• Journal of the Korea Academia-Industrial cooperation Society
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• v.9 no.5
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• pp.1210-1217
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• 2008

Sasang medicine is our country's unique traditional medicine based on the classification of individual physical constitution. In the Sasang medicine, what is considered the most important task is to categorize Sasang constitution exactly. Therefore, security of objective elements and diagnosis index is a problem awaiting solution in Sasang constitution classification. To this the paper abstracted result value from objectification, visualization, a fixed quantity of Sasang constitution to analyze face image signals and voice signals. And, comparing the differences constitution would like to develop system classification of Sasang constitution. Specially, image and voice signals are different because of gender, age, region so it composed Sasang constitution group to 40-50 years women in Seoul. To extract of these image and voice signals wanted to perform comparison, analysis in constitution. Finally, it would like to prove a significance of research result through experiment.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.13 no.4
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• pp.1665-1671
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• 2012

The purpose of this study was to look into changes in voice using acoustic analysis during the process of neuromuscular electrical stimulation targeting dysphagia treatment. Fifteen man with dysphagia caused by stroke was treated neuromuscular electrical stimulation for two months and intensity of voice, $F_0$, Jitter, Shimmer, NNE were measured. The results of this study that improvement in functions of dysphagia and Jitter, Shimmer were stabilized. But there was not significantly changes of $F_0$. NNE was improved after the intervention, but still showed abnormal levels. This result suggests a possibility of effects that Neuromuscular electrical stimulation has on stabilization of Jitter, Shimmer and intensity of voice.

• Journal of the Korea Society for Simulation
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• v.16 no.3
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• pp.75-87
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• 2007

As the VoIP service quality is getting better and many shortcomings are being overcome, users are getting interested in this service. Also, there are several additional features that provide a convenience to users such as presence service, instant messaging service and so on. But, as there are always two sides of rein, some security issues have users hesitate to make use of it. This paper deals with one of the issues, the VoIP spam problem. We took into account the signal pattern of voice message in spam call and we have constructed voice signal models of normal call, normal call with noise and spam call. Each voice signal case is inserted into our spam decision algorithm which detects the spam calls based on the amount of information in the call signal. We made use of the DEVS-$Java^{TM}$ for our modeling and simulation. The contribution of this work is in suggestion of a way to detect voice spam call signal and testing of the method using modeling and simulation methodology.

• Journal of Digital Convergence
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• v.19 no.7
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• pp.209-215
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• 2021

This study is a system development for voice information protection equipment in major meetings and places requiring security. Security performance and stability were secured with information leakage prevention technology through generation of false noise and ultrasonic waves. The cutoff frequency band for blocking the leakage of voice information, which has strong straightness due to the nature of the radio wave to the recording prevention module, blocks the wideband frequency of 20~20,000Hz, and the deception jamming technology is applied to block the leakage of voice information, greatly improving the security. To solve this problem, we developed a system that blocks the recording of a portable smartphone using a battery, and made the installation of a separate device smaller and lighter so that customers do not recognize it. In addition, it is necessary to continuously study measures and countermeasures for efficiently using the output of the anti-recording speaker for long-distance recording prevention.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1203-1206
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• 1998

Considerable research in the last three decades has examined the problem of enhancement of speech degraded by additive background noise. We compare traditional methods such as spectral subtraction and Wiener filter, recently proposed psychoacoustic model based methods such as perceptual filter and noise suppression in EVRC in terms of performance and complexity.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.62 no.12
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• pp.1744-1749
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• 2013

Acoustic feedback is perceived as whistling or howling, which is a major complaint of hearing-aids users. Acoustic feedback cancellation is important in hearing-aids because acoustic feedback degrades performance of the hearing aid device by reducing maximum insertion gain. Adaptive systems for estimate acoustic feedback path and feedback suppression algorithms have been proposed in order to solve this problem. A typical feedback cancellation algorithm is LMS(least mean squares) because of its computational efficiency. However it has problem of convergence performance in high correlated input signal. In this paper, we propose a new variable step-size normalized LMS(least mean squares) algorithm using VAD(voice activity detection) to overcome the limitation of the LMS algorithm. The VAD algorithm is GSAP(global speech absence probability) and the feedback cancellation algorithm is normalized LMS. The proposed algorithm applies different step-size between voice and non-voice using VAD, for high stability, fast convergence speed and low misalignment when correlated inputs, such as speech. The result of simulation with white noise mixed speech signal, the proposed algorithm shows high performance then traditional algorithm in terms of stability, convergence speed and misalignment.

• Journal of Korea Game Society
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• v.19 no.4
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• pp.97-108
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• 2019

Recently interests in voice game commands have been increasing due to the diversity and convenience of the input method, but also by the distance between commands. The command distance is the phonetic difference between command utterances, and as such distance increases, the recognition rate improves. In this paper, we propose an IP(Integer Programming) modeling of the problem which is to select a combination of commands from given candidate commands for maximizing the average distance. We also propose a SA(Simulated Annealing)-based algorithm for solving the problem. We analyze the characteristics of our method using experiments under various conditions such as the number of commands, allowable command length, and so on.

• Journal of the International Relations & Interdisciplinary Education
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• v.4 no.1
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• pp.35-56
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• 2024

Based on Kingdon's policy stream model, the logical mechanism of student burden double reduction policy agenda setting process in China was analyzed from three aspects: problem stream, policy stream, political stream. It was found that the direct reflection of data, the feedback and improvement of current policies constitute the problem stream.The high attention of government departments, the suggestions of CPPCC members and experts and scholars, the strong voice of schools, and the continuous voice of stakeholders constitute the policy stream. The ruling party's governing idea and educational policy, national mood and interest game during the transition period constitute the political policy stream. The three streams, coupling interaction, open the policy window, promoting the introduction of the 'double reduction' policy.

• Journal of Digital Convergence
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• v.13 no.7
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• pp.177-182
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• 2015

In a MPC(Multi-Pulse Coding) using excitation source of voiced and unvoiced, it would be a distortion of voice waveform. This is caused by normalization of synthesis speech waveform of voiced in the process of restoration. To solve this problem, this paper present APC-MPC of amplitude-position compensation in a multi-pulses each pitch interval in order to reduce distortion of synthesis waveform. Also, I was implemented that the APC-MPC in coding system. And I evaluate the SNRseg of APC-MPC in 8kbps coding condition of convergence system. As a result, SNRseg of APC-MPC was 13.9dB for female voice and 14.3dB for male voice respectively. And so, I expect to be able to this method for cellular phone and smart phone using excitation source of low bit rate.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.5
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• pp.842-856
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• 1994

The various services that a broadband integrated services digital network (B-ISDN) carries, have a wide range of delay, delay jitter and cell loss probability requirements. Design of appropriate control schemes for B-ISDN is an extremely important and challenging problem. In this paper, we proposed multiplexing algorithm with both delay and loss priorities in order to satisfy the diverse requirements. For the implementation of cell lose priority, we assumed that voice cells are generated as non-discardable(i.e., high priority) and discardable (i.e., low priotity)cells. The low priority voice cell may be discarded inside the network if congestion occurs. The cell dropping scheme is shown to reduce cell losses as well as delays for both voice and data. Such a load shedding scheme is expected to improve significantly utilization of B-ISDN.