• Journal of Biomedical Engineering Research
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• v.30 no.6
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• pp.529-534
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• 2009

One of the characteristics of the hearing impaired is that their frequency selectivity is poorer than that of the normal hearing. To compensate this, formant enhancement algorithms and spectral contrast enhancement algorithms have been developed. However in some cases, these algorithms fail to improve the frequency selectivity of the hearing impaired. One of the reasons is the acoustic masking among enhanced formants. In this study, we tried to enhance the formants based on the individual masking characteristic of each subject. The masking characteristic used in this study was minimum level difference (MLD) between the first formant to the second formant while acoustic masking was occurred. If the level difference between the two formants in each frame is larger than the MLD, the gain of the first formant was decreased to reduce the acoustic masking that occurred among formants. As a result of the speech discrimination test, using formant enhanced speeches, speech discrimination score (SDS) of the speeches having differently enhanced formants was significantly superior to SDS of the speeches having equally enhanced formants. It means that suppression of the acoustic masking among formants improve frequency selectivity of the hearing impaired.

• Journal of rehabilitation welfare engineering & assistive technology
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• v.1 no.1
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• pp.21-28
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• 2007

In most of hearing aids, amplification algorithms are used to compensate hearing loss, noise and feedback reduction algorithms are used and to increase the perception of speeches contrast enhancement algorithms are used. However, acoustic masking effect is occurred between formants if contrast is enhanced excessively. To confirm the masking effect in speeches, the experiment are composed of 6 tests; test pure tone test, speech reception test, word recognition test, pure tone masking test, formant pure tone masking test and speech masking test, and for objective evaluation, LLR is introduced. As a result of normal hearing subjects and hearing impaired subjects, more making is occurred in hearing impaired subjects than normal hearing subjects when using pure tone, and in the speech masking test, speech reception is also lower in hearing impaired subjects than in normal hearing subjects. This means that acoustic masking effect rather than distortion influences speech perception. So it is required to check the characteristics of masking effect before wearing a hearing aid and to apply this characteristics to fitting curve.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.56 no.3
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• pp.629-631
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• 2007

In hands-free full-duplex communication systems, acoustic signals picked up by the microphones can be mixed with echo signals as well as noises, which may result in poor performance of the corresponding communication system. Also, the system performance may decrease further if the reverberation occurs since it is harder to estimate the impulse response of the demixing system. For blind source separation (BSS) in such cases, a time-frequency masking approach can be employed to separate undesired echo signals and noises, but, permutation ambiguities also should be solved for the echo cancellation. In this paper, we propose a new acoustic echo cancellation (AEC) approach utilizing the time-frequency masking and higher-order statistics, whereby a desired signal selection, based on coherence and third-order statistics (i.e., kurtosis), is introduced along with output signal normalization. Simulation results demonstrate that the proposed approach yields better echo and noise cancellation performances than the conventional AEC approaches.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.45 no.5
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• pp.13-20
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• 2008

Although digital hearing aid algorithms have been developed to compensate hearing loss and to help hearing impaired people to communicate with others, digital hearing aid user still complain about difficulty of hearing the speech. The reason could be the quality of speech through digital hearing aid is insufficient to understand the speech caused by feedback, residual noise and etc. And another thing is masking effect among formants that makes sound quality low. In this study, we measured the masking characteristics of normal listeners and hearing impaired listeners having presbyacusis to confirm masking effect in speech itself. The experiment is composed of 5 tests; pure tone test, speech reception threshold (SRT) test, word recognition score (WRS) test, puretone masking test and speech masking test. In speech masking test, there are 25 speeches in each speech set. And log likelihood ratio (LLR) is introduced to evaluate the distortion of each speech objectively. As a result, the speech perception became lower by increasing the quantity of formant enhancement. And each enhanced speech in a speech set has statistically similar LLR, however speech perception is not. It means that acoustic masking effect rather than distortion influences speech perception. In actuality, according to the result of frequency analysis of the speech that people can not answer correctly, level difference between first formant and second formant is about 35dB, and it is similar to result of pure tone masking test(normal hearing subject:36.36dB, hearing impaired subject:32.86dB). Characteristics of masking effect is not similar between normal listeners and hearing impaired listeners. So it is required to check the characteristics of masking effect before wearing a hearing aid and to apply this characteristics to fitting.

• The Journal of the Acoustical Society of Korea
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• v.31 no.6
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• pp.367-372
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• 2012

This paper introduces noise masking algorithm for 64 channel digital hearing aid. 125 Hz spectral resolution is maintained for 64 channels from 125 Hz to 8000 Hz. The same spectral masking processing effects as the cochlea are considered and applied for the present hearing aid noise reduction processing algorithm. Theoretical algorithm has been ported into assembler language program software and been embedded into a DSP IC chip for the digital hearing aid. Some parts of noise masking software program code are explained, and the results of the real-time noise reduction are verified by electro-acoustic measurements.

• Speech Sciences
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• v.6
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• pp.197-217
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• 1999

In a noisy class, the acoustic-phonetic features of the teacher and the perceptual features of learners are changed comparison with a quiet environment. Acoustical analyses were carried out on a set of French monosyllables consisting of 17 consonants and three vowel /a, e, i/, produced by 1 male speaker talking in quiet and in 50, 60 and 70 dB SPL of masking noise on headphone. The results of the acoustic analyses showed consistent differences in energy and formant center frequency amplitude of consonants and vowels, $F_1$ frequency of vowel and duration of voiceless stops suggesting the increase of vocal effort. The perceptual experiments in which 18 undergraduate female students learning French served as the subjects, were conducted in quiet and in 50, 60 dB of masking noise. The identification scores on consonants were higher in Lombard speech than in normal speech, suggesting that the speaker's vocal effort is useful to overcome the masking effect of noise. And, with increased noise level, the perceptual response to the French consonants given had a tendency to be complex and the subjective reaction score on the noise using the vocabulary representative of 'unpleasant' sensation to be higher. And, in the point of view on the L2(second language) acquisition, the influence of L1 (first language) on L2 examined in the perceptual result supports the interference theory.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.18 no.2
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• pp.231-237
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• 2008

With the advent of sound field simulator, many sound fields have been reproduced by obtaining the impulse responses of specific acoustic spaces like famous concert hall, opera house. This sound field reproduction has been done by the linear convolution operation between the sound input signal and the impulse response of certain acoustic space. However, the conventional finite impulse response based linear convolution operation always makes real-time implementation of sound field generator impossible due to the large amount of computational burden. This paper introduces the fast convolution method using perceptual redundancy in the processed signals, input audio signal and room impulse response. Temporal and spectral real-time masking blocks are implemented in the proposed convolution structure. It reduces the computational burden of convolution methods for real-time implementation of a sound field generator. The conventional convolutions are compared with the proposed one in views of computational burden and sound quality. In the proposed method, a considerable reduction in the computational burden was realized with acceptable changes in sound quality.

• Journal of the Korean Institute of Electrical and Electronic Material Engineers
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• v.28 no.11
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• pp.748-751
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• 2015

Sound Masking System technology as by sound the same on all bands and artificially generates a constant sound shield People want to hear or recognize the people with the noise generated from the interior of the way. Prevent hearing or prevent recognition by using the technology to control the audible frequency band Continue to emit constant and uniform shielding sound audible frequency band Even the security content of speech (20 Hz~20 KHz). That interception laser eavesdropping, internal solicitations, during recording Or delay the decoding was a result of the effect of interference calculated Experience noise disturbance index is applied around the Stress Index is the average index is 10.16 was a luxury for the average index is then applied to the index 3.07 Noise is significantly lower stress level has improved noise conditions.

• The Journal of the Korea institute of electronic communication sciences
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• v.5 no.4
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• pp.352-356
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• 2010

In this study, it is discussed how the directional property of the sound lateralization is influenced by proceeding or succeeding tone. The acoustic source applied here is a reference sound which has 0.5 msec interaural time difference(ITD). Based on this reference sound, interfering sounds with five levels of magnitude are applied to the subjects with four kinds of inter-stimuli time intervals(ISI). The interfering sounds are also added as two different types, proceeding tone and succeeding tone. Additionally, in order to investigate a frequency influence, the reference sound and the interfering sounds are generated by using 2kHz, 4 kHz and a white noise. As a result, the influence on lateralization by proceeding tone is lager than that by succeeding tone. It can consider this result as the effect of temporal masking on lateralization. Moreover, there are small differences of masking effect on lateralization by combinations of pure tone. This result shows that the dependency of frequency domain between reference sound and interfering sound is small on the sound lateralization.

• Journal of the Korean Society of Fisheries and Ocean Technology
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• v.48 no.4
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• pp.495-503
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• 2012

An acoustic method for discriminating plankton from fish, in Lake Dom Helvecio of Brazil, is developed. The flow of data from this method is comprised of time varied threshold (TVT), dilation filter, bitmap and mask functions. The TVT can, of itself, precisely explain how to select an appropriate value. The final results of the echogram, which only shows plankton by masking fish signals, is used to examine the acoustic density of plankton by depth and time. The results indicate that the acoustic density of the plankton is at a depth of between 5m to 15m, its density is especially high at 10m to 15m. The results of the acoustic density of plankton by time indicate that May 7 is higher in density than May 8. Future study plans include the use of net samples, environmental datasets to identify the abundance and ecology described by the Chaoborus spp. from other species.