• Journal of information and communication convergence engineering
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• v.19 no.1
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• pp.54-60
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• 2021

Various digital devices were supplied throughout the Fourth Industrial Revolution. Accordingly, the importance of data processing has increased. Data processing significantly affects equipment reliability. Thus, the importance of data processing has increased, and various studies have been conducted on this topic. This study proposes a modified Gaussian filter algorithm based on a fuzzy membership function. The proposed algorithm calculates the Gaussian filter weight considering the standard deviation of the filtering mask and computes an estimate according to the fuzzy membership function. The final output is calculated by adding or subtracting the Gaussian filter output and estimate. To evaluate the proposed algorithm, simulations were conducted using existing additive white Gaussian noise removal algorithms. The proposed algorithm was then analyzed by comparing the peak signal-to-noise ratio and differential image. The simulation results show that the proposed algorithm has superior noise reduction performance and improved performance compared to the existing method.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.32B no.8
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• pp.1085-1090
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• 1995

The Discrete Cosine Transform(DCT) is effective technique for image compression, which is widely used in the area of digital signal processing. In this paper, an efficient DCT processor is proposed and simulated by using Verilog HDL. This algorithm is improved 60% in processing speed, but it's somewhat complicate compared with Y. Arai's algorithm. This algorithm will be used efficiently for real time image processing.

• Journal of Biomedical Engineering Research
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• v.11 no.1
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• pp.75-82
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• 1990

The purpose of this study is to help the hearing Impaired's speech training with a visible speech processing system. In brief, this system converts the features of speech signals into graphics on monitor, and adjusts the features of hearing impaired to normal ones. There are formant and pitch in the features used for this system. They are extracted using the digital signal processing such as linear predictive method or AMDF(Average Magnitude Difference Function). In order to effectively train for the hearing impaired's abnormal speech, easilly visible feature has been being studied.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1984.04a
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• pp.13-17
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• 1984

A hybrid version of the Hadamard and center Line Symmetric Haar Transform called H-CLSH is defined and developed. Efficient algorithms for fast computation of the H-CLSH and its inverse are developed. The H-CLSH is applied to digital signal and image processing and its utility and image processing and its utility and effectiveness are compared with Hadamard-Haar discrete transforms on the basis of some standard performance criteria.

• The Journal of the Acoustical Society of Korea
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• v.8 no.5
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• pp.95-101
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• 1989

For developing new algorithms or dedicated hardware by using general purpose Digital Signal Processor chip, emulator H/W and simulator S/W are indispensible. But the most of DSP emulators have limitations on H/W flexibility according to their generalized architectures. In this paper, a DSP evaluation system for real time signal processing was developed using TMS 32020. The I/O buffers storing acquisition data of the system were designed to have variable length $(1\sim2048samp1es) &$ sampling frequency $l00\sim8KHz$.

• Proceedings of the Korean Society of Precision Engineering Conference
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• 1993.04b
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• pp.198-203
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• 1993

In this study, byusing EMAT(Electro Magnetic Acoustic Transducer) the artificial slit is installed on 12B-SUS pipe test piece. By mading 4 cycle SH-bust wave (EMA) incidence to 45 .deg. angle, the signaldata of pulse, which is recevied from EMAT translated intodigital-signal-processing-method SSP and Deconvolution method by using FACOM. Results of these indicated that (1) this method of this study shows exellent result more than Ultrasonic testing method; (2) noise is well removed by SSP using signal dataa and resolving power and S/N ratio are advanced; (3) regradless of Ultrasonic wave, whichhas properties of generalstainless steel is generated into multiscattering and reflection phenomena, the resolving power of more than two times is progressed by being translated into Decon-volution method; and (4) as addition-averaging-processing number is increaing, the resolving power and S/N ratio are improved and the satisfactory signal is obtained.

• Journal of the Korean Institute of Telematics and Electronics
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• v.16 no.6
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• pp.1-10
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• 1979

Recent progresses in the signal processing technique in digital domain as well as that of analogue, impose a heavy burden on scientists and engineers intending to study this dis cipline, we surveyed basic tools for these vast branches to help those who have concerns on this field without being buried in detailed techniques. The first article is naturally confined to the basic tools namely random process analysis and characterization of random signal by covariance function.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.655-656
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• 2006

A parallel analog Viterbi decoder which decodes PR (1,2,2,1) signal of optical disc has been fabricated into chip. The proposed parallel analog Viterbi decoder implements the functions of the conventional digital Viterbi decoder utilizing the analog parallel processing circuits. In this paper, the analog parallel Viterbi decoding technology is applied for the PR signal. The benefit of analog processing is the low power consumption and the less silicon consumption. The test results of the fabricated chip are reported in this paper.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38C no.6
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• pp.497-502
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• 2013

In this paper, we aim to evaluate the performance of the digital signal processing (DSP) algorithm used in [8] in order to improve the resolution of spectrometers with fixed number of low-cost, non-ideal filters. In such spectrometers, the resolution is limited by the number of filters. We aim to demonstrate via new experiments that the resolution improvement by six times over the conventional limit is possible by using the DSP algorithm as claimed by [8].

• 제어로봇시스템학회:학술대회논문집
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• 1998.10a
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• pp.75-80
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• 1998

This paper proposes a novel computer human interface, named Virtual Wireless Microphone (VWM), which utilizes computer vision and signal processing. It integrates real-time face tracking and sound signal processing. VWM is intended to be used as a speech signal input method for human computer interaction, especially for autonomous intelligent agent that interacts with humans like as digital secretary. Utilizing VWM, the agent can clearly listen human master's voice remotely as if a wireless microphone was put just in front of the master.