• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.1
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• pp.150-156
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• 2007

Noise is generated by several causes, when signal is processed. Hence, it generates error in the process of data transmission and decreases recognition ratio of image and speech data. Therefore, after eliminating those noises, a variety of methods for reconstructing the signal have been researched. Recently, wavelet transform which has time-frequency localization and is possible for multiresolution analysis is applied to many fields of technology. Then threshold-and correlation-based methods are proposed for removing noise. But, conventional methods accept a lot of noise as an edge and are impossible to remove the additive white Gaussian noise (AWGN) and the impulse noise at the same time. Therefore, in this paper we proposed new wavelet-based algorithm for reconstructing degraded signal by noise and compared it with conventional methods.

• The Journal of the Acoustical Society of Korea
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• v.33 no.5
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• pp.316-325
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• 2014

The basic unit of school is a classroom and the aural environment of the classrooms is essential factor for education purposes. Therefore, many efforts have been undertaken for enhancing the acoustical performance of the classrooms in many countries. As a result, acoustic criteria including reverberation time and background noise level have been established in US and UK for school classrooms depending on the usage and size of the rooms. However, in Korea, there has been little researches concerning the room acoustical investigations of the classrooms. The present study investigates the current situation of the aural environment of the 15 classrooms in Korea including elementary, middle and high schools. The acoustic criteria measured include RT, $D_{50}$, STI, SNR and background noise level. As the results, it was found that the background noise levels of the schools adjacent to roads exceed the US and UK standard of 35 dB(A). Also, most schools have so low SNR that they may be interfered by noise, which may affect speech transmission. It was also revealed that some schools have longer RT than the US standard of 0.6 s, but they all have high speech intelligibility.

• Journal of Internet Computing and Services
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• v.19 no.6
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• pp.63-72
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• 2018

The human voice is one of the easiest methods for the information transmission between human beings. The characteristics of voice can vary from person to person and include the speed of speech, the form and function of the vocal organ, the pitch tone, speech habits, and gender. The human voice is a key element of human communication. In the days of the Fourth Industrial Revolution, voices are also a major means of communication between humans and humans, between humans and machines, machines and machines. And for that reason, people are trying to communicate their intentions to others clearly. And in the process, it contains various additional information along with the linguistic information. The Information such as emotional status, health status, part of trust, presence of a lie, change due to drinking, etc. These linguistic and non-linguistic information can be used as a device for evaluating the individual's credit worthiness by appearing in various parameters through voice analysis. Especially, it can be obtained by analyzing the relationship between the characteristics of the fundamental frequency(basic tonality) of the vocal cords, and the characteristics of the resonance frequency of the vocal track.In the previous research, the necessity of various methods of credit evaluation and the characteristic change of the voice according to the change of credit status were studied. In this study, we propose a personal credit discriminator by machine learning through parameters extracted through voice.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.45 no.4
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• pp.42-50
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• 2008

The real-time bio-signal monitoring system based on the ZigBee and SIP/RTP has proposed and implemented for telemedicine but that has some problems at the stabilities to transmit bio-signal from the sensors to the other sides. In this paper, we designed and implemented a real-time bio-signal monitoring system that is focused on the reliability and efficiency for transmitting bio-signal at real-time. We designed the system to have enhanced architecture and performance in the ubiquitous sensor network, SIP/RTP real-time transmission and management of the database. The Bluetooth network is combined with ZigBee network to distribute traffic of the ECG and the other bio-signal. The modified and multiplied RTP session is used to ensure real-time transmission of ECG, other bio-signals and speech information on the internet. The modified ECG compression method based on DWLT and MSVQ is used to reduce data rate for storing ECG to the database. Finally we implemented a system that has improved performance for transmitting bio-signal from the sensors to the monitoring console and database. This implemented system makes possible to make various applications to serve U-health care services.

• Journal of Internet Computing and Services
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• v.21 no.4
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• pp.35-41
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• 2020

This paper proposes an algorithm to improve the compression performance of the adaptive multi-rate (AMR) speech coding using the context tree weighting (CTW) method. AMR is the voice encoding standard adopted by IMT-2000, and supports 8 transmission rates from 4.75 kbit/s to 12.2 kbit/s to cope with changes in the channel condition. CTW as a kind of the arithmetic coding, uses a variable-order Markov model. Considering that CTW operates bit by bit, we propose an algorithm that re-orders AMR data and compresses them with CTW. To verify the validity of the proposed algorithm, an experiment is conducted to compare the proposed algorithm with existing compression methods including ZIP in terms of compression ratio. Experimental results indicate that the average additional compression rate in AMR data is about 3.21% with ZIP and about 9.10% with the proposed algorithm. Thus our algorithm improves the compression performance of AMR data by about 5.89%.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2007.06a
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• pp.758-761
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• 2007

Digital signal processing technique is applied in wide fields such as speech processing, image processing and spectrum analysis. Therefore, in order to do frequency selective operation digital filter is used in stead of analog filter and sharp filter characteristics can be implemented. Since finite impulse response (FIR) digital filter as nonrecursive type represents linear phase response characteristics and is always stable and is used in fields regarding wave information importantly such as data transmission. And due to frequency characteristics, in order to remove the Gibbs phenomenon generating around a discontinuous point, filter is designed through window function method. Therefore, in this paper to improve performance of FIR digital filter, a modified window function was applied. And the proposed method was compared with conventional methods using peak side-lobe and transition properties in simulations.

• The Journal of the Acoustical Society of Korea
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• v.13 no.2E
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• pp.32-40
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• 1994

It is the purpose of this paper to describe the methods for establishing loudness ratings and talker echo out of transmission quality of ISDN telephone connected to fully digital network. In order to design the desirable loudness ratings and talker echo for ISDN telephone, the model system of digital speech communication for subjective tests is developed. Using this model system, opinion tests which decide the optimal CODEC input level, the range of overall loudness rating, sidetone masking rating and talker echo are performed. From the results of tests, we decided that the loudness ratings are 6 to 8dB for sending, 0 to 2dB for receiving, and 8 to 12dB for sidetone masking rating. And, the terminal coupling loss of TCLw of at least 40dB is necessary to provide echo-free telephone communications to telophone users when the overall loudness rating of ISDN telephone is normalized to 10dB.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.19 no.12
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• pp.1269-1280
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• 2009

This study was performed to construct sound performance DB of royal palace buildings and to examine the special quality more scientifically. Research target of royal palace were Changdeokgung and Gyeongbokgung. Sound insulation performance between the adjacent room and facade, room acoustics of Pyeonjeon and Chimjeon which is representative building in royal palace were examined through field measurement. Measured values of RT($T_{mf}$) at Pyeonjeon were 0.78 sec. and 1.03 sec. in Seonjeongjoen and Sajeongjoen, respectively. The RTs of both Pyeonjeon buildings were estimated suitable for speech and lecture considering their volume. The RT($T_{mf}$)s at Chimjeon were measured in range of 0.29~0.55 sec. This meant that the acoustic energy in rooms was decreased by sound transmission through mulberry paper(Hanji) of traditional windows and doors. As a sound insulation performance, the single-number quantities($D_{ls,2m,nT,w}$) of the building facades in Pyeonjeon and Chimjeon were measured 4~20 dB. Also the single-number quantities($D_{p,w}$) between the adjacent rooms in Chimjeon were measured 3~18 dB. Sound insulation performance of traditional building elements such as window and door depended strongly on their layers and area.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.12
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• pp.2653-2663
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• 1997

The successive and parallel interference cancellation schemes are two well-known types of multi-stage interference cancellation schemes using the conventional correlator receivers as a basic building block, which has been known to significantly improve the performance of DS/CDMA system in the multiple access communication. Performance comparison between these two schemes is made strictly based on the analytical and it has been shown that the successive interference cancellation (SIC) scheme is more resistant to fading than the parallel interference cancellation (PIC) scheme. We further investigate the performance of the successive IC scheme subject to the delay constraint, which may be imposed typically on most of service applications with a real-time transmission requirement, including speech and video applications. Our analysis demonstrates that the performance may be significantly improved by the groupwise successive interference cancellation (GSIC) scheme, which can be properly optimized to meet the given delay constraint.

• The Journal of the Acoustical Society of Korea
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• v.38 no.3
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• pp.275-282
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• 2019

In airport terminal, aural information is transmitted during 24 hours a day including announcement, background music and emergency control. So, clear sound is mostly necessary to transmitted to the passengers in airports. IIA (Incheon International Airport) is one of the largest airports accommodating 45 million people a year which have been built since 2001. There are currently three passenger terminals including Terminal 1 & 2, and boarding concourse. The $2^{nd}$ passenger terminal is under construction to expand the spaces which will be finished in 2020. The present work aims to explain the design principles of both architectural acoustics and electo-acoustics which have been applied to all the terminal buildings in IIA including ticketing counter, great hall, departure concourse and transportation center. Also, the acoustical performances of those spaces were examined. As a result, acoustic standards for airport were suggested. Architectural concepts for designing ceiling spaces and sound absorption treatments were suggested. Also, electro-acoustic design principles were commented.