• Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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• v.24 no.1
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• pp.28-32
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• 2013

Because the human voice is produced through vibration of the vocal cords during the exhalation of airflow, it is important to observe the vibration pattern of the vocal cords in patients complaining of voice changes. However, it is not easy to observe the actual vibration pattern of the vocal cord because it vibrates so fast that it cannot be observed by the naked eye and it is located deep in the throat. Recently, with advances in instruments, including laryngoscopes and video camera systems, the vibration pattern of the vocal cords can now be observed. However, considering that present video camera systems can detect 30-60 images per second and the vocal cord vibrates at 100-200 and 200-300 times per second in men and women, respectively, it is not possible to record the whole mucosal wave of the vocal cord in real time. To overcome this limitation, a stroboscope, which converts fast movements of the vocal cord into slower images, has been developed. Since then, several instruments were developed to examine vocal vibration pattern. However, each instruments have advantages and disadvantages. Therefore, we should know about these things to apply them in patients with voice problem.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.12
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• pp.5917-5935
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• 2018

Adaptive Multi-Rate Codec is one of the codecs which is used for making voice calls. It helps to connect people who are scattered in various geographical areas. It adjusts its bit-rate according to the user's channel conditions. It plays a vital role in providing an improved speech quality of voice connection in Long Term Evolution (LTE). There are some constraints which need to be addressed in providing this service profitably. Quality of Service (QoS) is the dominant mechanism which determines the quality of the speech in communication. On several occasions, number of users are trying to access the same channel simultaneously by standing in a particular region for a longer period of time. It refers to Multi-user channel sharing problem which leads to resource loss very often. The main aim of this paper is to develop a novel RSA - QoS scheduling algorithm for reducing the Resource Loss Ratio. Eventually, it increases the throughput.The simulation result shows that the RSA - QoS increases the number of users for accessing the resources better than the existing algorithms in terms of resource loss and throughput. Ultimately, it enhances the QoS in Mobile Networks.

• Journal of the military operations research society of Korea
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• v.26 no.1
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• pp.56-69
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• 2000

A frequency assignment problem on time division duplex system is considered. Republic of Korea Army (ROKA) has been establishing an infrastructure of tactical communication (SPIDER) system for next generation and it will be a core network structure of system. VHF system is the backbone network of SPIDER, that performs transmission of data such as voice, text and images. So, it is a significant problem finding the frequency assignment with no interference under very restricted resource environment. With a given arbitrary configuration of communications network, we find a feasible solution that guarantees communication without interference between sites and relay stations. We formulate a frequency assignment problem as an Integer Programming model, which has NP-hard complexity. To find the assignment results within a reasonable time, we take a genetic algorithm approach which represents the solution structure with available frequency order, and develop a genetic operation strategies. Computational result shows that the network configuration of SPIDER can be solved efficiently within a very short time.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.41 no.2
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• pp.51-64
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• 2004

The purpose of this paper is to establish the Standard Korean Vocal Sound into Character Conversion Rule (Standard VSCC Rule) by reversely applying the Korean Standard Pronunciation Rule that regulates the way of reading written Hangeul sentences. The Standard VSCC Rule performs a crucially important role in Korean speech recognition. The general method of speech recognition is to find the most similar pattern among the standard voice patterns to the input voice pattern. Each of the standard voice patterns is an average of several sample voice patterns. If the unit of the standard voice pattern is a word, then the number of entries of the standard voice pattern will be greater than a few millions (taking inflection and postpositional particles into account). This many entries require a huge database and an impractically too many comparisons in the process of finding the most similar pattern. Therefore, the unit of the standard voice pattern should be a syllable. In this case, we have to resolve the problem of the difference between the Korean vocal sounds and the writing characters. The process of converting a sequence of Korean vocal sounds into a sequence of characters requires our Standard VSCC Rule. Making use of our Standard VSCC Rule, we have implemented a Korean vocal sounds into Hangeul character conversion system. The Korean Standard Pronunciation Rule consists of 30 items. In order to show soundness and completeness of our Standard VSCC Rule, we have tested the conversion system with various data sets reflecting all the 30 items. The test results will be presented in this paper.

• 제어로봇시스템학회:학술대회논문집
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• 2003.10a
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• pp.603-608
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• 2003

Recently, personal robots and home robots are developing by many companies and research groups. It is considered that a general effective interface for user of those robots is speech or voice. In this paper, Japanese speech based man-machine interface system is discussed for reflecting the fuzziness of natural language on robots, by using fuzzy reasoning. The present system consists of the derivation part of action command and the modification part of the derived command. In particular, a unique problem of Japanese is solved by applying the morphological analyzer ChaSen. The proposed system is applied for the motion control of a robot manipulator. It is proved from the experimental results that the proposed system can easily modify the same voice command to the actual different levels of the command, according to the current state of the robot.

• Speech Sciences
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• v.8 no.2
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• pp.49-59
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• 2001

The phoneme segmentation is a very difficult problem in speech sound processing because it has found out segmental algorithm in many kinds of allophone and coarticulation's trees. Thus system configuration for the speech recognition and voice retrieval processing has a complex system structure. To solve it, we discuss a possibility of new segmental algorithm, which is called the minus a thirds one or plus in tripartitioning(삼분손익) of twelve temporament(12 율려), first proposed by Prof. T. S. Han. It is close to oriental and western musical theory. He also has suggested a 3 consonant and 3 vowel phonemes in Hunminjungum(훈민정음) invented by the King Sejong in the 15th century. In this paper, we suggest to newly name it as ortho-phonic phoneme(OPP/정음소), which carries the meaning of 'the absoluteness and independency'. OPP also is acceptable to any other languages, for example IPA. Lastly we know that this algorithm is constantly applicable to the global language and is very useful to construct a voice recognition and retrieval structuring engineering.

• Industrial Engineering and Management Systems
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• v.10 no.3
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• pp.179-184
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• 2011

Recently, the fast development of communication technologies has brought a great convince for human beings' life. Lots of commercial services and transactions can be done by using mobile communication equipments such as smart phones. Consequently, smart phones have attracted lots of companies to invest them for their potential growth of market. Compared with basic feature phone, a smart phone can offer more advanced computing ability and connectivity. However, based on the responses of customers, there still are many defectives such as not friendly and smooth operation, short standby time of batteries, threat of virus infected and so on needed to be improved. Therefore, this study will propose a product innovative function development procedure into TRIZ (theory of inventive problem solving) to transform voice of customers into product design and to create novel functions, respectively. A case study of smart phones will be provided to illustrate the effectiveness of the proposed method.

• Smart Media Journal
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• v.9 no.3
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• pp.116-121
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• 2020

Digital Information Display and KIOSK have a problem that initial introduction cost and maintenance cost due to the development cost of dedicated contents and installation cost are high due to the characteristics of the product. In order to solve these problems, We designed and implemented of voice transmission system using Open Source Hardware and Event based Non-Blocking I/O Algorithm.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.38 no.5
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• pp.567-575
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• 2001

This paper suggests new method to get fixed size parameter from different length of voice signals. The efficiency of speech recognizer is determined by how to compare the similarity(distance of each pattern) of the parameter from voice signal. But the variation of voice signal and the difference of speech speed make it difficult to extract the fixed size parameter from the voice signal. The method suggested in this paper is to normalize the parameter at fixed size by using the 2 dimension DCT(Discrete Cosine Transform) after representing the parameter by spectrogram. To prove validity of the suggested method, parameter extracted from 32 auditory filter-bank(it estimates auditory nerve firing probabilities) is used for the input of neural network after being processed by 2 dimension DCT. And to compare with conventional methods, we used one of conventional methods which solve time alignment problem. The result shows more efficient performance and faster recognition speed in the speaker dependent and independent isolated word recognition than conventional method.

• Maxillofacial Plastic and Reconstructive Surgery
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• v.30 no.5
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• pp.437-446
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• 2008

Electroglottography (EGG) is a simple and non-invasive technique for analyzing the vibratory patterns of the vocal folds by detecting impedance changes across the larynx. An abnormal electroglottogram is shown in patients who have a dysphagia associated with neuromuscular disorder. Electroglottography offers reliable informations for diagnosis of swallowing disorder and gives quantitative datas. The purpose of this study is to provide the normal value of electroglottography in normal adults. We took electroglottograms of 80 adults who have no problem in swallowing and utterance. EGG data were analyzed to find out the value of Pitch, Jitter and Closed quotient with a commercially available software. There were significant differences between a usual voice and loud voice in 3 measures on the EGG signalmean pitch, Avg. jitter, mean quotient. To get a proper electroglottography, phonation of a usual voice was better than a loud voice. Four measurements- S.D pitch, Avg. Jitter, Mean closed quotient, S.D closed quotient- were independent of sex for adult. Three measurements- Mean pitch, S.D pitch, Mean closed quotient - were independent of age for adult aged twenties to fifties. The Avg. Jitter of twenties appeared to be lower than those of forties and fifties. The S.D closed quotient of twenties appeared to be lower than those of thirties, forties and fifties.