• ETRI Journal
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• v.28 no.2
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• pp.219-222
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• 2006

Virtual source location information (VSLI) has been newly utilized as a spatial cue for compact representation of multichannel audio. This information is represented as the azimuth of the virtual source vector. The superiority of VSLI is confirmed by comparison of the spectral distances, average bit rates, and subjective assessment with a conventional cue.

• 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.

• 한국HCI학회:학술대회논문집
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• 2009.02a
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• pp.522-525
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• 2009

This paper proposed the content-based method of classifying filthy Phornographic video, which causes a big problem of modern society as the reverse function of internet. Audio data was used to extract the features from Phornographic video. There are frequency spectrum, autocorrelation, and MFCC as the feature of audio used in this paper. The sound that could be filthy contents was extracted, and the Phornographic was classified by measuring how much percentage of relevant sound was corresponding with the whole audio of video. For the experiment on the proposed method, The efficiency of classifying Phornographic was measured on each feature, and the measured result and comparison with using multi features were performed. I can obtain the better result than when only one feature of audio was extracted, and used.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.5
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• pp.283-290
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• 2004

In this paper, a low-power structure for 512 tap FIR filter in MPEG audio algorithm is proposed. By using CSD(Canonic Signed Digit) form filter coefficients and maximum sharing of input signal sample, it is shown that the number of adders of proposed structure can be minimized. To minimize the number of adders, the proposed structure utilizes the 4 steps of sharing, i.e., common input sharing, linear phase symmetric filter coefficient sharing, block sharing for common input, and common sub-expression sharing. Through Verilog-HDL coding, it is shown that reduction rates in the implementation area and relative power consumption of the proposed structure are 60.3% and 93.9% respectively, comparison to those of the conventional multiplier structure.

• Journal of Korea Multimedia Society
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• v.20 no.9
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• pp.1551-1558
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• 2017

Audio power amplifiers have been designed based on the premise that the impedance of loudspeakers is fixed at nominal 4 ohms or 8 ohms. However, it is known that the impedance varies with frequency and takes on the nominal value at some limited frequencies. The principle of the loudspeaker operation reveals that the sound pressure produced by the loudspeaker is proportional to the current flowing in the voice coil, not the voltage between the two terminals. We take the characteristics of the loudspeaker into account and compare the frequency responses of the loudspeaker in voltage-drive mode and current-drive mode via computer simulations, to conclude that the audio amplifier drive mode should be re-considered in an effort to improve the sound quality.

• Radiation Oncology Journal
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• v.33 no.3
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• pp.250-255
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• 2015

Purpose: To compare audio-only biofeedback to conventional audiovisual biofeedback for regulating patients' respiration during four-dimensional radiotherapy, limiting damage to healthy surrounding tissues caused by organ movement. Materials and Methods: Six healthy volunteers were assisted by audiovisual or audio-only biofeedback systems to regulate their respirations. Volunteers breathed through a mask developed for this study by following computer-generated guiding curves displayed on a screen, combined with instructional sounds. They then performed breathing following instructional sounds only. The guiding signals and the volunteers' respiratory signals were logged at 20 samples per second. Results: The standard deviations between the guiding and respiratory curves for the audiovisual and audio-only biofeedback systems were 21.55% and 23.19%, respectively; the average correlation coefficients were 0.9778 and 0.9756, respectively. The regularities between audiovisual and audio-only biofeedback for six volunteers' respirations were same statistically from the paired t-test. Conclusion: The difference between the audiovisual and audio-only biofeedback methods was not significant. Audio-only biofeedback has many advantages, as patients do not require a mask and can quickly adapt to this method in the clinic.

• The Journal of the Acoustical Society of Korea
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• v.28 no.3E
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• pp.109-117
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• 2009

Identity recognition in real environment with a reliable mode is a key issue in human computer interaction (HCI). In this paper, we present a robust person identification system considering score-based optimal reliability measure of audio-visual modalities. We propose an extension of the modified reliability function by introducing optimizing parameters for both of audio and visual modalities. For degradation of visual signals, we have applied JPEG compression to test images. In addition, for creating mismatch in between enrollment and test session, acoustic Babble noises and artificial illumination have been added to test audio and visual signals, respectively. Local PCA has been used on both modalities to reduce the dimension of feature vector. We have applied a swarm intelligence algorithm, i.e., particle swarm optimization for optimizing the modified convection function's optimizing parameters. The overall person identification experiments are performed using VidTimit DB. Experimental results show that our proposed optimal reliability measures have effectively enhanced the identification accuracy of 7.73% and 8.18% at different illumination direction to visual signal and consequent Babble noises to audio signal, respectively, in comparison with the best classifier system in the fusion system and maintained the modality reliability statistics in terms of its performance; it thus verified the consistency of the proposed extension.

• International Journal of Computer Science & Network Security
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• v.22 no.7
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• pp.427-442
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• 2022

MP3 is one of the most widely used file formats for encoding and representing audio data. One of the reasons for this popularity is their significant ability to reduce audio file sizes in comparison to other encoding techniques. Additionally, other reasons also include ease of implementation, its availability and good technical support. Steganography is the art of shielding the communication between two parties from the eyes of attackers. In steganography, a secret message in the form of a copyright mark, concealed communication, or serial number can be embedded in an innocuous file (e.g., computer code, video film, or audio recording), making it impossible for the wrong party to access the hidden message during the exchange of data. This paper describes a new steganography algorithm for encoding secret messages in MP3 audio files using an improved least significant bit (LSB) technique with high embedding capacity. Test results obtained shows that the efficiency of this technique is higher compared to other LSB techniques.

• Journal of Broadcast Engineering
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• v.9 no.4 s.25
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• pp.384-390
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• 2004

In this paper we compare the performance of 4 different algorithms for converting the sampling frequency of an audio from 44.1KHz to 48KHz. The algorithms considered here include the basic polyphase method. sine function based method. multi-stage method. and B-spline based method. For a fair comparison, the sampling rate converters using the 4 algorithms are redesigned under a high fidelity condition. Then, their H/W complexities are compared in terms of the computational complexity and the memory size. As a result, it is shown that the basic polyphase method and sine function based method outperform the other two in terms of the computational complexity, while the B-spline based method requires less memory than the others.

• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.187-190
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• 2002

본 논문에서는 지금까지 소개된 44.1KHz compact disc (CD)에서 48KHz digital audio tape (DAT)로의 표본화율 변환기법들에 대해서 가청 주파수 대역에서 100dB 이상의 dynamic range와 ±5x10­4dB 이하의 리플 크기를 유지할 수 있도록 각 기법들을 재설계하였으며, 메모리 요구량 및 계산량에 대해서 살펴보고자한다.