• Journal of Broadcast Engineering
• /
• v.18 no.6
• /
• pp.919-926
• /
• 2013

The acoustic signal from a distance sound source in an enclosed space often produces reverberant sound that varies depending on room impulse response. The estimation of the level of reverberation or the quality of the observed signal is important because it provides valuable information on the condition of system operating environment. It is also useful for designing a dereverberation system. This paper proposes a speech quality estimation method based on the harmonicity of received signal, a unique characteristic of voiced speech. At first, we show that the harmonic signal modeling to a reverberant signal is reasonable. Then, the ratio between the harmonically modeled signal and the estimated non-harmonic signal is used as a measure of standard room acoustical parameter, which is related to speech clarity. Experimental results show that the proposed method successfully estimates speech quality when the reverberation time varies from 0.2s to 1.0s. Finally, we confirm the superiority of the proposed method in both background noise and reverberant environments.

• Speech Sciences
• /
• v.12 no.3
• /
• pp.139-151
• /
• 2005

Previous studies have shown that Korean students have difficulty identifying some English consonants which are not in the Korean sound inventory. The aim of this study was to examine the accuracy rate of English consonants correctly identified by 130 college students in order to find out which English consonants were difficult for the students to perceive. The subject's task was to identify one of the minimal pairs played in a quiet laboratory classroom. 100 minimal pairs consisted of syllables with various onsets or codas: stops, fricatives, affricates, liquids and nasals. Results were as follows: First, the average score of the English major group was significantly higher than that of the non-English major group. Second, there was a similar distribution in the rank order of minimal pairs sorted by the accuracy rate between the two groups. Third, the accuracy rate systematically decreased as each score range decreased. Fourth, the students showed higher accuracy in the perception of liquids than that of the stop-fricative contrast. Fifth, the accuracy score in onset position was higher than in coda position. Finally, the students still had problem telling voiced consonants from voiceless ones, especially in coda position. It would be desirable to extend the present research to middle or high school students to fundamentally resolve those listening problems.

• The Journal of the Acoustical Society of Korea
• /
• v.19 no.2E
• /
• pp.8-13
• /
• 2000

In this paper, we present an adaptive formant tracking algorithm for speech using closed phase WRLS-VFF-VT method. The pitch synchronous closed phase methods is known to give more accurate estimates of the vocal tract parameters than the pitch asynchronous method. However the use of a pitch-synchronous closed phase analysis method has been limited due to difficulties associated with the task of accurately isolating the closed phase region in successive periods of speech. Therefore we have implemented the pitch synchronous closed phase WRLS-VFF-VT algorithm for speech analysis, especially for formant tracking. The proposed algorithm with the variable threshold(VT) can provide a superior performance in the boundary of phone and voiced/unvoiced sound. The proposed method is experimentally compared with the other method such as two channel CPC method by using synthetic waveform and real speech data. From the experimental results, we found that the block data processing techniques, such as the two-channel CPC, gave reasonable estimates of the formant/antiformant. However, the data windows used by these methods included the effects of the periodic excitation pulses, which affected the accuracy of the estimated formants. On the other hand the proposed WRLS-VFF-VT method, which eliminated the influence of the pulse excitation by using an input estimation as part of the algorithm, gave very accurate formant/bandwidth estimates and good spectral matching.

• The KIPS Transactions:PartB
• /
• v.9B no.2
• /
• pp.231-236
• /
• 2002

The classification of the speech-sound block comes from the multi-resolution analysis property of the 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 vapid speech-sounds in terms of position and frequency range. It 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 signal-to-noise ratio and a useful system tuning for the system implementation.

• Phonetics and Speech Sciences
• /
• v.13 no.3
• /
• pp.53-64
• /
• 2021

It is known that the voicing contrast for Japanese word-initial stops is primarily realized as differences in the voice onset time (VOT). However, recent studies have reported that voiced stops are more often produced with a positive VOT than with a negative VOT among the younger generation nationwide. It is also known that post-stop F0 is associated with the stop contrast, but the degree of F0 use differs from region to region. This study explores whether the difference in post-stop F0 functions as a perceptual cue to the stop contrast along with VOT. Fifty-five college students who are native listeners from four different regions participated in two or three perception tests. The results show that VOT is a primary cue to the voiced-voiceless distinction of word-initial stops, but that the effect of post-stop F0 on the stop contrast is marginal. The post-stop F0 is involved in perception only when VOT is ambiguous, such that a sound with high F0 is more often perceived as a voiceless stop, but not vice versa. The results of this study indicate that the acoustic parameters associated with the stop contrast are not the same in production and perception, and suggest that other factors such as context, which is not an acoustic characteristic, may also be involved in the stop contrast.

• Korean Journal of Oriental Medicine
• /
• v.18 no.2
• /
• pp.25-38
• /
• 2012

Objective : This study was conducted to underpin the theory of Taegeuk Acupuncture systematically and increase the utility of clinical trial and a range of application. Method : This study analysed ByungHaeng Lee's theory, who is the originator of Taegeuk Acupuncture, by focusing on "Young Chu GuChimSibYiWon" Result & Conclusion : 1. ByungHaeng Lee's thought about Viscera and Bowels is following medicine of Internal Classic. Even though JeMa Lee's thought about Viscera and Bowels has same name with ByungHaeng Lee's, but the functional meaning and related part is totally different. So, four constitution types in Taegeuk Acupuncture are different from four constitution types in Four-Constitution Medicine. Therefore, Taegeuk Acupuncture isn't related with Four-Constitution Medicine 2. On the authority of "Young Chu GuChimSibYiWon", we can extend 4 therapeutic acupuncture point combinations of Taegeuk Acupuncture to ten therapeutic acupuncture point combinations which tentatively named 'Five Viscera Source Point Acupuncture'. 3. To consider an example of eight constitutional acupuncture, Taegeuk Acupuncture thoroughly says constitutional acupuncture. Especially, JaeGyuKim's Taegeuk Acupuncture has specific diagnosis standard that after treating a patient with acupuncture, he checks an oppressive pain at the precordial region and a voiced sound at liver and then he makes sure constitution of patient. So we can consider JaeGyuKim's Taegeuk Acupuncture as constitutional acupuncture. However, it has to change the name of four constitution types to taegeuk greater yang person, taegeuk lesser yang person, taegeuk greater yin person, taegeuk lesser yin person so as to avoid confusion. 4. Taegeuk Acupuncture is markedly simple in comparison with other acupuncture, so it needs to extend the utility of clinics by enlarging treatment acupuncture points.

• Journal of the Korean Institute of Intelligent Systems
• /
• v.17 no.7
• /
• pp.957-963
• /
• 2007

In this paper, Nonlinear Autoregressive (NAR) method based on Least Square-Support Vector Regression (LS-SVR) is introduced and tested for nonlinear sustained vowel modeling. In the database of total 43 sustained vowel of Benign Vocal Fold Lesions having aperiodic waveform, this nonlinear synthesizer near perfectly reproduced chaotic sustained vowels, and also conserved the naturalness of sound such as jitter, compared to Linear Predictive Coding does not keep these naturalness. However, the results of some phonation are quite different from the original sounds. These results are assumed that single-band model can not afford to control and decompose the high frequency components. Therefore multi-band model with wavelet filterbank is adopted for substituting single band model. As a results, multi-band model results in improved stability. Finally, nonlinear sustained vowel modeling using NAR based on LS-SVR can successfully reconstruct synthesized sounds nearly similar to original voiced sounds.

• The Journal of the Korea institute of electronic communication sciences
• /
• v.15 no.6
• /
• pp.987-994
• /
• 2020

In this paper, we studied the effect of kernel size of CNN layer on performance in acoustic noise attenuators. This system uses a deep learning algorithm using a neural network adaptive prediction filter instead of using the existing adaptive filter. Speech is estimated from a single input speech signal containing noise using a 100-neuron, 16-filter CNN filter and an error back propagation algorithm. This is to use the quasi-periodic property in the voiced sound section of the voice signal. In this study, a simulation program using Tensorflow and Keras libraries was written and a simulation was performed to verify the performance of the noise attenuator for the kernel size. As a result of the simulation, when the kernel size is about 16, the MSE and MAE values are the smallest, and when the size is smaller or larger than 16, the MSE and MAE values increase. It can be seen that in the case of an speech signal, the features can be best captured when the kernel size is about 16.

• Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
• /
• v.30 no.2
• /
• pp.112-117
• /
• 2019

Background and Objectives The purpose of this study was to develop the differential diagnosis scale containing items from adductor spasmodic dysphonia (ADSD) to muscle tension dysphonia (MTD) and the determine clinical utility of newly developed items. Materials and Method The four parts of pitch, redirected phonation, automatic speech and voiced sound were selected for analyzing the characteristics of ADSD in the literature. One part of tense voiceless sound was developed according to the Korean manner of articulation. The content validity was evaluated based on 5 scales (1-5 point) analysis from 30 experts. One hundred patients (50 ADSD and 50 MTD) were recorded in reading a sentence and sustained phonation. The two speech language pathologist evaluated recorded voices through a blind test using 4 scales (0-3 point) for newly developed items. Results As a result of verifying the content validity of items with experts, it was identified that the differentiated items were valid with 4.2 out of 5. Through the differential diagnosis between two groups according to the items, the correlation between sub-domains and total scores was shown as higher than 0.710. The result of analyzing the reliability on each diagnosis domain was 0.840-0.893, which showed the internal consistency of items was great. Newly developed five parts of ADSD were significantly higher than those of MTD with strong correlation (p<0.01). The reliability among the evaluators was analyzed as high with 0.892. Conclusion In this study, the differential diagnosis scale of ADSD was revealed as having validity and reliability. It is considered that it will be useful for differentiating ADSD and MTD in the clinical field.