• Journal of the Korean Institute of Telematics and Electronics
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• v.26 no.2
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• pp.118-123
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• 1989

In this paper, as we have integrated SCF for voice signal processing using CMOS circuit with the low power dissipation and the easy circuit design, it has been presented the simplified CMOS OP AMP design method with ${\pm}$5V pwoer source in order to use together with digital part. After an example about SCF CMOS OP AMP design, it has been performed layout appling channel width and length obtained by design method, and then its characteristics were simulated by SPICE 2G program. Therefoe, this design method will be applied the general CMOS OP AMP design in the electronic circuit.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.9A
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• pp.2365-2371
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• 1998

Discrete cosine transform (DCT) has wide applications in speech and image coding. In this paper, we propose a novel fast dCT scheme with the property of reduced multiplication stages and the smaller number of additions and multiplications. This exploits the symmetry property of the DCT kernel to decompose the N-point dCT to N/2 point, and can be generally applied recursively to $2^{m}$-point. The proposed algorithm has a structure that most of multiplications tend to be performed at final stage, and this reduces propagation of truncation error which could occur in the fixed-point computation. Also the minimization of the multiplication stages further decreases the error.

• Proceedings of the KIEE Conference
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• 1996.07b
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• pp.1383-1385
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• 1996

A network based interactive magnetic resonance imaging (MRI) system has been developed using the World Wide Web. For this purpose, an HTTP server is developed on the host computer of the MRI system. Capabilities of video and audio conferencing are included for monitoring experiment. Using the developed system. MRI imaging has been successfully carried out at the Signal Processing Lab in the Kwangwoon University with the remote MRI system located at the Medical Image Research Center at the KAIST in Daejon.

• Proceedings of the KOSOMBE Conference
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• v.1996 no.05
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• pp.139-142
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• 1996

A remotly controlled interactive magnetic resonance imaging (MRI) has been tried using the World Wide Web. For this purpose, an HTTP server is developed on the host computer of the MRI system. Video and audio conferencing capability is also included for the experiment. Using the developed system, MRI imaging has been successfully carried out at the Signal Processing Lab in the Kwangwoon University with the remote MRI system located at the Medical Image Research Center in the KAIST in Daejon.

• Journal of Korea Multimedia Society
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• v.6 no.2
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• pp.319-324
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• 2003

In this paper, a new wideband tunable analog element for multimedia system is proposed. The proposed active element is composed of the complementary cascode circuit which can extend transconductance of an element. Therefore, the unity gain frequency which is determined transconductance is increased than that of the conventional element. And then these results are verified by the 0.22$\mu\textrm{m}$ CMOS n-well parameter simulation. As a result, the gam and the unity gam frequency are 42dB and 200MHz on 2V supply voltage. And power dissipation of the designed element is 0.32mW.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.9
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• pp.1672-1678
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• 2007

The communication systems and characteristics of transmit are very importance a serious view of ubiquitores society. The high rate systems have to construct for data of voice, graphics and moving-video with the multimedia to transmit on the low frequency band. The rate of data are relation that there are the bandwidth of line and channels, the bit per second and communication-systems. This paper study for the error performance of 2, 4 8 level FSK to transmit amount of data in the communication systems and design to FDM system.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2022.05a
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• pp.41-43
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• 2022

In order to prevent and block infectious diseases caused by the recent COVID-19 pandemic, non-contact biometric information acquisition and analysis technology is attracting attention. The invasive and attached biometric information acquisition method accurately has the advantage of measuring biometric information, but has a risk of increasing contagious diseases due to the close contact. To solve these problems, the non-contact method of extracting biometric information such as human fingerprints, faces, iris, veins, voice, and signatures with automated devices is increasing in various industries as data processing speed increases and recognition accuracy increases. However, although the accuracy of the non-contact biometric data acquisition technology is improved, the non-contact method is greatly influenced by the surrounding environment of the object to be measured, which is resulting in distortion of measurement information and poor accuracy. In this paper, we propose a context-based bio-signal modeling technique for the interpretation of personalized information (image, signal, etc.) for bio-information analysis. Context-based biometric information modeling techniques present a model that considers contextual and user information in biometric information measurement in order to improve performance. The proposed model analyzes signal information based on the feature probability distribution through context-based signal analysis that can maximize the predicted value probability.

• The Journal of the Acoustical Society of Korea
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• v.28 no.6
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• pp.572-577
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• 2009

Large corpus-based concatenating Text-to-Speech (TTS) systems can generate natural synthetic speech without additional signal processing. To prune the redundant speech segments in a large speech segment DB, we can utilize a decision-tree based triphone clustering algorithm widely used in speech recognition area. But, the conventional methods have problems in representing the acoustic transitional characteristics of the phones and in applying context questions with hierarchic priority. In this paper, we propose a new clustering algorithm to downsize the speech DB. Firstly, three 13th order MFCC vectors from first, medial, and final frame of a phone are combined into a 39 dimensional vector to represent the transitional characteristics of a phone. And then the hierarchically grouped three question sets are used to construct the triphone trees. For the performance test, we used DTW algorithm to calculate the acoustic similarity between the target triphone and the triphone from the tree search result. Experimental results show that the proposed method can reduce the size of speech DB by 23% and select better phones with higher acoustic similarity. Therefore the proposed method can be applied to make a small sized TTS.

• Journal of the Korea Computer Industry Society
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• v.5 no.5
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• pp.737-746
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• 2004

Two-channel multiwavelet system is investigated for image compression application in this paper. Generally, multiwavelets are known for their superb capability of compressing non-stationary signals like voice. However, multivavelet system have a critical problem in processing and compressing image data due to mesh-grid visual artifacts. In our two-channel multiwavelet system we have investigated incorporation of pre and post filtering to the multiwavelet transform and compression system for alleviating those ingerent visual artifacts due to multiwavelet effect. In addition, to quantify the image data compression performance of proposed multiwavelet system, computer simulations have been performed using various image data. For bit allocation and quantization, the Lagrange multiplier technique considering data rate vs. distortion rate along with a nonlinear companding method are applied equallly to all systems considered, here. The simulation results have yielded 1 ~ 2 dB compression enhancement over the scalar savelet systems. If the more advanced compression methods like SPIHT and run-length channel coding were adopted for the proposed multiwavelet system, a much higher compression gain could be obtained.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2007.06a
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• pp.758-761
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• 2007

Digital signal processing technique is applied in wide fields such as speech processing, image processing and spectrum analysis. Therefore, in order to do frequency selective operation digital filter is used in stead of analog filter and sharp filter characteristics can be implemented. Since finite impulse response (FIR) digital filter as nonrecursive type represents linear phase response characteristics and is always stable and is used in fields regarding wave information importantly such as data transmission. And due to frequency characteristics, in order to remove the Gibbs phenomenon generating around a discontinuous point, filter is designed through window function method. Therefore, in this paper to improve performance of FIR digital filter, a modified window function was applied. And the proposed method was compared with conventional methods using peak side-lobe and transition properties in simulations.