• Journal of the Korean Institute of Telematics and Electronics
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• v.20 no.6
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• pp.58-61
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• 1983

This is a Loudness Control Circuit in an audio amplifier controlled by feedback type volume control variable resistors. This circuit consists of Bridged Twin T network and a ordinary variable resistor. The variably resistor acts not only as a volume control by varying feedback qupntity, but also as Loudness Control through the characteristics variation by Sound Level. This new Loudness Control Circuit showed ideal compensation characteristics that agree computer simulation and measured datas.

• The Journal of the Acoustical Society of Korea
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• v.19 no.8
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• pp.12-21
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• 2000

In this paper, we propose a novel hearing aid algorithm for sensorinural hearing loss restoration with multi-channel(band) dynamic range compression and psychoacoustics. In this way, we can present a normal perception condition to the impaired listener. The proposed algorithm make loudness scaling function achieve proper loudness level, and analysis masking property for the signal will be perceived to impaired listener, and then, restore normal spectral contrast using SMR(signal-to-masking ratio) defined by distance between the level of each frequency and masking threshold.

• 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.

• 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 Biomedical Engineering Research
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• v.18 no.4
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• pp.457-466
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• 1997

We developed a new hearing aid telephone which helps the hearing impaired person to improve the listening ability of telephone speech. Recently, the hearing impaired person and the elderly who has hearing loss have been continuously increased and their desire for participating society as a producer has been increased also. So they strong1y want the hearing aid devices which make compensation fortheir handicap. The hearing aid telephone is one of the basic aid devices that helps the hearing impaired to communicate well with other poeple and to acquire easily useful information through the phone. We analyze the hearing ability of the hearing impaired, design the new model of the hearing aid telephone and test the telephone in three fields-electrical, word perception, user test. Our new tolephone has lour band pass filter channels and the center frequencies of these filters are 500, 1000, 2000, 3000Hz which are considered psychoacoustic factors and telephone line characteristics. The hearing impaired can adjust the total gain characteristics of receiving sound to his hearing ability by setting four volumes in the telelphone. This procedure is called fitting which is a very important factor for the hearing impaired to take meaning of speech. The total gain of this telephone is over 20dB from 250Hz to 3200Hz range. From the results of the tests we certify that our new model is better for the hearing impaired to understand the meaning or telephone speech than the old general models. The next step of developing the hearing aid telephone is to study about compressing sidetone and noise, dividing frequency bands, selecting hearing aid pattern and compensating psychoacoustic loudness. we expect that the advanced hearing aid telephone can be developed by the research about speech perception characteristics of the hearing impaired in engineering and clinical side.

• Journal of IKEEE
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• v.16 no.4
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• pp.395-404
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• 2012

In this paper, the perceived loudness metering algorithm based on ITU-R BS.1387-1 was investigated and implemented, and its performance was evaluated by applying to 23 pure tones and 9 digital audio samples. Error of the tone test results compared with ISO226:2003 was below 5%, and sample test results, in comparison with Moore's algorithm, showed deviation of less than 4.7% and correlation of 0.96. On the other hand, it was investigated how the implemented algorithm's performance was subject to auditory pitch scale. Its result showed that the algorithm with 37 auditory filters, through correcting a bias effect, has a good performance of less than 2% in comparison with the one with 109 auditory filters.