• Clinical and Experimental Otorhinolaryngology
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• v.11 no.4
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• pp.267-274
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• 2018

Objectives. Two main digital signal processing technologies inside the modern hearing aid to provide the best conditions for hearing aid users are directionality (DIR) and digital noise reduction (DNR) algorithms. There are various possible settings for these algorithms. The present study evaluates the effects of various DIR and DNR conditions (both separately and in combination) on listening comfort among hearing aid users. Methods. In 18 participants who received hearing aid fitting services from the Rehabilitation School of Shahid Beheshti University of Medical Sciences regularly, we applied acceptable noise level (ANL) as our subjective measure of listening comfort. We evaluated both of these under six different hearing aid conditions: omnidirectional-baseline, omnidirectional-broadband DNR, omnidirectional-multichannel DNR, directional, directional-broadband DNR, and directional-multichannel DNR. Results. The ANL results ranged from -3 dB to 14 dB in all conditions. The results show, among all conditions, both the omnidirectional-baseline condition and the omnidirectional-broadband DNR condition are the worst conditions for listening in noise. The DIR always reduces the amount of noise that patients received during testing. The DNR algorithm does not improve listening in noise significantly when compared with the DIR algorithms. Although both DNR and DIR algorithms yielded a lower ANL, the DIR algorithm was more effective than the DNR. Conclusion. The DIR and DNR technologies provide listening comfort in the presence of noise. Thus, user benefit depends on how the digital signal processing settings inside the hearing aid are adjusted.

• Journal of Audiology & Otology
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• v.24 no.2
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• pp.91-98
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• 2020

Background and Objectives: This study aimed to compare functional hearing with the use of a personal sound amplification product (PSAP) or a basic hearing aid (HA) among sensorineural hearing impaired listeners. Subjects and Methods: Nineteen participants with mild-to-moderate sensorineural hearing loss (SNHL) (26-55 dB HL; pure-tone average, 0.5-4 kHz) were prospectively included. No participants had prior experience with HAs or PSAPs. Audiograms, speech intelligibility in both quiet and noisy environments, speech quality, and preference were assessed in three different listening conditions: unaided, with the HA, and with the PSAP. Results: The use of PSAP was associated with significant improvement in pure-tone thresholds at 1, 2, and 4 kHz compared to the unaided condition (all p<0.01). In the quiet environment, speech intelligibility was significantly improved after wearing a PSAP compared to the unaided condition (p<0.001), and this improvement was better than the result obtained with the HA. The PSAP also demonstrated similar improvement in the most comfortable levels compared to those obtained with the HA (p<0.05). However, there was no significant improvement of speech intelligibility in a noisy environment when wearing the PSAP (p=0.160). There was no significant difference in the reported speech quality produced by either device or in participant preference for the PSAP or HA. Conclusions: The current result suggests that PSAPs provide considerable benefits to speech intelligibility in a quiet environment and can be a good alternative to compensate for mild-to-moderate SNHL.

• Korean Journal of Audiology
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• v.24 no.2
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• pp.91-98
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• 2020

Background and Objectives: This study aimed to compare functional hearing with the use of a personal sound amplification product (PSAP) or a basic hearing aid (HA) among sensorineural hearing impaired listeners. Subjects and Methods: Nineteen participants with mild-to-moderate sensorineural hearing loss (SNHL) (26-55 dB HL; pure-tone average, 0.5-4 kHz) were prospectively included. No participants had prior experience with HAs or PSAPs. Audiograms, speech intelligibility in both quiet and noisy environments, speech quality, and preference were assessed in three different listening conditions: unaided, with the HA, and with the PSAP. Results: The use of PSAP was associated with significant improvement in pure-tone thresholds at 1, 2, and 4 kHz compared to the unaided condition (all p<0.01). In the quiet environment, speech intelligibility was significantly improved after wearing a PSAP compared to the unaided condition (p<0.001), and this improvement was better than the result obtained with the HA. The PSAP also demonstrated similar improvement in the most comfortable levels compared to those obtained with the HA (p<0.05). However, there was no significant improvement of speech intelligibility in a noisy environment when wearing the PSAP (p=0.160). There was no significant difference in the reported speech quality produced by either device or in participant preference for the PSAP or HA. Conclusions: The current result suggests that PSAPs provide considerable benefits to speech intelligibility in a quiet environment and can be a good alternative to compensate for mild-to-moderate SNHL.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.9 no.2
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• pp.205-212
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• 2016

In this paper, we implement the development platform of multichannel digital hearing aid algorithm using Simulink provided by Matlab. The digital hearing aids are considered medical devices designed to compensate for hearing loss, they need to be correctly selected, to help a person who has difficulty in hearing. The development platform that implemented in this paper, includes WOLA filterbank for analysis/synthesis of input signal, Wide dynamic range compression for hearing loss compensation and adaptive filter for feedback cancellation. Using the development platform, algorithm parameters for each block can be set depending on the hearing aid user. Thus it is possible to test the algorithm before the machine language. As a result, the time for algorithm development can be saved and performance and computational complexity can be optimized.

• Phonetics and Speech Sciences
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• v.6 no.2
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• pp.55-64
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• 2014

This study measured formant bandwidths of profound hearing impaired children and examined the characteristics of their articulation. For this study, 10 cochlear implanted children(CI), 10 hearing aid children(HA) and 10 normal hearing children(NH) were asked to read 7 Korean vowels(/ɑ, ʌ, o, u, ɯ, i, ɛ/). The subjects' readings were recorded by NasalView and analyzed by Praat. The analysis of the formant bandwidths explains the degree of vocal fold opening and the characteristics of radiation. Through the analysis of formant bandwidth, we can see that the hearing-impaired maintain vocal fold tension when they speak high vowels and characteristics of radiation. Narrower B1 means better maintain vocal fold tension, wider B2 means more front and wider B3 means the rounder lips. CI's B1 was widest and NH's was narrowest. And females' B1 was wider than males'. Among vowels, B1 of /a/ was widest, and B1 of /i/ was narrowest. In the case of B2, HA and NH's B2 was wider than CI's. Females' B2 was wider than males'. And B2 of /i/ was widest, and B2 of /ʌ/ was narrowest. In the case of B3, NH's was widest, and CI's was narrowest. Males' was wider than females'. Among vowels, B3 of /o/ was widest, and B3 of /ɛ/ was narrowest. As a result, first, through the analysis of B1, we can find that NH and males could better maintain vocal fold tension than the hearing-impaired or females, and all children articulate /i/ with vocal fold tension than other vowels. Second, through the analysis of B2, NH and HA articulate vowels with the weaker rounded than CI does. And females articulate vowels with the weaker rounded than males do. Third, through the analysis of B3, NH articulate vowels with the rounder than HA or CI do, and males articulate vowels with the rounder than females do. Through the results, we can expect that the analysis of formant bandwidth will be applied to the therapy of articulation for the hearing-impaired with hearing aids or cochlear implant.

• Korean Journal of Audiology
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• v.25 no.2
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• pp.72-79
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• 2021

Background and Objectives: The speech-in-noise test is typically performed using an audiometer. The results of the digit-in-noise recognition (DIN) test may be influenced by the flat frequency response of free-field audiometry and frequency of the hearing aid fit based on fitting rationale. This study aims to investigate the DIN test in unaided and aided conditions. Subjects and Methods: Thirty four adults with moderate and severe sensorineural hearing loss (SNHL) participated in the study. The signal-to-noise ratio (SNR) for 50% of the DIN test was obtained in the following two conditions: 1) the unaided condition, performed using an audiometer in a free field; and 2) aided condition, performed using a hearing aid with an unvented individual earmold that was fitted based on NAL-NL2. Results: There was a statistically significant elevation in the mean SNR for the severe SNHL group in both test conditions when compared with that of the moderate SNHL group. In both groups, the SNR for the aided condition was significantly lower than that of the unaided condition. Conclusions: Speech recognition in hearing-impaired patients can be realized by fitting hearing aids based on evidence-based fitting rationale rather than by measuring it using free-field audiometry measurement that is utilized in a routine clinic setup.

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.457-460
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• 2005

When a hearing aid' s user is listening through telephone or cellular phone, he/she usually suffers from severe electrical magnetic interference noise. It is because hearing aids amplify voice signal as well as background noise. A telecoil, an induction coil, is a possible solution for the problem. Because a telecoil has the characteristic of high pass filter, it has some problem of resulting increased high frequency noise. For solving this problem, we can use a capacitor connected with the telecoil in parallel. According to capacitance, receiving signal quality may change. In this paper, proper capacitor values for the best sound quality are investigated by experimental work.

• The Journal of the Acoustical Society of Korea
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• v.18 no.4
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• pp.38-46
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• 1999

In this paper, two methods were proposed to improve the performance of adaptive feedback cancellation in hearing aids. One is “Orthogonality principle acoustic feedback cancellation method(Orthogonality principle method)” to track optimal solution with monitoring the instantaneous gradient, the other is a method using the CLMS algorithm(CLMS method). In many simulation conditions, adaptive feedback cancellation method proposed in this paper was much better than Greenberg method by Sum-method LMS algorithm which is known the most excellent method by now in case of system mismatch, SNR and segmental SMR. Also. Orthogonality principle method is as good as CLMS method in terms of adaptive feedback cancellation in many simulation conditions.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.1C
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• pp.96-102
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• 2010

This paper proposes a new adaptive algorithm to cancel the acoustic feedback and noise signals in the digital hearing aids. The proposed algorithm improves its convergence performances by canceling the speech signal from the residual signal using two microphones. The feedback canceller firstly cancels the feedback signal among the mic signal, and then it is reduced the noise using the beamforming method. To verify the performances of the proposed algorithm, the simulations were carried out for some cases. As the results of simulations, it was proved that the feedback canceller and the noise canceller advance about 14.43 dB for SFR, 10.19 dB for SNR respectively during speech, in the case of using the new algorithm.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.5
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• pp.887-892
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• 2008

This paper is on a new adaptive algorithm which can cancel the acoustic feedback signals in the digital hearing aids. The proposed algorithm uses the normalized LMS algorithm with decorrelators. By doing so, it can be reduced the autocorrelation for the voice signals. To analyze the convergence characteristics of the proposed algorithm, the simulations were carried out about various input signals. And we had compared the performances of convergence for this algorithm with the ones for the NLMS algorithm. As the results of simulations, it is proved that the feedback canceller adopting this algorithm shows about 5-10 dB more high SNR than the NLMS algorithm for the colored inputs.