• Phonetics and Speech Sciences
• /
• v.5 no.2
• /
• pp.43-51
• /
• 2013

English native listeners have a tendency to treat strong syllables in a speech stream as the potential initial syllables of new words, since the majority of lexical words in English have a word-initial stress. The current study investigates whether Korean (L1) - English (L2) late bilinguals perceive strong syllables in English continuous speech as word onsets, as English native listeners do. In Experiment 1, word-spotting was slower when the word-initial syllable was strong, indicating that Korean listeners do not perceive strong syllables as word onsets. Experiment 2 was conducted in order to avoid any possibilities that the results of Experiment 1 may be due to the strong-initial targets themselves used in Experiment 1 being slower to recognize than the weak-initial targets. We employed the gating paradigm in Experiment 2, and measured the Isolation Point (IP, the point at which participants correctly identify a word without subsequently changing their minds) and the Recognition Point (RP, the point at which participants correctly identify the target with 85% or greater confidence) for the targets excised from the non-words in the two conditions of Experiment 1. Both the mean IPs and the mean RPs were significantly earlier for the strong-initial targets, which means that the results of Experiment 1 reflect the difficulty of segmentation when the initial syllable of words was strong. These results are consistent with Kim & Nam (2011), indicating that strong syllables are not perceived as word onsets for Korean listeners and interfere with lexical segmentation in English running speech.

• The Journal of the Acoustical Society of Korea
• /
• v.15 no.4E
• /
• pp.28-36
• /
• 1996

In this paper, we propose a new evaluation methodology which can more clearly show the performance of the allophone modeling algorithm generally used in large vocabulary speech recognition. The proposed evaluation method shows the running characteristics and limitations of the modeling algorithm by testing how the variation of phonetic environments of training data affects the recognition performance and the desirable number of free parameters to be estimated. Using the method, we experiment results, we conclude that, in vocabulary-independent recognition task, the phonetic diversity of training data greatly affects the robustness of model, and it is necessary to develop a proper measure which can determine the number of states compromizing the robustness and the precision of the HMnet better than the conventional modeling efficiency.

• Proceedings of the IEEK Conference
• /
• 2003.07e
• /
• pp.2459-2462
• /
• 2003

In this paper, we Propose a wavelet based adaptive algorithm which improves the convergence speed and reduces computational complexity using the fast running FIR filtering efficiently. We compared the performance of the proposed algorithm with time and frequence domain adaptive algorithm using computer simulation of adaptive noise canceler based on synthesis speech.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.29 no.11C
• /
• pp.1573-1580
• /
• 2004

In this paper, an adaptation mode controller for adaptive microphone array system is proposed for high-quality speech acquisition in real environments. To ensure proper adaptation of the adaptive array algorithm, the proposed adaptation mode controller uses not only temporal information, but also spatial information. The proposed adaptation mode controller is constructed with two processing stages: an initialization stage and a running stage. In the initialization stage, a sound source localization technique is adopted, and a signal correlation characteristic is used in the running stage. For the adaptive may algorithm, a generalized sidelobe canceller with an adaptive blocking matrix is used. The proposed adaptation mode controller can be used even when the adaptive blocking matrix is not adapted, and is much stable than the power ratio method. The proposed algorithm is evaluated in real environment, and simulation results show 13dB SINR improvement with the speaker sitting 2m distance from the may.

• The Journal of the Korea Contents Association
• /
• v.19 no.12
• /
• pp.270-278
• /
• 2019

This study was performed to compare the aerodynamic characteristics of young and elderly adult male patients with voice disorders during continuous speech. Aerodynamic measurements were obtained after 12 young male patients and 9 elderly male patients read a paragraph. The elderly group showed longer duration, lower airflow rate and air volume than the younger group, but the differences were not significant except phonation time. So, when interpreting the meaning of aerodynamic measures of elderly voice disorder patients in the aspects of airflow and air volume, it should take into account various conditions(e. g. reading materials, pulmonary functions) as well as age.

• Journal of Digital Convergence
• /
• v.13 no.1
• /
• pp.397-405
• /
• 2015

This survey was to identify voice symptoms and vocal-health service related experiences of occupational voice users(teachers, telemarketers, speech therapists). The 91.8% of teachers, 97.9% of telemarketers, 86% of speech therapists surveyed reported more than one voice symptom. The symptoms were classified as 9 categories(running a temperature, getting dry, dry and cough, pain, phlegm, tingled, hoarseness, cracks, swollen) and the most frequently reported from 3 groups was 'getting dry'. The 85.7% of teachers, 87.8% of telemarketers, 66% of therapists surveyed had no experience of vocal-health related services. The 19.6%, 19.9%, and 72% of each group reported they have heard both of 'voice/speech therapist'. The 36.8% of teachers and 43.6% of telemarketers answered they don't know how to use their voice efficiently and 45.3% of the teachers, 43.6% of the telemarketers, 28% of the therapists surveyed asked professional help for their voice. The result showed that most of the occupational voice users surveyed experienced voice symptoms but rarely knew professional vocal-health related services.

• The Journal of the Acoustical Society of Korea
• /
• v.24 no.4E
• /
• pp.123-127
• /
• 2005

In this paper, we propose a new method for reducing the number of distance calculations in the LBG (Linde, Buzo, Gray) algorithm, which is widely used method to construct a codebook in vector quantization of speech recognition system. The proposed algorithm can reduce the distance calculation between input vector and codeword by utilizing the observation that codewords are quickly stabilized as the number of iteration increases. From the simulation results, it is shown that we can reduce the running times over $43.77\%$ on average in comparison with current LBG algorithm without sacrificing the performance of codebook.

• Journal of Korean Society of Industrial and Systems Engineering
• /
• v.20 no.42
• /
• pp.161-169
• /
• 1997

A variety of learning methods are used for neural networks. Among them, the backpropagation algorithm is most widely used in such image processing, speech recognition, and pattern recognition. Despite its popularity for these application, its main problem is associated with the running time, namely, too much time is spent for the learning. This paper suggests a method which maximize the convergence speed of the learning. Such reduction in e learning time of the backpropagation algorithm is possible through an adaptive adjusting of the slope of the activation function depending on total errors, which is named as the variable slope algorithm. Moreover experimental results using this variable slope algorithm is compared against conventional backpropagation algorithm and other variations; which shows an improvement in the performance over pervious algorithms.

• Proceedings of the Acoustical Society of Korea Conference
• /
• 1998.06c
• /
• pp.47-50
• /
• 1998

본 논문은 자동차의 편의성 및 안전성의 동시 확보를 위하여, 보조적 스위치의 조작없이 상시 음성의 입,출력이 가능하도록 하며, band pass filter를 이용하여 잡음환경에서 자동으로 정확하게 음성구간 검출(End Point Detection)을 하게 하였다. Reference Pattern은 Dynamic Multi-Section(DMS)[1] 모델을 사용하였고 차량의 속도에 따라 자동으로 잡음환경에 강인한 모델을 선택하도록 하였으며, 음성의 특징 파라미터와 인식 알고리즘은 Perceptual Linear Predictive(PLP) 13차와 One Stage Dynamic Programming(OSDP)를 사용하였다. 주행중인 자동차 환경(30~70km/h)에서 자주 사용되는 차량제어 명령 33개에 대하여 화자독립 92.98%, 화자종속 94.44% 인식율을 구하였다. 또한 주행중인 차량에서 카폰, 핸드폰 사용으로 인한 사고를 줄이기 위하여 음성으로 전화를 걸 수 있도록 하는 Voice Dialing 기능도 구현하였다.

• Proceedings of the KOSOMBE Conference
• /
• v.1997 no.11
• /
• pp.79-82
• /
• 1997

In this paper, a body-worn type digital hearing aid (DHA) based on a dedicated DSP chip is developed. A fitting software running on a PC supported by the Win95 OS is also developed. The fitting protocol is based on the NAL-R procedure applied to eight frequency bands, but it is designed to support a curvilinear fitting to cope with the nonlinear perception of hearing-impaired listeners. Preliminary subjective tests regarding the speech intelligibility and perceived quality revealed that the new DHA could be of benefit to hearing aid users.