• Phonetics and Speech Sciences
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• v.4 no.1
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• pp.3-11
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• 2012

The aim of this paper is to examine the phonetic realizations of English unstressed vowels produced by advanced Korean learners (KLs) of English compared with English native speakers (NSs) focusing on the comparative study of English words and English loanwords. The result shows that KLs are usually not native-like in producing the English unstressed vowel /ə/ and loanword orthography affects the way the KLs produce /?/. The vowel quality of the unstressed vowels produced by the KLs is different from that of the NSs. In duration and pitch, KLs show significantly less difference between the stressed and unstressed vowels than do the NSs. The KLs usually have a high pitch in the stressed and the last syllable while the NSs usually produce peak F0 in the stressed syllable. When the KLs have a similar vowel quality with that of the NSs, they produce a shorter duration of the unstressed vowels. However, there is no correlation between the realization of the pitch and the vowel quality in KLs speech.

• Speech Sciences
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• v.14 no.3
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• pp.111-125
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• 2007

It is important to examine patients' subjective evaluation as well as objective measures and clinician's rating to assess voice disorders. This study aimed to evaluate validity and reliability of Korean-version of Voice Handicap Index (KVHI) and Voice-Related Quality of Life (KVQOL) with 113 adults with voice disorders and 111 normal adults. Content validity was verified by three experienced speech-language pathologists. Concurrent validity was revealed by examining the correlation among KVHI, KVQOL, and Voice Rating Scale as well as item discrimination coefficients. Total scores of KVHI and KVQOL of adults with voice disorders were significantly different from those of normal adults. Test-retest reliability and internal consistencies were significantly high in both KVHI and KVQOL. Correlations among scores of each subscale and total score were also significantly high in each tool. The study revealed that KVHI and KVQOL are suitable tools to be used in clinics and research areas in Korea, which can subjectively evaluate the effects of voice disorders on daily life as well as on quality of life.

• Phonetics and Speech Sciences
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• v.2 no.4
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• pp.223-228
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• 2010

This study investigated the closed quotient and other voice quality parameters using electroglottography (EGG) in sustaining the vowel /a/ and reading a sentence at the comfortable pitch and loudness in healthy Korean adults. Seventy two healthy adults (36 men, 36 women) aged 20~40 years were included in the study. The tasks were recorded and analyzed using Lx Speech Studio. In vowel sustaining task, closed quotient (Qx), fundamental frequency (Fx), sound pressure level (SPL), Jitter, and Shimmer were measured. In sentence reading task, closed quotient (DQx), fundamental frequency (DFx), and sound pressure level (DAx) were measured. The sex effects were observed on Qx, Fx, Shimmer, DQx, and DFx. Men had significantly higher Qx and DQx than women, but had significantly lower Shimmer than women. However, there was no sex effect on Jitter. The task effects on Qx and SPL as well as DQx and DAx were also assessed. Qx and SPL were significantly higher than DQx and DAx in both gender. This study showed that the closed quotients in both vowel sustaining and sentence reading tasks were significantly related to other voice quality parameters. Therefore, clinicians and researchers should describe the voice quality parameters like fundamental frequency, sound pressure level, Jitter, Shimmer, and so on when reporting closed quotients using EGG.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.1
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• pp.175-182
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• 2010

In this paper, we propose a robust speech reinforcement technique to enhance the intelligibility of the degraded speech signal under the ambient noise environments based on soft decision scheme incorporating a speech absence probability (SAP) with speech reinforcement gains. Since the ambient noise significantly decreases the intelligibility of the speech signal, the speech reinforcement approach to amplify the estimated clean speech signal from the background noise environments for improving the intelligibility and clarity of the corrupted speech signal was proposed. In order to estimate the robust reinforcement gain rather than the conventional speech reinforcement method between speech active periods and nonspeech periods or transient intervals, we propose the speech reinforcement algorithm based on soft decision applying the SAP to the estimation of speech reinforcement gains. The performances of the proposed algorithm are evaluated by the Comparison Category Rating (CCR) of the measurement for subjective determination of transmission quality in ITU-T P.800 under various ambient noise environments and show better performances compared with the conventional method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.6C
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• pp.588-592
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• 2009

In this paper, we propose a novel method to improve the performance of the global soft decision which is based on the second-order conditional maximum a posteriori (CMAP). Conventional global soft decision scheme has an disadvantage in that the speech absence probability adjusted by a fixed-parameter was sensitive to the various noise environments. In proposed approach using the second-order CMAP, speech absence probability value is more flexible which exploit not only the current observation but also the speech activity decisions in the previous two frames. Experimental results show that the proposed improved global soft decision method based on second-order conditional MAP yields better results compared to the conventional global soft decision technique with the performance criteria of the ITU-T P. 862 perceptual evaluation of speech quality (PESQ).

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.38 no.5
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• pp.576-582
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• 2001

The Multi-Band Excitation(MBE) speech coder uses a different approach for the representation of the excitation signal. It replaces the frame-based single voiced/unvoiced classification of a classical speech coder with a set of such decision over harmonic intervals in the frequency domain. This enables each speech segment to be a mixture of voiced and unvoiced, and improves the synthetic speech quality by reducing decision errors that might occur on the frame-based single voiced and unvoiced decision process when input speech is degraded with noise. The IMBE-LP, improved version of MBE with linear prediction, represents the spectral information of MBE model with linear prediction coefficients to obtain low bit rate of 2.4 kbps. In this Paper, we proposed a variable rate IMBE-LP vocoder that has lower bit rate than IMBE-LP without degrading the synthetic speech quality. To determine the LP order, it uses the spectral band information of the MBE model that has something to do with he input speech's characteristics. Experimental results are riven with our findings and discussions.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.4
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• pp.619-627
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• 2015

This paper presents a speech enhancement method using non-negative matrix factorization (NMF). In the training phase, each basis matrix of source signal is obtained from a proper database, and these basis matrices are utilized for the source separation. In this case, the performance of speech enhancement relies heavily on the basis matrix. The proposed method for which speech basis matrix is made a high reconstruction error for noise signal shows a better performance than the standard NMF which basis matrix is trained independently. For comparison, we propose another method, and evaluate one of previous method. In the experiment result, the performance is evaluated by perceptual evaluation speech quality and signal to distortion ratio, and the proposed method outperformed the other methods.

• The Journal of the Acoustical Society of Korea
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• v.32 no.3
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• pp.279-285
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• 2013

In this paper, we propose a novel speech enhancement method to improve the performance of the conventional global soft decision which is based on the spectral gradient method applied to the ratio of a priori speech absence and presence probability value (q). Conventional global soft decision scheme used a fixed value of q in accordance with the hypothesis assumed, but the proposed algorithm is a technique for improving the speech absence probability which is applied adaptively variable value of q according to the speech presence or absence in the previous two frames and the conditions of the spectral gradient value. Experimental results show that the proposed improved global soft decision method based on the spectral gradient method yields better results compared to the conventional global soft decision technique based on the performance criteria of the ITU-T P. 862 PESQ (Perceptual Evaluation of Speech Quality).

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.49 no.3
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• pp.98-105
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• 2012

This paper describes a new method for a speech playback speed control with high quality. The proposed method provides an adaptive threshold filtering solution for detecting active regions of a speech signal that are followed by playback speed. For a given playback speed, threshold value is adaptively determined with the statistics(:mean and standard deviation) of each frame in speech, and is used to select only active blocks within the current frame. To minimize quality degradation(i.e., pitch degradation) caused due to high-speed playback, the threshold filtering priorly eliminates relatively low-activity blocks including voice and unvoice. Simulation results show that the proposed scheme provides a playback speed control solution with higher quality than SOLA(Synchonized OverLap Add) method using the pitch extraction of speech.

• The Journal of the Acoustical Society of Korea
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• v.40 no.5
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• pp.503-514
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• 2021

Reverberation in speech signals tends to significantly degrade the performance of the Blind Source Separation (BSS) system. Especially in online systems, the performance degradation becomes severe. Methods based on joint diagonalizability constraints have been recently developed to tackle the problem. To improve the quality of separated speech, in this paper, we add the proposed de-reverberation method to the online BSS algorithm based on the constraints in reverberant environments. Through experiments on the WSJCAM0 corpus, the proposed method was compared with the existing online BSS algorithm. The performance evaluation by the Signal-to-Distortion Ratio and the Perceptual Evaluation of Speech Quality demonstrated that SDR improved from 1.23 dB to 3.76 dB and PESQ improved from 1.15 to 2.12 on average.