each frame of sub-band noise speech signal; S2) calculating a gain function of each frame of sub-band noise speech signal; S3) damping each frame of sub-band noise speech signal according to the gain function to obtain each frame of sub-band enhancing signal; and S4) enabling the sub-band enhancing signals to pass through synthesis filter banks and be added to obtain noise-reduced voice and then, outputting the noise-reduced voice. According to the method, computation complexity is reduced, and thus time delay and power consumption is reduced; and meanwhile, an improved algorithm enables signal-to-noise ratio of the noise-reduced voice to be improved by more than 5 dB, so that noise in the speech signals is effectively suppressed, intelligibility of the speech is improved and higher practical value is achieved.

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.

The invention discloses a hearing-aid device and method based on a smart phone. The device and method have the advantages that 1) a digital hearing-aid function is realized by the smart phone, use of an expensive special-purposed DSP chip of a digital hearing aid is avoided, the cost can be reduced, convenience is provided for use and carrying, and thus, corrective therapy can be carried out on more patients with hearing impairment; 2) a single-channel microphone of the smart phone is used to realize the digital hearing-aid function, it is not required to use a microphone array to collect signals, and the cost and circuit complexity are reduced; 3) sound signals collected by the microphone of the smart phone are divided into N channels, the hearing compensation parameter of the patients with hearing impairment is adjusted in each channel, so that the hearing compensation effect is better, and use is more comfortable; and 4) in each hearing compensation channel, hearting compensation is carried out by using sound pressure level segmented compression figure line, residual hearing of the patients can be protected, and the compensation effect can be ensured.