• Korean Journal of Adult Nursing
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• v.25 no.2
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• pp.231-239
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• 2013

Purpose: This study was conducted to identify the effects of video-audio information provision on physical discomfort, anxiety and nursing satisfaction of the clients for gastroscopy. Methods: The study design was nonequivalent control group pre-post test design. The subjects were 50 patients who visited H hospital health examination center for gastroscopy. Video-audio information developed by the authors was used as educational material for the treatment group. The data were collected between September 15 and November 15, 2010. The study instruments were the State-Trait Anxiety Inventory, the Physical Discomfort Scale, and the Nursing Satisfaction Scale. Results: The level of anxiety and physical discomfort in the treatment group were not significantly different from that in the comparison group (t=-0.28, p=.781; t=-0.34, p=.741). The level of clients' satisfaction with nursing care in the treatment group was significantly higher than in the comparison group (t=-4.12, p<.001). Conclusion: Use of video-audio information was effective in the increase in satisfaction with care. Therefore, it could be useful in the nursing practice, and be utilized as a way of nursing intervention to improve nursing satisfaction.

• Journal of the Korea Society of Computer and Information
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• v.22 no.10
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• pp.63-71
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• 2017

In this paper, we develop the system for audio volume equalization of video on demand(VoD) content and propose the solution for it. In recent years, there has been a steady increase in the number of VoD users in addition to linear channels. However, viewers ought to sit in an uncomfortable way, adjusting the volume intermittently while they are broadcasted. Sudden changes of volume occur between the broadcasting channels, the programs from the co-channel, or the linear channels and the VoDs. Especially, upsurged dissatisfaction from the televiewers has been found due to the unequalized volume when shifting between the linear channel and the VoD. In order to solve this problem, multilateral efforts were put forth, such as a system for keeping the volume at a certain level in digital broadcasting program has been legislated domestically. It leads success in equalizing linear channel volume. On contrary, too little notice has been taken for distorted volume problem of video on demand(VoD) content. In this paper, we developed and applied the volume equalization system into VoD content to achieve uniformization, a similar condition with linear channel(-24LKFS). This suggestion helped uneven current of volume which was in the stage -16 ~ -20LKFS to stable condition by lowering into the stage of -24LKFS. It also brought 20% increase in perspective of volume quality satisfaction level.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.6
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• pp.1433-1442
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• 1998

The thesis presents a dual audio decoder which can decode both AC-3 and MPEG-2 bitstream. The MPEG-2 synthesis processi s optimized via FFT to establish the common data path with AC-'3s. A dual audio decoder consists of a DSP core which performs the control-intensive part of each algorithm and a common synthesis filter which perfomrs the computation-intensive part. All the components of the dual audio decoder have been described in VHDL and simulated with a SYNOPSYS tool. The software modeling of the DSP core was used for functional validation. After being synthesized using 0.6 .mu.m-3ML technology standard cell, the dual audio decoder was simulated at gate-level with a COMPASS tool for hardware validation.

• JSTS:Journal of Semiconductor Technology and Science
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• v.5 no.4
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• pp.249-254
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• 2005

This paper presents a system-on-a-chip (SOC) design for digital TV. The single LSI incorporates almost all essential parts such as CPU, ISO/IEC 11172/13818 system/audio/video decoders, a video post-processor, a graphics/OSD processor and a display processor. It has analog IP's inside such as video DACs, an audio PLL, and a system PLL to reduce the system-level implementation cost. Descramblers and Smart Card interface are included to support widely used conditional access systems. The video decoder can decode two video streams simultaneously. The DSP-based audio decoder can process various audio coding specifications. The functional blocks for video quality enhancement also form outstanding features of this SoC. The SoC supports world-wide major DTV services including ATSC, ARIB, DVB, and DIRECTV.

• The Journal of the Acoustical Society of Korea
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• v.28 no.8
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• pp.832-842
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• 2009

Automatic speech recognition (ASR) has been successfully applied to many real human computer interaction (HCI) applications; however, its performance tends to be significantly decreased under noisy environments. The invention of audio visual speech recognition (AVSR) using an acoustic signal and lip motion has recently attracted more attention due to its noise-robustness characteristic. In this paper, we describe our novel integration scheme for AVSR based on a late integration approach. Firstly, we introduce the robust reliability measurement for audio and visual modalities using model based information and signal based information. The model based sources measure the confusability of vocabulary while the signal is used to estimate the noise level. Secondly, the output probabilities of audio and visual speech recognizers are normalized respectively before applying the final integration step using normalized output space and estimated weights. We evaluate the performance of our proposed method via Korean isolated word recognition system. The experimental results demonstrate the effectiveness and feasibility of our proposed system compared to the conventional systems.

• Transactions on Electrical and Electronic Materials
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• v.9 no.3
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• pp.123-127
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• 2008

The paper is to solve the problem of existing audio signal transfer system which has a difficulties of system organization and the increase of additional install cost and unfriendly interior. To solve the existing system, we drew the new audio signal transfer system based on PLC and evaluated it. A transmitter and a receiver were designed using the PLC chip INT5500CS. An audio signal transfer system was configured with a CD player to which audio signals are sent from the transmitter and a speaker connected to the receiver. For performance evaluation of this system, a USBPre external sound card and Smaart Live 5 which is a PC-based sound measuring program were added. As a result of our experiment, the measured signal level is $2{\sim}3$ dB lower than reference signal, latency is 16.69 ms, and the specific character of coherency is bad in high frequency band. Otherwise, this system transmits and receives signals over 90 % in good condition as a result of measuring pink noise, frequency (1 kHz), and phase, magnitude. In view of the result so far achieved, the system designed this study has excellent performance, it resolves defect of existing audio signal transfer system.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.43 no.3 s.309
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• pp.76-83
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• 2006

A main objective of audio equalizer is for user to tailor acoustic frequency response to increase sound comfort and example applications of audio equalizer includes large-scale audio system to portable audio such as mobile MP3 player. Up to now, all the audio equalizer requires manual setting to equalize frequency bands to create suitable sound quality for each genre of music. In this paper, we propose an intelligent audio graphic equalizer system that automatically classifies the music genre using music content analysis and then the music sound is boosted with the given frequency gains according to the classified musical genre when playback. In order to reproduce comfort sound, the musical genre is determined based on two-step hierarchical algorithm - coarse-level and fine-level classification. It can prevent annoying sound reproduction due to the sudden change of the equalizer gains at the beginning of the music playback. Each stage of the music classification experiments shows at least 80% of success with complete genre classification and equalizer operation within 2 sec. Simple S/W graphical user interface of 3-band automatic equalizer is implemented using visual C on personal computer.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.48 no.6
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• pp.56-61
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• 2011

A class-D audio amplifier is widely used in coil speaker. This paper presented the technique for applying a class-D audio amplifier to the ceramic flat speaker. This technique can be achieved by employing a matching transmitter in order to replace class-G amplifier that is drven by voltage level to class-D amplifier employing power driving method. Consequently, the presented technique can improve the efficiency by making the voltage driving level a litter larger. We evaluate the sound-level efficiency using the various mediums such as wood, plastic, and paper. From the simulation results, the proposed technique employing a class-D audio amplifier rather than a class-G one showed a 10% improvement. The proposed system can be applicable for the mobile appliances as an external slim speaker.

• Proceedings of the IEEK Conference
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• 2000.06b
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• pp.173-176
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• 2000

Silicon semiconductor technology agree that the number of transistors on a chip will keep growing exponentially, and it is pushing technology toward the System-On-Chip. In SoC Design, Specification at system level is key of success. Executable Specification reduce verification time. This Paper describe the design of IMDCT for MPEG Audio Decoder employing system-level design methodology and Executable Specification Methodology in the VHDL simulator with FLI environment.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.82-88
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• 2002

In this paper, we proposed a audio indexing method that is used wavelet transform for audio data retrieval. It is difficult for audio data to make a efficient audio data index because of its own particular properties, such as requirement of large storage, real time to transfer and wide bandwidth. An audio data in del using wavelet transform make it possible to index and retrieval by using the particular wavelet transform properties. Our proposed indexing method doesn't separate data to several blocks. Therefore we use both high-pass and low-pass parts of last level coefficient of wavelet transform. Audio data indexing is made by applying the string matching algorithm to high-pass part and zero-crossing histogram to low-pass part. These are transformed to the continued strings, Through this method, we described a retrieval efficiency. The retrieval method is done by comparing the database index string to the query string and then data of minimum values is chosen to the result. Our simulation decided proper comparative coefficient and made known changing of retrieval efficiency versus audio data length. The results show that the proposed method improves retrieval efficiency compared to conventional method.