• 말소리와 음성과학
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• 제13권1호
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• pp.65-72
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• 2021

본 연구의 목적은 말소리장애 아동이 산출하는 이중모음의 특성을 파악하여 평가 및 중재에 활용할 수 있는 기초 자료를 마련하는 것이다. 현재까지 말소리장애 아동의 이중모음 산출 특성에 관한 음향학적 연구는 미비하였다. 이에 말소리장애 아동과 일반 아동을 대상으로 집단 간 이중모음 산출 특성의 차이를 파악하고자 하였다. 이를 위해 각 10명의 만 4-5세 말소리장애와 일반 아동을 대상으로, 무의미 2음절 '이중모음+다'를 모방하도록 하였다. 산출된 이중모음의 활음 구간 내 제1, 2 포먼트 기울기, 포먼트 변화량, 활음 지속시간을 Praat(version 6.1.16)을 이용해 분석하였다. 연구 결과, 두 집단 간 /유/의 F1 기울기에 집단 간 유의한 차이가 있었다. 또한, 말소리장애 아동이 일반 아동에 비해 전반적으로 작은 포먼트 변화량과 더 짧은 활음 지속시간을 보였다. 유의한 포먼트 변화량의 집단 간 차이는 /유, 예/의 F1과 /야, 예/의 F2에서 나타났으며, 유의한 활음 지속시간의 차이는 /유, 예/에서 나타났다. 본 연구의 결과는 말소리장애 아동이 이중모음을 조음하는 범위가 일반 아동보다 상대적으로 작아 그만큼 조음하는데 걸리는 시간이 줄었음을 보여준다. 이러한 점은 말소리장애 아동의 이중모음에 관한 평가와 중재를 할 때 말소리장애 아동의 조음 범위를 고려해야 하며, 이에 음향학적 도구를 활용하는 것이 필요함을 뒷받침한다.

• 대한청각학회지
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• 제24권2호
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• pp.71-78
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• 2020

Background and Objectives: Currently limited information is available on speech stimuli processing at the subcortical level in the recipients of cochlear implant (CI). Speech processing in the brainstem level is measured using speech-auditory brainstem response (S-ABR). The purpose of the present study was to measure the S-ABR components in the sound-field presentation in CI recipients, and compare with normal hearing (NH) children. Subjects and Methods: In this descriptive-analytical study, participants were divided in two groups: patients with CIs; and NH group. The CI group consisted of 20 prelingual hearing impairment children (mean age=8.90±0.79 years), with ipsilateral CIs (right side). The control group consisted of 20 healthy NH children, with comparable age and sex distribution. The S-ABR was evoked by the 40-ms synthesized /da/ syllable stimulus that was indicated in the sound-field presentation. Results: Sound-field S-ABR measured in the CI recipients indicated statistically significant delayed latencies, than in the NH group. In addition, these results demonstrated that the frequency following response peak amplitude was significantly higher in CI recipients, than in the NH counterparts (p<0.05). Finally, the neural phase locking were significantly lower in CI recipients (p<0.05). Conclusions: The findings of sound-field S-ABR demonstrated that CI recipients have neural encoding deficits in temporal and spectral domains at the brainstem level; therefore, the sound-field S-ABR can be considered an efficient clinical procedure to assess the speech process in CI recipients.

• Journal of Audiology & Otology
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• 제24권2호
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• pp.71-78
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• 2020

Background and Objectives: Currently limited information is available on speech stimuli processing at the subcortical level in the recipients of cochlear implant (CI). Speech processing in the brainstem level is measured using speech-auditory brainstem response (S-ABR). The purpose of the present study was to measure the S-ABR components in the sound-field presentation in CI recipients, and compare with normal hearing (NH) children. Subjects and Methods: In this descriptive-analytical study, participants were divided in two groups: patients with CIs; and NH group. The CI group consisted of 20 prelingual hearing impairment children (mean age=8.90±0.79 years), with ipsilateral CIs (right side). The control group consisted of 20 healthy NH children, with comparable age and sex distribution. The S-ABR was evoked by the 40-ms synthesized /da/ syllable stimulus that was indicated in the sound-field presentation. Results: Sound-field S-ABR measured in the CI recipients indicated statistically significant delayed latencies, than in the NH group. In addition, these results demonstrated that the frequency following response peak amplitude was significantly higher in CI recipients, than in the NH counterparts (p<0.05). Finally, the neural phase locking were significantly lower in CI recipients (p<0.05). Conclusions: The findings of sound-field S-ABR demonstrated that CI recipients have neural encoding deficits in temporal and spectral domains at the brainstem level; therefore, the sound-field S-ABR can be considered an efficient clinical procedure to assess the speech process in CI recipients.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2007년도 춘계학술대회논문집
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• pp.649-654
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• 2007

Eavesdropping prevention performance is evaluated on a commercial window shaker, which is used to prevent a glass window from eavesdropping. Speech transmission index (STI) is introduced in order to estimate quantitatively the speech intelligibility of the sound detected on the glass window. Objective test by IEC standard using modulation transfer function (MTF) is performed to determine STI. Using Maximum Length Sequency (MLS) signal as a sound source, MTF is measured by accelerometers and laser doppler vibrometer. STI under different level of disturbing wave are compared to confirm the disturbing effect on the speech intelligibility.

• 제어로봇시스템학회논문지
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• 제10권9호
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• pp.763-770
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• 2004

The important issue of emotion recognition from speech is a feature extracting and pattern classification. Features should involve essential information for classifying the emotions. Feature selection is needed to decompose the components of speech and analyze the relation between features and emotions. Specially, a pitch of speech components includes much information for emotion. Accordingly, this paper searches the relation of emotion to features such as the sound loudness, pitch, etc. and classifies the emotions by using the statistic of the collecting data. This paper deals with the method of recognizing emotion from the sound. The most important emotional component of sound is a tone. Also, the inference ability of a brain takes part in the emotion recognition. This paper finds empirically the emotional components from the speech and experiment on the emotion recognition. This paper also proposes the recognition method using these emotional components and the transition probability.

• 대한음성학회:학술대회논문집
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• 대한음성학회 2006년도 춘계 학술대회 발표논문집
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• pp.127-130
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• 2006

This paper suggests an algorithm that can estimate the direction of the sound source with three microphones arranged on a circle. The algorithm is robust to microphones' gains because it uses only the time differences between microphones. To make this possible, a cost function which normalizes the microphone's gains is utilized and a procedure to detect the rough position of the sound source is also proposed. Through our experiments, we obtained significant performance improvement compared with the energy-based localizer.

• 대한전자공학회논문지
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• 제26권9호
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• pp.1436-1443
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• 1989

In this paper, we propose a maximum-likelihood estimation(MLE) method to obtain the location and the amplitude of the pulses in MPE( multi-pulse excitation)-LPC speech synthesis using multi-pulses as excitation source. This MLE method computes the value maximizing the likelihood function with respect to unknown parameters(amplitude and position of the pulses) for the observed data sequence. Thus in the case of overlapped pulses, the method is equivalent to Ozawa's crosscorrelation method, resulting in equal amount of computation and sound quality with the cross-correlation method. We show by computer simulation: the multi-pulses obtained by MLE method are(1) pseudo-periodic in pitch in the case of voicde sound, (2) the pulses are random for unvoiced sound, (3) the pulses change from random to periodic in the interval where the original speech signal changes from unvoiced to voiced. Short time power specta of original speech and syunthesized speech obtained by using multi-pulses as excitation source are quite similar to each other at the formants.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1994년도 FIFTH WESTERN PACIFIC REGIONAL ACOUSTICS CONFERENCE SEOUL KOREA
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• pp.991-996
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• 1994

Surrounding noise often affects the performance of speech recognition system when it is used in office or home. Especially situation is more serious when colored and nonstational noise such as an sound from television or other audio equipment is introduced. The authors proposed a voice control system for television set using an adaptive noise canceler, and it works well even is sound of television set has comparable level of speech. In this paper, a new front-end of speech recognition is introduced for the voice control system. This font-end utilizes a simplified masking model to reduce the effect of residual noise. According to experimental results, 90% correct recognition is achieved even if the level of television sound is almost 15dB higher than one of speech.

• 제어로봇시스템학회논문지
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• 제16권12호
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• pp.1143-1149
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• 2010

Because the noise is the main cause for decreasing the performance at speech recognition, the place or environment is very important in speech recognition. To improve the speech recognition performance in the real situations where various extraneous noises are abundant, a novel combination of FIR and Wiener filters is proposed and experimented. The combination resulted in improved accuracy and reduced processing time, enabling fast analysis and response in emergency situations. Usually, there are many dangerous situations in our city life, so for the smarter city it is necessary to detect many types of sound in various environment. Therefore this paper is about how to detect many types of sound in real city, especially on CCTV. This paper is for implementing the smarter city by detecting many types of sounds and filtering one of the emergency sound in this sound stream. And then it can be possible to handle with the emergency or dangerous situation.

• 음성과학
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• 제15권3호
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• pp.17-28
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• 2008

In many practical applications of robots, finding the location of an incoming sound is an important issue for the development of efficient human robot interface. Most sound source localization algorithms make use of only those microphones that are acoustically visible from the sound source or do not take into account the effect of sound diffraction, thereby degrading the sound source localization performance. This paper proposes a new sound source localization method that can utilize those microphones that are acoustically shadowed from the sound source. The experiment results show that use of the acoustically shadowed microphones, which receive higher signal-to-noise ratio signals than the others and are closer to the sound source, improves the performance of sound source localization.