• Journal of Korea Multimedia Society
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• v.15 no.5
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• pp.651-663
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• 2012

Networked digital signage downloads necessary multimedia contents from a large-sized storage on WAN to its local disk of a limited size before starting their playback. If the required time to download the entire contents gets longer, a start time to play the contents at the digital signage could be delayed. In this paper, we propose a hybrid storage architecture that not only inserts an iSCSI storage layer between the existing local disk and the WAN storage, but offers a contents caching scheme in order to obtain all the necessary contents in digital signage rapidly. The proposed caching scheme determines how to place the downloaded contents both in the local disk and the iSCSI storage. Uniquely, the proposed caching scheme manages the iSCSI storage space by dividing it into two regions: (1) in one region, the digital signage can play the contents directly without downloading them into the local disk; (2) in the other region, the digital signage cannot. Performance evaluations on a simulator and an actual system with workloads of various contents show that a contents-downloading time of the hybrid storage architecture is at maximum three times shorter than that of the existing storage architecture.

• Journal of the Korean Society of Fisheries and Ocean Technology
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• v.14 no.2
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• pp.57-62
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• 1978

When fisherman use the boat seine net to catch anchovy, a large noise (drum can, small drum and small gong) is used to scare the anchovy school along the wing nets, and into the bag net were they are caught. We want to know how much of an effect these s:mnds have on forceing the anchovy school towards the bag net. The underwater sounds of ancho\'y, drum can, small drum and small gong were analyzed in the labroatory. The behavioral responeses to the playback sounds of anchovy feeding and sounds of artificial instruments were also investigated. The feeding and artificial sounds of the samples were recorded by a tape recorder through a hydrophone in an anechoic aquarium. The sound intensity level was measured by means of a sound level meter in an anechoic chamber. The frequency and intensity of various sounds were analyzed with an analyzing system consisting of a ~-octave filter set, a high speed level recorder, an amplifier and an oscilloscope. The most successful recording was edited into a 9 to 10 second sound track and was repeated in a sequence of 9 to 10 second intervals. The sequence was then reproduced into an anechoic aquarium through the underwater speaker. The results of investigation are as follows; 1. The frequency of the feeding sound was 63~80Hz, and the pressure level produced was less than 32db. 2. The frequencies of the artificial sounds were 315~ 1,OOOHz, and the pressure levels were 88~95 db in the air. 3. When a hydrophone was placed 70cm below the surface with artificial sounds (drum can, small drum and small gong) produced 1 meter above the surface, the pressure level decreased about 30db. 4. The feeding sound was ineffective in attracting the anchovy, because of interference from ambient noise. 5. The artificial sounds had such a small effect on the anchovy's that they could not be used in ocean fisheries.

• The KIPS Transactions:PartA
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• v.15A no.1
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• pp.53-60
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• 2008

Streaming media has contributed to a significant amount of today's Internet Traffic. Like traditional web objects, rich media objects can benefit from proxy caching, but caching streaming media is more of challenging than caching simple web objects, because the streaming media have features such as huge size and high bandwidth. And to support various bandwidth requirements for the heterogeneous ubiquitous devices, a transcoding proxy is usually necessary to provide not only adapting multimedia streams to the client by transcoding, but also caching them for later use. The traditional proxy considers only a single version of the objects, whether they are to be cached or not. However the transcoding proxy has to evaluate the aggregate effect from caching multiple versions of the same object to determine an optimal set of cache objects. And recent researches about multimedia caching frequently store initial parts of videos on the proxy to reduce playback latency and archive better performance. Also lots of researches manage the contents with segments for efficient storage management. In this paper, we define the 9-events of transcoding proxy using 4-atomic events. According to these events, the transcoding proxy can define the next actions. Then, we also propose the segment-based caching policy for the transcoding proxy system. The performance results show that the proposing policy have a low delayed start time, high byte-hit ratio and less transcoding data.

• Journal of the Institute of Convergence Signal Processing
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• v.12 no.4
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• pp.333-340
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• 2011

KJJVC(Korea-Japan Joint VLBI Correlator) was installed at the KJCC(Korea-Japan Correlation Center) and has been operated by KASI(Korea Astronomy and Space Science Institute) from 2009. KJNC is able to correlate the VLBI observed data through KVN(Korean VLBI Network), VERA(VLBI Exploration of Radio Astrometry), and JVN(Japanese VLBI Network) and its joint network array. And it is used exclusively as computer in order to process the observed data for the scientific purpose KJJVC used the VSI(VLBI Standard Interface) as the VLBI international standard at the data input-output specification between each component. Especially, for correlating the observed data, the data is transmitted with 1024Mbps speed between Mark5B high-speed playback and RVDB(Raw VLBI Data Buffer). The EMI(Electromagnetic lnterference), which is occurred by data transmission with high-speed, cause the data loss and the loss occurrence is frequently often for long transmission cable. Finally it will be caused the data recognition error by decreasing the voltage level of digital data signal. In this paper, in order to minimize the data loss by measuring the EMI noise level in transmission of the VSI specification, the 3 methods such as 1) RC filtering method, 2) lmpedance matching using Microstrip line, and 3) Signal buffering method using Differential line driver, were proposed. To verify the effectiveness of each proposed method, the performance evaluation was conducted by implementing and simulations for each method. Each proposed method was effectively confirmed as the high-speed data transmission of the VSI specification.

• Journal of KIISE:Information Networking
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• v.33 no.4
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• pp.333-342
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• 2006

The cdma2000 lxEV - DO mobile communication system provides broadcast and multicast services (BCMCS) to meet an increasing demand from multimedia data services. The servicing of video streams over a BCMCS network must, however, face a challenge from the unreliable and error-prone nature of the radio channel. The BCMCS network uses Reed-Solomon coding integrated with the MAC protocol for error recovery. We analyze this coding technique and show that it is not effective in the case of slowly moving mobiles. To improve the playback quality of an MPEG-4 FGS video stream, we propose the Hybrid error recovery scheme, which combines Reed-Solomon with ARQ, using slots which are saved by reducing the Reed-Solomon coding overhead. The target packets to be retransmitted are prioritized by a utility function to reduce the packet error rate in the application layer within a fixed retransmission budget. This is achieved by considering of the map of the error control block at each mobile node. The proposed Hybrid error recovery scheme also uses the characteristics of MPEG-4 FGS (fine granularity scalability) to improve the video quality even when conditions are adverse: slow-moving nodes and a high error rate in the physical channel.

• Journal of Broadcast Engineering
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• v.13 no.1
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• pp.6-16
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• 2008

An instable wireless channel condition causes more packet losses and retransmissions due to interference, fading, station mobility, and so on. Therefore video streaming service over a wireless networks is a challenging task because of the changes in the wireless channel conditions and time-constraints characteristics of the video streaming services. To provide efficient video streaming over a wireless networks, QoS-enhanced MAC protocol, IEEE 802.11e, is standardized recently. Tn this paper, we propose a new network-adaptive H.264 video streaming mechanism in the IEEE 802.11e wireless networks. To improve the quality of video streaming services, video stream has to adapt to the changes in the wireless channel conditions. The wireless channel conditions are estimated by the packet loss probability and informed to the application layer by the cross-layering. According to the wireless channel information, the video streaming application filters out the low-priority data. This adaptation mechanism efficiently uses system resources because it drops the low-priority data in advance. Therefore, our cross-layer design can provide improved video streaming services to the end-user. Through the implementation and performance evaluation, we prove that the proposed mechanism improves the QoS of the video streaming by providing the smoothed playback.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.8A
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• pp.739-752
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• 2002

Recently, to utilize the limited bandwidth effectively, the concept of partial response (PR) signaling has widely been adopted in both the high-speed data transmission and high-density digital recording/playback systems such as digital microwave, digital subscriber loops, hard disk drives, digital VCR's and digital versatile recordable disks and so on. This paper is concerned with adaptive equalization of partial response channels particularly for the magnetic recording channels. Specifically we study how the PR channel equalizers work for different choices of desired or reference signals used for adjusting the equalizer weights. In doing so, we consider three different configurations that are actually implemented in the commercial products mentioned above. First of all, we show how to compute the theoretical values of the optimum Wiener solutions derived by minimizing the mean-squared error (MSE) at the equalizer output. Noting that this equalizer MSE measure cannot be used to fairly compare the three configurations, we propose to use the data MSE that is computer just before the final detector for the underlying PR system. We also express the data MSE in terms of the channel impulse response values, source data power and additive noise power, thereby making it possible to compare the performance of the configurations under study. The results of extensive computer simulation indicate that our theoretical derivation is correct with high precision. Comparing the three configurations, it also turns out that one of the three configurations needs to be further improved in performance although it has an apparent advantage over the others in terms of memory size when implemented using RAM's for the decision feedback part.

• Journal of Korea Entertainment Industry Association
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• v.15 no.4
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• pp.325-335
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• 2021

Chronic diseases have been increasing recently as the average life expectancy of humans has been extended, and this trend has caused problems such as the widespread demand for health and rehabilitation services and rising medical costs. In order to solve this problem, the community-based rehabilitation has been developed and strengthened in Korea and gradually promoted since 2019. It is important to secure access to clients who want to use services to revitalize community-based rehabilitation. So in this study, as part of the community-based rehabilitation, intends to devise smartphone applications designs and develop a prototype to secure access to community-based occupational therapy services based on occupation and activities. For Occupational Therapy Practice Framework (OTPF), International Classification of Functioning, Disability and Health (ICF), and Allen Diagnostic Module-2 (ADM-2) were used to devise and categorize occupation and activity based application content, and link OTPF, ICF, and ADM-2 through prior research analysis and expert meetings. The derived content was visualized through literature review and activity analysis, and was implemented to enable direct playback within the application using the YouTube API, and finally developed a prototype application. The Android Studio 3.5.2 for Windows 64-bit was used to build the application prototype. In further research, converging various digital technologies for user convenience and additionally researching community-based occupational therapy service providers opinions and service user satisfaction will improve accessibility to community-based occupational therapy services for clients who have difficulty occupational performance in the community.

• Journal of the Korean Society of Fisheries and Ocean Technology
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• v.14 no.1
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• pp.15-36
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• 1978

Underwater sounds of some fishes and crabs were analyzed in the laboratory. The behavioral responses to the playback sounds of their feeding and croaking sound were investigated. The samples used in the experiment were as follows: Nibea albiflora, seriola quinqueradiata, Navodon modestus, Fugu xanthopterus, chrysophrys major, Scylla serrata, Telmessus acutidens, Charybdis japonica, and Portunus trituberculatus. The feeding and croaking sounds of the samples were recorded by a tape recorder through a hydrophone in an anechoic aquarium. The sound intensity level was measured by means of a sound level meter at an anechoic chamber. The frequency, intensity and wave form of various sounds were analyzed with an analyzing system consisting of a 1/3 octave filter set, a high speed level recorder, an amplifier, an octave band analyzer and an oscilloscope. The most successful recording was edited into a sequence of sound track which repeats sound emitting for 5 to 7 seconds after pausing for 5 to 7 seconds. The sequence was then reproduced into an anechoic aquarium through the under water speaker. The experimental anechoic aquarium used for the sample fishes was divided into the four sections with any three screens selected from 40$\times$40mm, 60$\times$60mm, 80$\times$80mm and 100$\times$100mm mushes according to the species of the fishes, besides that for crabs were not sectioned. The results of the investigation are as follows: 1. Of the feeding sound of fish, the frequency of wave from of the sound produced by Nibea albiflora and seriola quinqucradiata was 125～250Hz, that by Navodon modestus 63～125Hz, and that by Fugu xanthopterus 400～500Hz. The pressure level of the feeding sound produced by Nibea albiflora and Seriola quinqueradiata was 56～62db, that by Navodon modestus 57～59db, and that by Fugu xanthopterus 60～64db. 2. Of the croaking sound of Nibea albiflora, the frequency of the sound was 125～250Hz almost equivalent to that of feeding sound, and the pressure level was 62～63db, slightly higher than that of feeding sound. 3. Of the croaking sounds of crabs, the frequency of the sound produced by scylla serrata was 125～250Hz, that by Charybdis japonica and Telmessus acutidens 500～1,000Hz, and that by Portunus trituberculatus 250～500Hz. The pressure level of the croaking sound by Scylla serrata was 68～70db, and that by Charybdis japonica, Telmessus acutidens and Portuens trituberculatus 50～62db. 4. Phonotactic responses of Nibea albiflora and Seriola quinqueradiata to the feeding sounds produced by their own species, the same body length were conspicuous with the phonotactic index of 56～87%, but that of Navodon modestus, Chrysophrys major and Fugu xanthopterus were hardly recognized. 5. Phonotactic responses of the sample fishes to the sinusoidal sound with the frequency range of 50 to 9,000 Hz were observed not conspicuous. 6. Phonotactic responses of Portunus trituberculatus to the croaking sounds produced by their own species was varied in the range of 40～100%, according to the carapace length and the sex.

• Journal of Intelligence and Information Systems
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• v.24 no.2
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• pp.21-35
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• 2018

Since the introduction of MP3 players, CD recordings have gradually been vanishing, and the music consuming environment of music users is shifting to mobile devices. The introduction of smart devices has increased the utilization of music through music playback, mass storage, and search functions that are integrated into smartphones and tablets. At the time of initial MP3 player supply, the bitrate of the compressed music contents generally was 128 Kbps. However, as increasing of the demand for high quality music, sound quality of 384 Kbps appeared. Recently, music content of FLAC (Free License Audio Codec) format using lossless compression method is becoming popular. The download service of many music sites in Korea has classified by unlimited download with technical protection and limited download without technical protection. Digital Rights Management (DRM) technology is used as a technical protection measure for unlimited download, but it can only be used with authenticated devices that have DRM installed. Even if music purchased by the user, it cannot be used by other devices. On the contrary, in the case of music that is limited in quantity but not technically protected, there is no way to enforce anyone who distributes it, and in the case of high quality music such as FLAC, the loss is greater. In this paper, the author proposes an audio watermarking technology for copyright protection of high quality stereo music. Two kinds of information, "Copyright" and "Copy_free", are generated by using the turbo code. The two watermarks are composed of 9 bytes (72 bits). If turbo code is applied for error correction, the amount of information to be inserted as 222 bits increases. The 222-bit watermark was expanded to 1024 bits to be robust against additional errors and finally used as a watermark to insert into stereo music. Turbo code is a way to recover raw data if the damaged amount is less than 15% even if part of the code is damaged due to attack of watermarked content. It can be extended to 1024 bits or it can find 222 bits from some damaged contents by increasing the probability, the watermark itself has made it more resistant to attack. The proposed algorithm uses quantization in DCT so that watermark can be detected efficiently and SNR can be improved when stereo music is converted into mono. As a result, on average SNR exceeded 40dB, resulting in sound quality improvements of over 10dB over traditional quantization methods. This is a very significant result because it means relatively 10 times improvement in sound quality. In addition, the sample length required for extracting the watermark can be extracted sufficiently if the length is shorter than 1 second, and the watermark can be completely extracted from music samples of less than one second in all of the MP3 compression having a bit rate of 128 Kbps. The conventional quantization method can extract the watermark with a length of only 1/10 compared to the case where the sampling of the 10-second length largely fails to extract the watermark. In this study, since the length of the watermark embedded into music is 72 bits, it provides sufficient capacity to embed necessary information for music. It is enough bits to identify the music distributed all over the world. 272 can identify $4*10^{21}$, so it can be used as an identifier and it can be used for copyright protection of high quality music service. The proposed algorithm can be used not only for high quality audio but also for development of watermarking algorithm in multimedia such as UHD (Ultra High Definition) TV and high-resolution image. In addition, with the development of digital devices, users are demanding high quality music in the music industry, and artificial intelligence assistant is coming along with high quality music and streaming service. The results of this study can be used to protect the rights of copyright holders in these industries.