• Child Health Nursing Research
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• v.11 no.2
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• pp.167-178
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• 2005

Purpose: The purpose of this study was to describe quality of life (QOL) in Korean school-age children by identifying dimensions and attributes of QOL from the child's point of view. Method: In-depth interviews with focus questions were used for the study. Twelve children, aged 10 to 13 years, were recruited from Seoul and rural areas. The interviews were audio-taped and transcribed before content analysis. The data were analyzed for themes and attributes. The researchers read the data together and discussed their conclusions until a consensus was reached. Results: Eight dimensions, 57 subdimensions and 101atttributes were identified for QOL in school-age children. The eight dimensions of QOL were physical, social, emotional, learning, leisure, family, self-value, and material aspects. Conclusion: The study results can be utilized in developing reliable instruments to measure quality of life specific to school-age children. It is proposed that a consistent and unified policy should be established by school, family, and community for the purpose of improving the QOL of school-age children.

• The Journal of the Korea institute of electronic communication sciences
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• v.8 no.12
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• pp.1811-1818
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• 2013

As the broadband network are being deployed recently, the IPTV (Internet Protocol TV) has appeared and become a most representative service that incorporates audio, video, etc. In spite of the long-period efforts, however, no standardized method exists to measure and evaluate the quality of video that is experienced through TV in the customer's premises. Therefore, before developing a method to measure and evaluate the quality of experience (QoE) for video that is the ultimate goal, this paper reviews the outcomes from the past standardization activities and drive a few important implications that would be reflected to the method to measure and evaluate QoE for video.

• Proceedings of the Korea Contents Association Conference
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• 2009.05a
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• pp.373-377
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• 2009

This research was conducted in order to know how influential the acknowledgment of factors such as the quality of educational contents and the conducting of off-line lectures is in terms of effective learning for leaners. Based on the satisfaction-importance model of the multi-attribute attitude model, this study would like to clarify the degree to which the quality of on-line contents of on-line education and the simultaneous conducting of off-line lectures influences one's learning attitude. On-line contents satisfaction will evaluated through the three categories: audio lectures, video lecture and WBI lectures, which make up the quality of on-line contents. We would also like to do a survey on the transformation of learning attitudes when on-line and off-line lectures were conducted simultaneously.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2002.05a
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• pp.384-387
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• 2002

Internet telephony is an application that transmits voice data for conversation. Therefore it must provide high sound quality. However while audio packets are transferred through the network, they are affected by delay variations and jitters, which could result in poor sound quality if the receiving end does not have an appropriate jitter buffer to overcome network factors. This thesis introduces a buffer management algorithm that could be used to provide better sound quality for Internet phone terminals. This algorithm actively responds to both the compression algorithms that are used by the terminals, as well as to the received data to provide an improvement in sound quality. In order to confirm the validity of the suggested algorithm, comparisons of the performance have been made between the existing buffer management algorithms and this new algorithm in various network settings.

• Journal of Intelligence and Information Systems
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• v.17 no.2
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• pp.97-107
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• 2011

Nowadays there are a lot of issues about copyright infringement in the Internet world because the digital content on the network can be copied and delivered easily. Indeed the copied version has same quality with the original one. So, copyright owners and content provider want a powerful solution to protect their content. The popular one of the solutions was DRM (digital rights management) that is based on encryption technology and rights control. However, DRM-free service was launched after Steve Jobs who is CEO of Apple proposed a new music service paradigm without DRM, and the DRM is disappeared at the online music market. Even though the online music service decided to not equip the DRM solution, copyright owners and content providers are still searching a solution to protect their content. A solution to replace the DRM technology is digital audio watermarking technology which can embed copyright information into the music. In this paper, the author proposed a new audio watermarking algorithm with two approaches. First, the watermark information is generated by two dimensional barcode which has error correction code. So, the information can be recovered by itself if the errors fall into the range of the error tolerance. The other one is to use chirp sequence of CDMA (code division multiple access). These make the algorithm robust to the several malicious attacks. There are many 2D barcodes. Especially, QR code which is one of the matrix barcodes can express the information and the expression is freer than that of the other matrix barcodes. QR code has the square patterns with double at the three corners and these indicate the boundary of the symbol. This feature of the QR code is proper to express the watermark information. That is, because the QR code is 2D barcodes, nonlinear code and matrix code, it can be modulated to the spread spectrum and can be used for the watermarking algorithm. The proposed algorithm assigns the different spread spectrum sequences to the individual users respectively. In the case that the assigned code sequences are orthogonal, we can identify the watermark information of the individual user from an audio content. The algorithm used the Walsh code as an orthogonal code. The watermark information is rearranged to the 1D sequence from 2D barcode and modulated by the Walsh code. The modulated watermark information is embedded into the DCT (discrete cosine transform) domain of the original audio content. For the performance evaluation, I used 3 audio samples, "Amazing Grace", "Oh! Carol" and "Take me home country roads", The attacks for the robustness test were MP3 compression, echo attack, and sub woofer boost. The MP3 compression was performed by a tool of Cool Edit Pro 2.0. The specification of MP3 was CBR(Constant Bit Rate) 128kbps, 44,100Hz, and stereo. The echo attack had the echo with initial volume 70%, decay 75%, and delay 100msec. The sub woofer boost attack was a modification attack of low frequency part in the Fourier coefficients. The test results showed the proposed algorithm is robust to the attacks. In the MP3 attack, the strength of the watermark information is not affected, and then the watermark can be detected from all of the sample audios. In the sub woofer boost attack, the watermark was detected when the strength is 0.3. Also, in the case of echo attack, the watermark can be identified if the strength is greater and equal than 0.5.

• Journal of Broadcast Engineering
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• v.11 no.3 s.32
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• pp.363-378
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• 2006

The Digital Multimedia Broadcasting(DMB) system is developed to offer high quality audio-visual multimedia contents to the mobile environment. The system adopts MPEG-4 standard for the main video, audio and other media format. It also adopts the MPEG-4 scene description for interactive multimedia contents. The animated and interactive contents can be actualized by BIFS(Binary Format for Scene), the binary format for scene description that refers to the spatio-temporal specifications and behaviors of the individual objects. As more interactive contents are, the scene description is also needed more high bitrate. However, the bandwidth for allocating meta data such as scene description is restrictive in mobile environment. On one hand, the DMB terminal starts demultiplexing content and decodes individual media by its own decoder. After decoding each media, rendering module presents each media stream according to the scene description. Thus the BIFS stream corresponding to the scene description should be decoded and parsed in advance of presenting media data. With these reason, the transmission delay of BIFS stream causes the delay of whole audio-visual scene presentation although the audio or video streams are encoded in very low bitrate. This paper presents the effective optimization technique for adapting BIFS stream into expected MPEG-2 TS bitrate without any bandwidth waste and avoiding the transmission delay of the initial scene description for interactive DMB contents.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.16 no.4 s.95
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• pp.427-437
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• 2005

In this paper, the perceived audio level metering algorithm of digital audio sound to be able to operate in real-time is proposed. Through analyzing a conventional recommendation ITU-RBS1387-I for objective audio quality analysis, FFT-based loudness metering algorithm is implemented and the real-time method of that algorithm was advised and proved. The proposed method is based on look-up table. In order to prove the proved method, using 23 pure tones and 30 preselected digital audio samples, its performance and operation time is evaluated. Its performance, compared with an original algorithm's, have a good figure of less than $2\;\%$ error even if look-up table related with spectral spreading have large level resolution of $10\;\cal{dB}$. The proposed algorithm take only 1/21 of original algorithm's measuring time. Also, in the proposed algorithm auditory pitch group energy calculation take 1/450 of original algorithm's and excitation calculation take 1/3.57. In conclusion, the proposed algorithm is expected to be implemented into DSP-based real-time loudness meter.

• The Journal of the Acoustical Society of Korea
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• v.29 no.6
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• pp.381-387
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• 2010

Both timbral features and spatial features are important in the assessment of multichannel audio coding systems. The prediction model, extending the ITU-R Rec. BS. 1387-1 to multichannel audio coding systems, with the use of spatial features such as ITDDist (Interaural Time Difference Distortion), ILDDist (Interaural Level Difference Distortion), and IACCDist (InterAural Cross-correlation Coefficient Distortion) was proposed by Choi et al. In that model, ITDDistswere only computed for low frequency bands (below 1500Hz), and ILDDists were computed only for high frequency bands (over 2500Hz) according to classical duplex theory. However, in the high frequency range, information in temporal envelope is also important in spatial perception, especially in sound localization. A new model to compute the ITD distortions of temporal envelopes in high frequency components is introduced in this paper to investigate the role of such ITD on spatial perception quantitatively. The computed ITD distortions of temporal envelopes in high frequency components were highly correlated with perceived sound quality of multichannel audio sounds.

• The Journal of Information Systems
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• v.30 no.4
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• pp.329-347
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• 2021

Purpose The purpose of this study is to examine the review of previous studies based on audiobooks, 1) using the IPA method, we examine the quality of audiobooks that listeners perceive as important and the quality of preferred audiobooks, 2) improvement directions for activating audio content and audiobook platform, to seek ways to establish strategic plans for the audiobook market. Design/methodology/approach This study conducted and experiment using three factors(usability, information and interaction) used in web service quality(WebQual). Usability consists of the availability of time, such as the use of audiobooks for hobbies and the ease of search. Information consists of the trends of the times, topics of converation and the latest trends. Interaction consists of the proportion and depth of celebrities. After that, grid analysis was performed using the traditional IPA method and the revised IPA method. It was judged that identifying and analyzing the preference for service provision and new content was more valuable than the Performance used in the existing method. Findings According to the results of the empirical analysis, it was found that the conversation material corresponding to informativity did not affect the audiobook during the online quality analysis. It is believed that improvements can be found through content production related to entertainment, webtoons, and general common sense that can increase listeners' interest. In addition, it was found that the advertising effect of celebrities corresponding to interactivity did not affect the results of this study, which should increase the concentration and immersion of listeners by increasing the technology for audiobooks.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.8
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• pp.59-66
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• 2009

Recently, as the expectation about the IPTV (Internet Protocol TV) service quality is rapidly increased by the development of the national high-speed internet and TPS (the Triple Play Service : data + image + audio) service Therefore, the enactment of the national quality standards about the IPTV service quality guaranteeing the real time video quality of a subscriber and the international standards are hastily needed. This paper built a test bed with the network domain and the subscriber set-top box domain including the headend area and commercial network characteristic in order to test in the environment which is similar to the characteristic of the service business network. And by using the constructed environment, the characteristics required for SLA(service Level Agreement) of the IPTV service are presented as the quality test according to the service environment change.