• 대한음성학회:학술대회논문집
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• 대한음성학회 2007년도 한국음성과학회 공동학술대회 발표논문집
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• pp.102-105
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• 2007

This study was designed to examination effects of concurrent linguistic or cognitive tasks on speech rate. Eight normal speakers were repeated sentences either with or without simultaneous a linguistic task and a cognitive task. Linguistic task was conducted by generating verbs from nouns and cognitive task meaned performing mental arithmetic. Speech rate was measured from acoustic data. One-way ANOVA conducted to know speech rate difference among 3 different type of tasks. The results showed there was no significant difference between sentence repeat and linguistic tasks. But There was significant difference findings: sentence repeat and linguistic task, linguistic and cognitive task.

• 말소리와 음성과학
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• 제6권3호
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• pp.33-40
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• 2014

The purpose of the study was to examine alternating motion rate and its variability in children with childhood apraxia of speech (CAS) compared to typically developing children. Six children with CAS aged 9-12 years old and 10 children who were age-matched participated in the study. This study measured tokens per second and variabilities of the rates during the production of /$p^*$ a/, /$t^*$ a/, and /$k^*$ a/. For variability measures of the rates, each participant was asked to repeat speech tasks three times and the average value of the rates and its standard deviation were obtained. The results revealed that the CAS group showed slower rate only at /$k^*$ a/ than the control group. The CAS group exhibited greater variability of AMR at all the tasks than the control group. The results suggested that variability of AMR might be a more distinctive speech feature to children with CAS than the rate of the speech task.

• 말소리와 음성과학
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• 제9권4호
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• pp.91-97
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• 2017

This study categorizes several voice characteristics by subjective listening assessment, and investigates correlation between voice characteristics and speech feature parameters. A model was developed to classify voice characteristics into the defined categories using SVM algorithm. To do this, we extracted various speech feature parameters from speech database for men in their 20s, and derived statistically significant parameters correlated with voice characteristics through ANOVA analysis. Then, these derived parameters were applied to the proposed SVM model. The experimental results showed that it is possible to obtain some speech feature parameters significantly correlated with the voice characteristics, and that the proposed model achieves the classification accuracies of 88.5% on average.

• 음성과학
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• 제10권3호
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• pp.29-36
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• 2003

The purposes of this study was to conduct a correlational analysis among $F_^{0}$, Jitter, Shimmer, and NHR (HNR), and NNE estimated by three speech analysis softwares, MDVP, Praat and Dr. Speech. Thirty females and 15 males with normal voice participated in the study. We used Sound Forge 6.0 to record their voice. MDVP, Praat and Dr. Speech were used to measure the acoustic parameters. The Pearson correlation coefficient was determined through a statistical analysis. The results came out as follows: Firstly, there was a strong correlation between $F_^{0}$ and Shimmer of both instruments. However, there was no correlation between Jitter of both instruments. Secondly, Shimmer showed a stronger correlation with HNR, NHR, and NNE than Jitter. Therefore, Shimmer was considered to be more useful and sensitive parameter to identify dysphonic voice compared to jitter.

• 음성과학
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• 제10권3호
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• pp.79-86
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• 2003

This study compared the voices of ministers and priests. There. has been a common notion that ministers is more passionate than priests in delivering their speech. Therefore, it can be assumed that ministers abuses or misuses his/her voice compared to priests. This study attempted acoustic analysis of the voices of 6 ministers and .5 priests before and after their speech. We measured F0, jitter, shimmer, NNE and HNR using Dr. Speech (Version 4.0, Tiger DRS). A t-test was performed to determine any objective differences of their voices. The results showed that there were no significant differences in the voices of ministers and priests before and after their speech. However, there seemed to be an interesting reversed tendency between ministers and priests, although it did not reach a statistical significance. That is, P0 tended to increase after the speech in ministers, whereas it tended to decrease in priests. In addition, HNR tended to decrease after the speech in priests, while it tended to increase in ministers.

• 음성과학
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• 제11권2호
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• pp.247-257
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• 2004

There has been many research efforts to overcome the problems of speech recognition in the noisy conditions. Among them, the model-based compensation methods such as the parallel model combination (PMC) and vector Taylor series (VTS) have been found to perform efficiently compared with the previous speech enhancement methods or the feature-based approaches. In this paper, a data-driven model compensation approach that adapts the HMM(hidden Markv model) parameters for the noisy speech recognition is proposed. Instead of assuming some statistical approximations as in the conventional model-based methods such as the PMC, the statistics necessary for the HMM parameter adaptation is directly estimated by using the Baum-Welch algorithm. The proposed method has shown improved results compared with the PMC for the noisy speech recognition.

• 말소리와 음성과학
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• 제2권1호
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• pp.77-85
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• 2010

This paper proposes an efficient feature vector processing technique to guard the Speech Emotion Recognition (SER) system against a variety of noises. In the proposed approach, emotional feature vectors are extracted from speech processed by comb filtering. Then, these extracts are used in a robust model construction based on feature vector classification. We modify conventional comb filtering by using speech presence probability to minimize drawbacks due to incorrect pitch estimation under background noise conditions. The modified comb filtering can correctly enhance the harmonics, which is an important factor used in SER. Feature vector classification technique categorizes feature vectors into either discriminative vectors or non-discriminative vectors based on a log-likelihood criterion. This method can successfully select the discriminative vectors while preserving correct emotional characteristics. Thus, robust emotion models can be constructed by only using such discriminative vectors. On SER experiment using an emotional speech corpus contaminated by various noises, our approach exhibited superior performance to the baseline system.

• 대한음성학회:학술대회논문집
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• 대한음성학회 2006년도 추계학술대회 발표논문집
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• pp.131-134
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• 2006

Quality of narrowband speech $(0{\sim}4kHz)$ can be enhanced by the bandwidth expansion technique, by which the high- band components are estimated. This paper proposes the bandwidth expansion method using the spline codebook based spectral folding. For the performance evaluation, the PESQ(Perceptual Evaluation of Speech Quality) scores are measured as the objective measurement In addition, the MOS (Mean Opinion Score) and the preference tests are performed as the subjective measurement. The results show our proposed method outperforms the existing spline based one.

• 음성과학
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• 제7권3호
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• pp.153-165
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• 2000

During the phonation of voiced sounds, instants exist where the glottis is opened or closed, due to the periodic vibration of the vocal cord. When closed, this is called the glottal closure instant(GCI) or epoch.. The correct detection of the GCI is one of the important problems in speech processing for pitch detection, pitch synchronous analysis, and so on. Recently, it has been shown that the local maxima points of the wavelet transformed speech signal correspond to the GCIs of speech signal. In this paper, we investigate the accuracy of Gels estimated from this wavelet transformed speech signal. For this purpose we compare them with the negative peak points of the differentiated EGG signal that represents the actual GCIs of speech signal.

• 음성과학
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• 제7권3호
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• pp.87-98
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• 2000

Timing in speech is determined by many factors. In this paper, we introduce and discuss some factors that have generally been regarded as important in speech timing. They include stress, syllable structure, consonant insertion or deletion, tempo, lengthening at clause, phrase and word boundaries, preconsonantal vowel shortening, and compensation between segments or within phonological units (e.g., word, foot), compression due to the increase of syllables in word or foot level, etc. and each of them may playa crucial role in the structuring of speech timing in a language. But some of these timing factors must interact with each other rather than be independent and the effects of each factor on speech timing will vary from language to language. On the other hand, there could well be many other factors unknown so far. Finding out and investigating new timing factors and reinterpreting the already-known timing factors should enhance our understanding of timing structures in a given language or languages.