• Journal of the Institute of Electronics Engineers of Korea SC
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• v.42 no.4 s.304
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• pp.45-50
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• 2005

In digital hearing aid system, one of the main problems is acoustic feedback which is known as howling because of miniaturization md high-gain amplification. In this paper, we proposed a feedback cancellation algorithm for hearing aid using time delay and time-varying decorrelation filter. The proposed algorithm has a kind of adaptive filter structure, which is combined with time delay and time-varying decorrelation filter to improve feedback cancellation. An all pass filter was implemented as the time-varying decorrelation filter using low frequency modulator. From the result of computer simulation, it is verified that the proposed algorithm has good ability to cancel feedback.

• The Journal of the Acoustical Society of Korea
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• v.31 no.5
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• pp.273-279
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• 2012

In this thesis, a nonlinear compression fitting method was studied for each frequency channel of a 64 channel digital hearing aid. Unlike conventional fitting formula method done from the result of the hearing loss test, the present fitting method uses the auditory threshold of sound pressure measured near the tympanic membrane while ITE (In-The-Ear) hearing aid is fitted into the user's ear canal. Also, the spectral distribution of the voice sound pressure was used for realizing of output sound pressure compression curves against input sound pressure level. Theoretical research results of FFT-iFFT compression algorithm has been evaluated by experimental gain measurements at each different input sound pressure level 50 dB, 70 dB, 90 dB respectively.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.2 no.2
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• pp.37-43
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• 2009

In this study, multi-channel digital hearing aid algorithm for low power system is proposed. First, MDCT(Modified Discrete Cosine Transform) method converts time domain of input speech signal into frequency domain of it. Output signal from MDCT makes a group about each channel, and then each channel signal adjusts a gain using LCF(Loudness Compensation Function) table depending on hearing loss of an auditory person. Finally, compensation signal is composed by TDAC and IMDCT. Its all of process make progress 16-bit fixed-point operation. We use fast-MDCT instead of MDCT for reducing system complexity and previously computed tables instead of log computation for estimating a gain. This algorithm evaluate through computer simulation.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.190-195
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• 2005

The hearing aid chip described in this paper is an analog & digital mixed system. The design focuses on the$\mu$DSP core. This $\mu$DSP core includes internal time delays to two inputs from front and rear microphones. The paper consists of two parts; one is the composure and signal processing algorithm of digital hearing aids and the other is Verilog HDL codes for$\mu$DSP cores. All digital modules in the design were coded and synthesized by Verilog HDL codes which were verified by Mentor Graphics and Synopsis semiconductor chip design tools.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.7C
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• pp.413-420
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• 2011

This paper proposes a new adaptive algorithm to cancel the acoustic feedback and noise signals in the binaural hearing aids. The convergence performances of the proposed algorithm are improved by updating coefficients of the feedback canceller after the speech signal is cancelled from the residual signal with dual microphones. The feedback canceller firstly cancels the feedback signal from the microphone signal, and then the noise canceller reduces the noise by the beamforming method. To assure that binaural hearing aids converge stably, the left-sided hearing aid only is converged firstly, next the right-sided hearing aid only is converged. To verify performances of the proposed algorithm, simulations were carried out for a speech. As the results of simulations, it was proved that we can advance 14.43dB SFR(Signal to Feedback Ratio) on the average for the feedback canceller, 10.19dB SNR(Signal to Noise Ratio) improvement on the average for the noise canceller, in case that this algorithm is used.

• Journal of Biomedical Engineering Research
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• v.28 no.6
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• pp.732-743
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• 2007

Background noise is one of the major complaints of not only hearing impaired persons but also normal listeners. This paper describes the results of two experiments in which speech recognition performance was determined for listeners with normal hearing and sensorineural hearing loss in noise environment. First, we compared speech enhancement algorithms by evaluation speech recognition ability in various speech-to-noise ratios and types of noise. Next, speech enhancement algorithms by reducing background noise were presented and evaluated to improve speech intelligibility for sensorineural hearing impairment listeners. We tested three noise reduction methods using single-microphone, such as spectrum subtraction and companding, Wiener filter method, and maximum likelihood envelop estimation. Their responses in background noise were investigated and compared with those by the speech enhancement algorithm that presented in this paper. The methods improved speech recognition test score for the sensorineural hearing impaired listeners, but not for normal listeners. The results suggest the speech enhancement algorithm with the loudness compression can improve speech intelligibility for listeners with sensorineural hearing loss.

• Journal of Biomedical Engineering Research
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• v.18 no.4
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• pp.467-476
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• 1997

This paper presents a designing of a digital hearing aid processor (DHAP) chip being operated by a dedicated DSP core. The DHAP for hearing aid devices must be feasible within a size and power consumption required. Furthermore, it should be able to compensate for wide range of hearing losses and allow sufficient flexibility for the algorithm development. In this paper, a programmable 16-bit fixed-point DSP core is employed thor the designing of the DHAP. The designed DHAP performs a nonlinear loudness correction of 8 frequency bands based on audiometric measurements of impaired subjects. By employing a programmable DSP, the DHAP provides all the flexibility needed to implement audiological algorithms. In addition, the chip has low-power feature and $5, 500\times5000$$\mu$$m^2$ dimensions that fit for wearable hearing aids.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.41 no.1
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• pp.41-54
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• 2004

In this paper, we propose a new digital hearing aid circuit that compensates the impaired threshold level changing nonlinearly using a multi-band compensation technique. In the algorithm the hearing frequency range 8kHz is divided into 64 bands which is 125Hz resolution. Each band is controlled finely to compensate the hearing impaired proportional to personal ROM table. The multi-band is introduced using a FFT/IFFT Processor which makes to control in frequency domain. As a result, the proposed circuit is more efficient $15\%$ than a conventional ones such as FIR filter architecture in terms of the compensation gun and accuracy. The hardware size was reduced $65\%$ than a general FFT by pre-handling of the input data.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.59 no.6
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• pp.1120-1125
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• 2010

In the speech system, for example hearing aid as well as speech communication, speech quality is degraded by environmental noise. In this study, to enhance the speech quality which is degraded by environmental speech, we proposed an algorithm to reduce the noise and reinforce the speech. The minima controlled recursive averaging (MCRA) algorithm is used to estimate the noise spectrum and spectral weighting factor is used to reduce the noise. And partial masking effect which is one of the human hearing properties is introduced to reinforce the speech. Then we compared the waveform, spectrogram, Perceptual Evaluation of Speech Quality (PESQ) and segmental Signal to Noise Ratio (segSNR) between original speech, noisy speech, noise reduced speech and enhanced speech by proposed method. As a result, enhanced speech by proposed method is reinforced in high frequency which is degraded by noise, and PESQ, segSNR is enhanced. It means that the speech quality is enhanced.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.62 no.12
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• pp.1744-1749
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• 2013

Acoustic feedback is perceived as whistling or howling, which is a major complaint of hearing-aids users. Acoustic feedback cancellation is important in hearing-aids because acoustic feedback degrades performance of the hearing aid device by reducing maximum insertion gain. Adaptive systems for estimate acoustic feedback path and feedback suppression algorithms have been proposed in order to solve this problem. A typical feedback cancellation algorithm is LMS(least mean squares) because of its computational efficiency. However it has problem of convergence performance in high correlated input signal. In this paper, we propose a new variable step-size normalized LMS(least mean squares) algorithm using VAD(voice activity detection) to overcome the limitation of the LMS algorithm. The VAD algorithm is GSAP(global speech absence probability) and the feedback cancellation algorithm is normalized LMS. The proposed algorithm applies different step-size between voice and non-voice using VAD, for high stability, fast convergence speed and low misalignment when correlated inputs, such as speech. The result of simulation with white noise mixed speech signal, the proposed algorithm shows high performance then traditional algorithm in terms of stability, convergence speed and misalignment.