• Phonetics and Speech Sciences
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• v.9 no.4
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• pp.59-67
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• 2017

Regardless of their intention, listeners tend to assess speakers' personalities based on the sounds of the speech they hear. Assessment criteria, however, have not been fully investigated to indicate whether there is any relationship between the acoustic cue of produced speech sounds and perceived personality impression. If properly investigated, the potential relationship between these two will provide crucial insights on the aspects of human communications and further on human-computer interaction. Since human communications have distinctive characteristics of simultaneity and complexity, this investigation would be the identification of minimum essential factors among the sounds of speech and perceived personality impression. The purpose of this study, therefore, is to identify significant associations between the speech sounds and perceived personality impression of speaker by the listeners. Twenty eight subjects participated in the experiment and eight acoustic parameters were extracted by using Praat from the recorded sounds of the speech. The subjects also completed the Neo-five Factor Inventory test so that their personality traits could be measured. The results of the experiment show that four major factors(duration average, pitch difference value, pitch average and intensity average) play crucial roles in defining the significant relationship.

• Proceedings of the KOSOMBE Conference
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• v.1997 no.11
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• pp.67-70
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• 1997

In this paper, we evaluated the character of speech of hearing impaired person (HIP) who acquire his hearing loss after the youth. It is usually observed that severe HIP decreased not only speech perception but also vocalization. so there is a need for sensitive and quantitative measures or the assesment of the speech of the HIP to serve both diagnostic and prognosic purposes, 7 HIP and 12 normal hearing person(NHP) were studied with pure tone test and speaking test using word/sentence table which consists of vowel(a:), mono and two syllables and a sentence. we analyzed formant frequency, pitch, sound intensity, speech duration of HIP and NHP speech. According to the results, in the HIP's speech we find that formant frequency was shifted, first-formant prominence was reduced, the dynamic range of sound intensity was decreased, speech duration was prolonged. In the next, we expect the correlation between hearing and speech character of HIP is cleared through analysis of more acoustic parameters and precise selection of HIP group.

• Speech Sciences
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• v.12 no.3
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• pp.125-136
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• 2005

Automatic speech recognition is much more difficult in real world. Speech recognition according to SIR (Signal to Interface Ratio) is difficult in situations in which noise of surrounding environment and multi-speaker exists. Therefore, study on main speaker's voice extractions a very important field in speech signal processing in binaural sound. In this paper, we used directional filter and harmonic filter among other existing methods to extract the main speaker's information in binaural sound. The main speaker's voice was extracted using directional filter, and other remaining speaker's information was removed using harmonic filter through main speaker's pitch detection. As a result, voice of the main speaker was enhanced.

• Proceedings of the Acoustical Society of Korea Conference
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• 1991.06a
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• pp.7-10
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• 1991

In a text-to-speech system, sound units (phonemes, words, or phrases, etc.) can be concatenated together to produce required utterance. The quality of the resulting speech is dependent on factors including the phonological/prosodic contour, the quality of basic concatenation units, and how well the units join together. Thus although the quality of each basic sound unit is high, if occur the discontinuity in the concatenation part then the quality of synthesis speech is decrease. To solve this problem, a smoothing operation should be carried out in concatenation parts. But a major problem is that, as yet, no method of parameter smoothing is available for joining the segment together. Thus in this paper, we proposed a new aigorithm that smoothing the unnatural discountinuous parts which can be occured in speech waveform editing. This algorithm used the exponential smoothing method.

• 제어로봇시스템학회:학술대회논문집
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• 2003.10a
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• pp.633-638
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• 2003

In this paper, we propose a humanoid active audition system which detects the direction of sound and performs speech recognition using just three microphones. Compared with previous researches, this system which has simpler algorithm, fewer microphones and better amplifier shows better performance. In order to verify our system's performance, we install the proposed active audition system to the home service robot, called Hombot II, which has been developed at the KIST (Korea Institute of Science and Technology), thus we confirm excellent performance by experimental results

• Phonetics and Speech Sciences
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• v.7 no.4
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• pp.123-130
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• 2015

Given a recent finding showing that female speakers of South Kyungsang Korean is undergoing a sound change of the lexical pitch accent, this study tested whether the change is also reflected for male speech. This study compared F0 scaling and timing properties of accent words produced by younger female and male speakers of South Kyungsang Korean. The results indicated clear gender-related differences, showing more distinct acoustic properties across the accent words for male production compared to females. Despite the better distinction, however, younger male speakers showed peak delay where the F0 peaks are located further to the right compared to conservative speakers' production. Therefore, it might be suggested that younger male speakers' accent productions are in between conservative and innovative phonetic forms.

• Journal of KIISE:Computing Practices and Letters
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• v.8 no.3
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• pp.289-294
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• 2002

A valid speech-sound block can be classified to provide important information for speech recognition. The classification of the speech-sound block comes from the MRA(multi-resolution analysis) property of the DWT(discrete wavelet transform), which is used to reduce the computational time for the pre-processing of speech recognition. The merging algorithm is proposed to extract valid speech-sounds in terms of position and frequency range. It needs some numerical methods for an adaptive DWT implementation and performs unvoiced/voiced classification and denoising. Since the merging algorithm can decide the processing parameters relating to voices only and is independent of system noises, it is useful for extracting valid speech-sounds. The merging algorithm has an adaptive feature for arbitrary system noises and an excellent denoising SNR(signal-to-nolle ratio).

• 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.

• Proceedings of the Acoustical Society of Korea Conference
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• 1985.10a
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• pp.68-71
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• 1985

In this paper, Sentence-Sip E Il Ka Gi Seo U1 E Gan Da was spoken by 4 men and 3 see sound is used for the experiment. A/D conversion time is 30 sec. Data are obtained using the microcomputer and compressed by ADPCM Rate of compression is 1/8. Data compressed by ADPCM are synthesized and compared to the original sound. Rate of speech identification is analysed using the sound pressure, white noise. Coding of ADPCM is done for 5bit. As the result of fixing starting voltage by 2.6V. It is acertained that variable value increases in initial speech signal and then process is made by minimum value "3". From the result of processing, synthesized sound is almost eaual to original sound. Minimum values cause distorition, Dummy Head System is used in this experiment.xperiment.