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Showing papers on "Digital hearing aid published in 2010"


Journal ArticleDOI
TL;DR: The proposed low-power ANSI 1/3-octave bank makes itself being able to precisely apply the prescribed gains obtained by NAL-NL1 prescription formula for hearing-impaired people.
Abstract: Due to well matching the frequency characteristics of human ears, ANSI S1.11 1/3-octave filter bank is popular in acoustic applications, such as acoustic analyzers and equalizers. It is also desirable in hearing aids because the famous hearing aid prescription formula, NAL-NL1, prescribes its gains at ANSI 1/3-octave frequencies. However, the high computation complexity limits its usage, in which the power consumption is a critical concern. To address this issue, a low-power design and implementation of ANSI S1.11 filter bank for digital hearing aids is present. We first develop the complexity-effective multirate FIR filter bank algorithm. And, a systematic coefficient design flow is elaborated for the proposed filter bank to minimize the order of the FIR filter thereof. In an 18-band digital hearing aid with 24-kHz sampling rate, the proposed algorithm saves about 96% of multiplications and additions, comparing that with a straightforward FIR filter bank. Moreover, various low-power VLSI design techniques are investigated in detail and applied on our design. The proposed complexity-effective ANSI S1.11 FIR filter bank has been implemented in the TSMC 0.13-μ m CMOS technology with an area-efficient architecture. The test chip consumes only 87 μW, which is 30%-79% of that of the others available in the literature. The proposed low-power ANSI 1/3-octave bank makes itself being able to precisely apply the prescribed gains obtained by NAL-NL1 prescription formula for hearing-impaired people.

65 citations


Journal ArticleDOI
TL;DR: The analysis indicated that in these test conditions there was no change in sound quality when varying the delay in the range 5–10 ms and that there was a preference for 2000 Hz high-pass filtering in most conditions, regardless of the hearing losses tested.
Abstract: The combination of delayed sound from a digital hearing aid with direct sound through an open or vented fitting can potentially degrade the sound quality due to audible changes in timbre and/or perception of echo. The present study was designed to test a number of delay and high-pass combinations under worst-case (i.e. most sensitive) conditions. Eighteen normal-hearing and 18 mildly hearing-impaired subjects performed the test in a paired comparison (A/B) task. The subjects were asked to select a preferred setting with respect to sound quality. The test was set in an anechoic chamber using recorded speech, environmental sounds, and own voice. Experimental hearing aids were fitted binaurally with open domes thus providing maximum ventilation. The preference data were processed using a statistical choice model that derives a ratio-scale. The analysis indicated that in these test conditions there was no change in sound quality when varying the delay in the range 5–10 ms and that there was a preferen...

24 citations


Journal ArticleDOI
TL;DR: It is considered that both prostheses help children with hearing loss to have a more normalized voice quality than what scientific literature has traditionally stated.

19 citations


Patent
26 Apr 2010
TL;DR: In this article, a fitting system for a digital hearing aid capable of varying a frequency band and channel is provided to control fitting information using the same digital signal process chip regardless of design or model of the digital hearing aids using a fitting program.
Abstract: PURPOSE: A fitting system for a digital hearing aid capable of varying a frequency band and channel is provided to control fitting information using the same digital signal process chip regardless of design or model of the digital hearing aid using a fitting program CONSTITUTION: A digital signal process chip(110) changes fitting information including at least one of the number of frequency channels, the number of frequency bands, and the band range A controller(120) changes the fitting information to be suitable for a person who has difficulty in hearing by controlling a digital signal process chip based on the hearing test result A fitting information selection unit(121) selects the fitting information suitable for the person who has difficulty in hearing based on the hearing test result An information display unit(122) displays the hearing test result and the fitting information selected from the fitting information selecting unit A controller(123) changes the fitting information by controlling the digital signal process chip

15 citations


Patent
26 Feb 2010
TL;DR: In this paper, a power management system for a digital processing core (12) of a battery-powered hearing aid is adapted for providing power to the hearing aid circuit in a particularly efficient manner.
Abstract: A power management system (1) for a digital processing core (12) of a battery-powered hearing aid is adapted for providing power to the hearing aid circuit in a particularly efficient manner. The power management system (1) comprises a first linear voltage regulator (25, 26, 28), and a second linear voltage regulator (25, 27) in series with a switched-capacitor 2:1 SC converter (21), a positive bulk biasing voltage supply (10), and a negative bulk biasing voltage supply (11), for controlling the switching speed, threshold voltage, and current leak from the semiconductor elements (13, 14) of the digital processing core (12) when the core (12) is operated at the reduced voltage provided by the power management system (1). The power management system (1) may save between 50% and 70% of the power consumed by the digital processing core (12) of the hearing aid circuit when compared to existing hearing aids, and may thus prolong the battery life. The invention further provides a method for providing a supply voltage to a digital hearing aid.

15 citations


Journal ArticleDOI
TL;DR: The hearing aid presented in this article was developed with these specifications in mind together with additional contemporary features such as four channels with wide dynamic range compression, an adjustable compression rate for each channel, four comfort programs, an adaptive feedback manager, and full volume control.
Abstract: Hearing loss is a common health issue that affects nearly 10% of the world population as indicated by many international studies. The hearing impaired typically experience more frustration, anxiety, irritability, depression, and disorientation than those with normal hearing levels. The standard rehabilitation tool for hearing impairment is an electronic hearing aid whose main components are transducers (microphone and receiver) and a digital signal processor. These electronic components are manufactured by supply chain rather than by hearing aid manufacturers. Manufacturers can use custom-designed components or generic off-the-shelf components. These electronic components are available as application-specific or off-the-shelf products, with the former designed for a specific manufacturer and the latter for a generic approach. The choice of custom or generic components will affect the product specifications, pricing, manufacturing, life cycle, and marketing strategies of the product. The World Health Organization is interested in making available to developing countries hearing aids that are inexpensive to purchase and maintain. The hearing aid presented in this article was developed with these specifications in mind together with additional contemporary features such as four channels with wide dynamic range compression, an adjustable compression rate for each channel, four comfort programs, an adaptive feedback manager, and full volume control. This digital hearing aid is fitted using a personal computer with minimal hardware requirements in intuitive three-step fitting software. A trimmer-adjusted version can be developed where human and material resources are scarce.

14 citations


Proceedings ArticleDOI
01 Dec 2010
TL;DR: A digital hearing aid chip designed for Mandarin user to enhance speech quality and intelligibility and reduce the power consumption of these algorithms through algorithmic and architecture optimization.
Abstract: This paper presents a digital hearing aid chip designed for Mandarin user to enhance speech quality and intelligibility. The hearing aid consists of an 18 subbands analysis and synthesis filter bank, insertion gain stage, and three channels wide dynamic range control for the new Mandarin-specific auditory compensation algorithm. A noise reduction block based on multiband spectral subtraction and enhanced entropy voice activity detection is also included to enhance quality. We reduce the power consumption of these algorithms through algorithmic and architecture optimization. In addition, for the data storage requirement, a low power SRAM that can operate at 0.6V and below is developed. Moreover, several strategies such as multi-clock domain, bypass mode, and voltage scaling are also adopted for power reduction. The chip measurement shows that the hearing aid consumes 314uW at 0.6V.

13 citations


Proceedings ArticleDOI
03 Aug 2010
TL;DR: The experimental result shows that the design totally reduces 96% computing complexity of the straightforward implementation without introducing audible difference in the output signals.
Abstract: Dynamic range compression (DRC) is an essential function in digital hearing aids, which have stringent power consumption constraints This work minimized the complexity of the DRC algorithm through adopting functional approximation for those involved nonlinear operations The tradeoff between computation reduction and approximation error was explored for reducing hardware cost without introducing audible artifact Moreover, finite wordlength analysis is conducted for the wordlength minimization and further complexity saving The experimental result shows that our design totally reduces 96% computing complexity of the straightforward implementation without introducing audible difference in the output signals Compared with other approximation methods, our design also requires lower hardware complexity

11 citations


Patent
14 Apr 2010
TL;DR: In this article, a multi-channel wide dynamic range compressing system based on an audition perception model is proposed, where audio digital signals x (n) are divided into K channels after passing through the analysis filter group, and the sound pressure level detecting module is used for detecting the specific gain value of each channel, while the multiplier can multiply the gain values of the channels with corresponding sub-band signals.
Abstract: 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.

10 citations


Patent
Qingyun Wang, Xin Wei, Ji Xi, Li Zhao, Cairong Zou 
07 Jul 2010
TL;DR: In this article, a sound source positioning method for a glasses type digital hearing aid is presented, which is based on a quaternionic microphone matrix model of the hearing aid, and a formula for carrying out sound source position on the model is deduced, and the time delay difference from a sound sources to a microphone in the formula is estimated by a self-adaptation characteristic value decomposition algorithm.
Abstract: The invention provides a sound source positioning method for a glasses type digital hearing aid. A sound source space position obtained according to the calculation of the sound source positioning method can be further utilized by other modules of the hearing aid, so as to carry out speech enhancement and noise suppression. A quaternionic microphone matrix model of the glasses type digital hearing aid is first established, a formula for carrying out sound source positioning on the model is deduced, and the time delay difference from a sound source to a microphone in the formula is estimated by a self-adaptation characteristic value decomposition algorithm. Under the environment of noise and resonance, the provided method has the characteristics of good robust performance, high positioning precision, small calculating amount and easy real-time realization.

5 citations


01 Jan 2010
TL;DR: Directional microphone when combined with noise reduction algorithm in a digital hearing aid, adds benefit in improving speech intelligibility for patients with moderately-severe sensory neural hearing loss whatever the duration of the hearing loss.
Abstract: Objective: We aims to detect the efficacy of directional microphone when combined with noise reduction algorithm in improving speech intelligibility in noisy environments for hearing impaired subjects. Materials and Methods: Twenty adult subjects with bilateral symmetrical sensory neural hearing loss of moderate to moderately severe degree were examined. Aided assessment was done in two settings using BTE digital hearing aid. First setting included: evaluation using noise reduction algorithm alone. Evaluation in the second setting was done using both noise reduction algorithm and directional microphone. Aided evaluation consisted of speech discrimination scores in quiet and speech in noise test in different noise scenarios. Just follow conversation test was done to detect the least signal to noise ratio a subject can tolerate in each of the two settings. The subjective impression of the patient was assessed by a modified questionnaire for hearing aid assessment. Results: Investigation showed statistically significant improvement of aided speech discrimination scores in noise in the second aided setting when speech was at zero degree azimuths and noise was at zero and 180 degrees azimuth. Conclusion: Directional microphone when combined with noise reduction algorithm in a digital hearing aid, adds benefit in improving speech intelligibility for patients with moderately-severe sensory neural hearing loss whatever the duration of the hearing loss.

Patent
14 Oct 2010
TL;DR: In this article, a method for controlling a digital hearing aid using a mobile communication terminal, and a mobile communications terminal and a hearing aid thereof are provided to supply a current state of a hearing aids to a user, thereby providing convenience to the user.
Abstract: PURPOSE: A method for controlling a digital hearing aid using a mobile communication terminal, and a mobile communication terminal and a digital hearing aid thereof are provided to supply a current state of a hearing aid to a user, thereby providing convenience to the user. CONSTITUTION: A mobile communication terminal(250) includes a wireless unit(201), an environment measuring unit, and a hearing aid controller. The wireless unit communicates with a wireless digital hearing aid(150). The environment measuring unit measures a surrounding circumstance signal. The hearing aid controller analyzes a measurement result of the surrounding circumstance signal. The hearing aid controller outputs information or a command for controlling the hearing aid by analyzing the measurement result.

Journal ArticleDOI
TL;DR: In this article, a digital behind-the-ear hearing aid based on standardized components coming from the very supply chain of the hearing aid manufacturers is proposed, and the developed hearing aids did not show lesser electroacoustic characteristics when compared to those acquired by the Government.
Abstract: The treatment of sensorineural hearing loss is based on hearing aids, also known as individual sound amplification devices. The hearing aids purchased by the Brazilian Government, aiming at fulfilling public policies, are based on dedicated components, which bring about benefits, but also render them expensive and may impair repair services after manufacture's warranty expires. Aim: to design digital behind-the-ear hearing aids built from standardized components coming from the very supply chain of these manufacturers. Study design: experimental. Materials and Methods: to identify the supply chain of these manufacturers, request samples and set up hearing aids in the laboratory. Results: The developed hearing aids did not show lesser electroacoustic characteristics when compared to those acquired by the Government, also being tested by the same reference international technical standard. Conclusion: It is possible to develop digital behind-the-ear hearing aids based on off-the-shelf components from hearing aid manufacturers' supply chain. Their advantages include low operational costs – for acquisition (with clear advantages for the Government) and service (advantage for the patient).

Proceedings Article
21 Jul 2010
TL;DR: Initial results indicate that using the device improves user's satisfaction in terms of ease of communication and aversiveness of sounds, and the devices under test also proved to be robust, passing all the designated tests.
Abstract: This paper discusses preliminary results on REAT institute's attempt in developing an alternative digital hearing aid targeted for rural usage. Following our previous publication on its design and acoustical performance, this paper reports on its 3-month trial at Chiang Mai University Hospital, both on engineering and user's satisfaction aspects. The patient outcome measurement protocol used is adopted from APHAB, the first time it is implemented in Thai language. Initial results indicate that using the device improves user's satisfaction in terms of ease of communication and aversiveness of sounds. In terms of engineering, the devices under test also proved to be robust, passing all the designated tests.

01 Jan 2010
TL;DR: In this paper, the authors analyzed the reduction of noise in a noise environment using 2, 3, 4 or 5 microphones in digital hearing aids and found that the improvement in performance was highest when three or four microphones were used.
Abstract: In this study, we analyzed the reduction of noise in a noise environment using 2, 3, 4 or 5 microphones in digital hearing aids. In order to be able to use this in actual digital hearing aids, we made the experiment microphone set similar to the behind-the-ear type (BTE) and then recorded the signal accordingly, with each situation. With the recorded signals, we reduced the noise in each signal by a noise reduction algorithm using multi-microphones. As a result, in the case of By comparing the SNR (Signal to Noise Ratio) and PESQ (Perceptual Evaluation of Speech) measurements, before and after the noise reduction, the results showed that the improvement in performance was highest when three or four microphones were used. Generally, when two or more microphones were used, we found that as the number of microphones increased there was an increase in performance.


Patent
19 May 2010
TL;DR: 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.
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.

Patent
08 Apr 2010
TL;DR: In this paper, the authors presented an in-the-ear digital hearing aid that can be controlled wirelessly and easily by anybody including an old person, which enables a user who has thick fingers to clearly hear sound depending on various listening environments by selecting a news listening parameter, a music listening parameter and a conversation listening parameter through a remoter controller.
Abstract: The purpose of the present invention is to provide an in-the-ear digital hearing aid that can be controlled wirelessly and easily by anybody including an old person. The digital hearing aid enables a user who has thick fingers to clearly hear sound depending on various listening environments by selecting a news-listening parameter, a music-listening parameter and a conversation-listening parameter through a remoter controller, wherein the conversation-listening parameter is for listening to a conversation in a noisy place.

Journal ArticleDOI
TL;DR: In this paper, the authors proposed a solution of digital hearing aid which can be independently controlled at each audiogram test frequency and provide superb voice quality with less noise due to continuous variable gain characteristics in a wide dynamic range.
Abstract: Recently, the digital and LSI based miniaturization technologies drastically have changed the hearing aids from the conventional single function of amplification to multiple functions. However, since the conventional hearing aids mostly control the flat gain at the center of a frequency band divided every octave to recover the auditory loss of users, so that the discontinuous gain tends to deteriorate the hearing quality due to the unnatural saw-teeth figured error. This paper proposes a solution of digital hearing aid which can be independently controlled at each audiogram test frequency and provide superb voice quality with less noise due to continuous variable gain characteristics in a wide dynamic range. The simulation results show the outstanding effect of the proposed scheme in the voice quality improvement which potential hearing aid users have considered to be a key to satisfy them.