Showing papers on "Digital hearing aid published in 2011"

Adaptive Feedback Cancellation With Band-Limited LPC Vocoder in Digital Hearing Aids

Abstract: Feedback oscillation is one of the major issues with hearing aids. An effective way of feedback suppression is adaptive feedback cancellation, which uses an adaptive filter to estimate the feedback path. However, when the external input signal is correlated with the receiver input signal, the estimate of the feedback path is biased. This so-called “bias problem” results in a large modeling error and a cancellation of the desired signal. This paper proposes a band-limited linear predictive coding based approach to reduce the bias. The idea is to replace the hearing-aid output with a synthesized signal, which sounds perceptually the same as or similar to the original signal but is statistically uncorrelated with the external input signal at high frequencies where feedback oscillation usually occurs. Simulation results show that the proposed algorithm can effectively reduce the bias and the misalignment between the real and the estimated feedback path. When combined with filtered-X adaptation in the feedback canceller, this approach reduces the misalignment even further.

A MEMS-Based Power-Scalable Hearing Aid Analog Front End

Abstract: A dual-channel directional digital hearing aid front end using microelectromechanical-systems microphones, and an adaptive-power analog processing signal chain are presented. The analog front end consists of a double differential amplifier-based capacitance-to-voltage conversion circuit, 40-dB variable gain amplifier (VGA) and a power-scalable continuous time sigma delta analog-to-digital converter (ADC), with 68-dB signal-to-noise ratio dissipating 67 μ W from a 1.2-V supply. The MEMS microphones are fabricated using a standard surface micromachining technology. The VGA and power-scalable ADC are fabricated on a 0.25-μ m complementary metal-oxide semciconductor TSMC process.

Low-Delay Signal Processing for Digital Hearing Aids

Abstract: Digital signal processing in modern hearing aids is typically performed in a subband or transform domain that introduces analysis-synthesis delays in the forward path. Long forward-path delays are not desirable because the processed sound combines with the unprocessed sound that arrives at the cochlea through the vent and changes the sound quality. Nonetheless, subband domain processing for digital hearing aids is the most popular choice for hearing aids because of the associated computational simplicity. In this paper, we present an alternative digital hearing aid structure with low-delay characteristics. The central idea in the paper is a low-delay spectral gain shaping method (SGSM) that employs parallel parametric equalization (EQ) filters. The low-delay SGSM provides frequency-dependent amplification for hearing loss compensation with low forward path delays and performs dynamic signal processing such as noise suppression and dynamic range compression. Parameters of the parametric EQ filters and associated gain values are selected using a least-squares approach to obtain the desired spectral response. The low-delay structure also employs an off-the-forward-path, frequency domain adaptive filter to perform acoustic feedback cancellation. Extensive MATLAB simulations and subjective evaluations of the results indicate that the method of this paper is competent with a state-of-the-art digital hearing aid system, but exhibits much smaller forward-path delays.

Method and equipment for full frequency domain digital hearing aid

Abstract: 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 performingthe 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 proposalprovided 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 0.964mW digital hearing aid system

Abstract: This paper concerns the design and optimization of a digital hearing aid application. It aims to show that a suitably adapted ASIP can be constructed to create a highly optimized solution for the wide variety of complex algorithms that play a role in this domain. These algorithms are configurable to fit the various hearing impairments of different users. They pose significant challenges to digital hearing aids, having strict area and power consumption constraints. First, a typical digital hearing aid application is proposed and implemented, comprising all critical parts of today's products. Then a small area and ultra low-power 16-bit processor is designed for the application domain. The resulting hearing aid system achieves a power reduction of > 56 × over the RISC implementation and can operate for > 300 hours on a typical battery.

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

Abstract: 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.

Adaptive expert system for audiologists

Abstract: Presently audiological investigations are done in a speciality hospital and the test results are analyzed and diagnosed by the audiologists. However, the fact remains that most of us do not undergo regular checking for hearing due to the reasons of inconvenient timing and ease of accessibility. The object of this work is to design and develop a computerized audiometer, which could be effectively used for mass screening of level of hearing impairment instead of the conventional audiometer. This would be user friendly, cost effective and efficient in terms of analysis, data storage and maintenance. The present work successfully enables subjects to conduct hearing screening tests fully with the help of multimedia computers without any additional accessories. The audiological tests could be conducted regularly so as to facilitate the early detection of hearing loss at home or any place and time convenient to the user. At first, the design requirements for a digital hearing aid is being arrived by using the standard Real Ear Insertion Gain (REIG) formulae followed in Australia and European countries. Subsequently, based on the estimated value of minimum threshold of hearing arrived from this proposed set up, in addition to inputs from expert audiologists, the REIG formula could be made distinct for every language.

Signal processing method of digital hearing aid

Abstract: PURPOSE: A method for processing the signal of a digital hearing aid is provided to automatically erase specific peripheral noises of a narrow frequency band using a spectrum noise erasing algorithm. CONSTITUTION: In a frame, a digital signal is processed with respect to 128 input signals. A final voice signal is transferred to an output buffer. A programming algorithm is implemented in order to synchronize 32 output buffers to a receiver in 0.0625msec of sampling time. A spectrum amplitude modulation signal is processed after a fast Fourier transform is implemented. A fast Fourier reverse transform algorithm is implemented to calculate output voice data. A spectrum amplitude modulation algorithm is implemented to mainly concentrate conversation voice.

Digital hearing aid and method for adjusting parameters thereof by using sound from dual-tone multi-frequency key

Abstract: The invention relates to a digital hearing aid which comprises a voice pickup device, an A/D converter for converting a first analog signal received by the voice pickup device into a first digital signal, a microprocessor for performing data processing to the first digital signal, a D/A converter for converting the obtained second digital signal into a second analog signal, and a voice output device, wherein the microprocessor comprises an inspection judgment module for analyzing the frequency spectrum characteristics of the first digital signal and analyzing key sound commands in the frequency spectrum characteristics, and a parameter adjustment module for adjusting the parameters of the digital hearing aid according to the key sound commands. The invention also relates to a method for adjusting the parameters of the digital hearing aid by using sound from a dual-tone multi-frequency key. The parameters of the digital hearing aid can be conveniently adjusted by a wearer at any moment according to the key arranged based on the dual-tone multi-frequency technology, so that the heading-aid service is closer to the subjective feelings of a user.

Application of neural networks to speech/music/noise classification in digital hearing aids

Abstract: This paper focuses on the development of an automatic sound classifier embedded in a digital hearing aid aiming at enhancing the listening comprehension when the user goes from a sound environment to another different one. The approach we propose in this paper consists in using a neural network-(NN-) based sound classifier that aims to classify the input sound signal among speech, music or noise. The key reason that has compelled us to choose the NN-based approach is that neural networks are able to learn from appropriate training pattern sets, and properly classify other patterns that have never been found before. This ultimately leads to very good results in terms of higher percentage of correct classification when compared to those from other popular algorithms, such as, for instance, the k-nearest neighbor (k-NN) or mean square error (MSE) classifier, as clearly shown in the results obtained in this paper.

Survey of the Development of Digit Hearing Aid and its Algorithms

Abstract: The paper introduced the development of digital hearing aid firstly,then analyzed and compared three kinds of signal processing algorithms applied in digital hearing aid:multi-channel frequency compensation,denoise and acoustic feedback cancellation.Finally,the future research prospect towards the development trend of hearing aids is given.

64 Channel Digital Hearing Aid Emdedded Firmware

Abstract: This paper presents some results of a 64 channel digital hearing aid firmware development. Along with nonlinear voice compression, we present noise reduction, microphone & receiver calibration and in-situ algorithms with some results. The FFT-iFFT based compression method can generalize the nonlinear arbitrary compression scheme in connection with spectral noise reduction scheme. Spectral subtraction method was used for noise reduction, but noise spectrum was continuously updated by detecting signal level.

디지털 보청기 사용자를 위한 압신 알고리즘의 성능 연구

Abstract: Companding algorithms have been used to enhance speech recognition in noise for cochlea implant users. The efficiency of using companding for digital hearing aid users is not yet validated. The purpose of this study is to evaluate the performance of the companding for digital hearing aid users in the various hearing loss cases. Using HeLPS, a hearing loss simulator, two different sensorinerual hearing loss conditions were simulated; mild gently sloping hearing loss(HL1) and moderate to steeply sloping hearing loss(HL2). In addition, a non-linear compression was simulated to compensate for hearing loss using national acoustic laboratories-non-linear version 1(NAL-NL1) in HeLPS. In companding, the following four different companding strategies were used changing Q values(q1, q2) of pre-filter(F filter) and post filter(G filter). Firstly, five IEEE sentences which were presented with speech-shaped noise at different SNRs(0, 5, 10, 15 dB) were processed by the companding. Secondly, the processed signals were applied to HeLPS. For comparison, signals which were not processed by companding were also applied to HeLPS. For the processed signals, log-likelihood ratio(LLR) and cepstral distance(CEP) were measured for evaluation of speech quality. Also, fourteen normal hearing listeners performed speech reception threshold(SRT) test for evaluation of speech intelligibility. As a result of this study, the processed signals with the companding and NAL-NL1 have performed better than that with only NAL-NL1 in the sensorineural hearing loss conditions. Moreover, the higher ratio of Q values showed better scores in LLR and CEP. In the SRT test, the processed signals with companding(SRT

System and method for remotely carrying out audiometric measurements and for adjusting hearing aids via the internet between an audiologist belonging to a network of audiologists and a connected patient

Abstract: System and method for remotely carrying out audiometric measurements and for adjusting hearing aids via the Internet between an audiologist belonging to a network of audiologists and a patient connected to a hearing point. The method is carried out in a system by organizing communication, via the Internet, between the terminal associated with an audiologist belonging to a network of associated centres and the patient who uses one of the terminals in associated audiological centres or his own access terminal. The signals for adjusting the hearing aid are sent by the terminal associated with the audiologist, via the Internet, to the digital hearing aid connected to an access terminal belonging to the client or to a hearing-aid manufacturer and/or distributor authorized by the client.

Low-Delay Signal Processing for

Abstract: Digital signal processing in modern hearing aids is typically performed in a subband or transform domain that introduces analysis-synthesis delays in the forward path. Long forward-path delays are not desirable because the processed sound combines with the unprocessed sound that arrives at the cochlea through the vent and changes the sound quality. Nonethe- less, subband domain processing for digital hearing aids is the most popular choice for hearing aids because of the associated computational simplicity. In this paper, we present an alternative digital hearing aid structure with low-delay characteristics. The central idea in the paper is a low-delay spectral gain shaping method (SGSM) that employs parallel parametric equalization (EQ) filters. The low-delay SGSM provides frequency-dependent amplification for hearing loss compensation with low forward path delays and performs dynamic signal processing such as noise suppression and dynamic range compression. Parameters of the parametric EQ filters and associated gain values are selected using a least-squares approach to obtain the desired spectral response. The low-delay structure also employs an off-the-forward-path, frequency domain adaptive filter to perform acoustic feedback cancellation. Extensive MATLAB simulations and subjective evaluations of the results indicate that the method of this paper is competent with a state-of-the-art digital hearing aid system, but exhibits much smaller forward-path delays. Index Terms—Adaptive filters, hearing aids, low-delay.

Acoustic Source Localization Based on Multichannel Compressed Sensing and Auditory Bionics in Hearing Aids

Abstract: The microphone array is an effective means in acoustic location. However, it has become very difficult to acquire adequate positional parameters when the space between the microphone pairs is too small. To this problem, the paper proposed a signal model which contains source information constructed by differential microphone. Moreover, to decrease computation complexity, we obtain acoustic source data by multichannel compressed sensing(CS). Finally, acoustic source location is obtained by the estimation of energy in reconstructed signal. Theoretical analysis and simulation results conclude that the proposed approach exhibits a number of advantages over other source localization techniques.

Digital Hearing Aid and Cochlear Implant

Abstract: Approximately 70 million individuals worldwide suffer from hearing loss, which makes it the most common sensory disorder in the world [1–3]. There are estimated 28 million individuals with hearing loss in the United States. Hearing loss affects 17 in 1,000 children under the age of 18, with the incidence increasing with age. Approximately 314 in 1,000 people over the age of 65 have hearing loss, and 40–50% of people 75 and older have hearing loss.