• Journal of Korea Society of Digital Industry and Information Management
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• v.10 no.4
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• pp.161-170
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• 2014

Analog cassette tapes are easily changed due to environmental factors. To digitize is the best way to preserve a sound source. The way to digitize is to deal with the original sound to be enhanced to a variety of sources by playing through the audio card after recording. In this process to occur, it's the most important to remove various noise and equalizing sound in a method for preserving. It's studied about how to remove noise by using one of softwares, Cubase 5. 5, to remove hiss noise, which happens changing analog tape into digitalization. A amount of hiss noise is reduced to use X-Noise software of Wave which uses in Cubase 5.0, one of PLUG-IN. The noise is removed changing value of threshold and reduction every 10 times in no change of origin sound. To keep regular condition, the experiment to remove the hiss noise is conducted based on sound meondle, which is one of sound Nonmaegi. The noise is removed easily when the value of threshold is getting high in spite of giving a little value of reduction. However, as it gives a amount of reduction high, the damage rate of the sound source gets high.

• Korean Journal of Digital Imaging in Medicine
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• v.10 no.2
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• pp.17-22
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• 2008

It is to improve customer satisfaction measurement and CT Scan process without delay of examination time when is using Scan positioning time(Planning time) that time is happened always between research reactor CT examination to increase fear and examination satisfaction by the customer's comprehension tribe which get the latest contrast enhancement CT examination. Needs and interests that customer wants to compose visual and auditory Contents to be played to Scan positioning time did questionnaire about curiosity later before CT examination to 600 people for October - November 2 months of 2006 to customer whole that get CT examination on source. Data getting through questionnaire investigated examination comprehension and satisfaction through questionnaire after experiment Scan Positioning to 500 coming to help customers to be source CT examination for 3 months February December - 2007 year in 2006 manufacturing Voice and Visual announce media for reference. To customer who interest degree appeared, and answers preparatory audit from preparatory audit about curiosity of CT examination customer to order of examination time required(43%), contrast media side effect(26%), examination region(20%), breath(10%), etc..(1.5%) audio-visual materials in questionnaire that attain after do reclamation among examination age, sex, reception type of irrelatively in 91% of target increase of hailing degree and examination satisfaction appear. Searched that customer hailing and satisfaction are increased greatly when use of audio-visual materials in satisfaction result that use CT Positioning delay time. In experiment process, It took lacking part by method that use hearing in case of do not use sight as is unavoidable in subject position or old age. Through this, audio-visual materials could analogize that it is more useful method that use sight and hearing at the same time.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.3 s.303
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• pp.31-40
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• 2005

In this paper, we develop an approach toward JSC(Joint Source-Channel Coding) method for MPEG-4 based FGS(Fine Granular Scalability) video coding and transmission in fixed and mobile receiving environment(Digital Audio Broadcasting, DAB). The source coder used MPEG-4 FGS video codec, the channel coder used RCPC(Rate Compatible Punctured Convolution) code and the modulation method used QPSK modulation. We have considered channel environment of AWGN and mobile receiving environment. This study determined optimum Trade-off point between source bit rate and channel coding rate in variable channel states. We compared FGS-JSC method and general single layer CBR(Constant Bit Rate) transmission. In this results, FGS-JSC was appeared better performance than CBR transmission.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.16 no.5
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• pp.165-172
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• 2016

This paper presents an implementation of a live audio streaming system using the Raspberry Pi 3 embedded computer. This system is a live streaming system not file-based streaming. This is a push streaming system which converts the incoming analog audio signal to digital samples and broadcasts them to multiple connected users concurrently. Since the server software is developed in Java language, it can be installed on any other embedded computers without any modification. We concluded that ESCatcher can service live streaming about 60 users concurrently through calculations and experiments, And also we achieved the delay time of a little bit more than 40ms between arrival of audio source and play on the android device.

• The KIPS Transactions:PartA
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• v.15A no.6
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• pp.317-324
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• 2008

In this paper, we propose a system which easily controls the existing serially connected multi-channel speakers in a general personal computer by using a USB(Universal Serial Bus) interface. The personal computer as a host of the USB interface analyzes a sound source and sends audio data in a real-time fashion by the use of the isochronous transmission, one of four transmission methods provided by the USB interface. In addition, a channel is assigned by means of the bulk transmission, one of four transmission methods provided by the USB interface. Transmitted data from the USB host are sent to each speaker through compression and packet generation process. Each speaker detects corresponding digital data and regenerates audio signals through DAC(Digital-to-Analog Converter). A user can easily select a sound source file and a channel by the use of a GUI environment in a personal computer.

• Journal of Digital Convergence
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• v.9 no.5
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• pp.225-233
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• 2011

Internet telephony is an Internet service which supports voice telephone using VoIP technology on the IP-based Internet. It has some advantages in that voice telephone services can be accompanied with multimedia services such as video communication and messaging services. Recently, the introduction of smart phones has led to a growth in social networking services and thus, the research and development of Internet telephony has been actively progressed and has the potential to become a replacement for the telephone service that is currently being used. In this paper we designed and implemented a recording system which records voice data of SIP-based Internet telephone's voice calls. It is developed on the linux system and has some features such as audio mixing of two in/out voice channels, live packet sniffing, and the ability to transfer mixed audio files to the log file server. These functions are implemented using various open source softwares. Afterwards, this VoIP recording system will be applied as a base technology to advanced services like a VoIP-based call center system.

• Journal of Korea Society of Industrial Information Systems
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• v.19 no.2
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• pp.73-83
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• 2014

Thanks to the development of diverse audio editing Technology, audio file can be easily revised. As a result, diverse social problems like forgery may be caused. Digital forensic technology is actively studied to solve these problems. In this paper, a hand-held device identification method, an area of digital forensic technology is proposed. It uses the noise features of devices caused by the design and the integrated circuit of each device but cannot be identified by the audience. Wiener filter is used to get the noise sounds of devices and their acoustic features are extracted via MIRtoolbox and then they are trained by multi-layer neural network. To evaluate the proposed method, we use 5-fold cross-validation for the recorded data collected from 6 mobile devices. The experiments show the performance 99.9%. We also perform some experiments to observe the noise features of mobile devices are still useful after the data are uploaded to UCC. The experiments show the performance of 99.8% for UCC data.

• Journal of Satellite, Information and Communications
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• v.9 no.4
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• pp.69-72
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• 2014

Recently, lighting systems using audio signal of audible frequency and frequency spectrum of visible lighting are studied. And various related products are being sold and released commercially. Also demands of emotional matching algorithm and system which includes effective and methodical designs are being increased. And the importance related with this scheme has increased. In this Paper, we configures a system for digital sound source based LED color control. And we develop algorithm to control LED color for the system configuration. Also we demonstrated the usefulness of the algorithm through experiment with simulation using LED color control system. We expected to be useful in a variety of fields and applications using proposed digital music based LED color control system.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.6
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• pp.180-187
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• 2013

In this paper, we propose a LED communication based multi-hop audio data transmission network system. The main contribution and features of the proposed system are as follows. First, the contribution of this research is to develope the LED communication based multi-hop transmission network system which can transmit audio data signal with long distance via multi-hops. Second, the developed system has the following features: In transmitter, audio data is transmitted after encoding with S/PDIF format via a general LED. The relay receives digital audio signal by using photo diode and then transmits the signal to receiver after error checking and amplifying. The receiver receives the encoded audio data via photo diode and then converts to analog audio signal by using decoding and amplifying. The performance evaluation of the proposed system is conducted in the laboratory with fluorescent light source. The results of the performance evaluation confirm that the system can provide high quality audio transmission from transmiter to receiver via multi-hop relays in a long distance while we can see there are differences in the transmitted audio quality according to the used LED colors.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.10A
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• pp.951-961
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• 2002

In general, there is discontinuity phenomenon like a black screen and an irregular sound for repeated play of a finite digital stream. In this paper, for repeated play we analyze the relation between source and stream time causes this phenomenon. We obtain the time relation between video frame rate, audio frame rate, and TS packet transmission rate to eliminate this phenomenon. Using this time relation, we propose a new generation method of elementary stream (ES) and transport stream (TS) to eliminate discontinuity phenomenon. The test results of the generate ES and TS using the proposed method show that the discontinuity phenomenon can be eliminated for repeated play of a finite proposed stream.