• Journal of the Korean Society for Nondestructive Testing
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• v.33 no.3
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• pp.264-269
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• 2013

Since MFC(macro-fiber composite) transducer has been developed, many researchers have tried to apply this transducer on SHM(structural health monitoring), because it is so flexible and durable that it can be easily embedded to various kinds of structures. The objective of this paper is to figure out the benefits and feasibility of applying MFC transducers to guided wave technique. For this, we have experimentally tested the performance of MFC patches as transmitter and sensors for excitation and reception of guided waves on the thin aluminum alloy plate. In order to enhance the signal accuracy, we applied the FIR filter for noise reduction as well as used STFT(short-time Fourier transform) algorithm to image the guided wave characteristics clearly. From the results, the guided wave generated based on MFC showed good agreement with its theoretical dispersion curves. Moreover, the ultrasonic Lamb wave techniques based on MFC patches in pitch-catch manner was tested for detection of surface notch defects of which depths are 10%, 20%, 30% and 40% of the aluminum plate thickness. Results showed that the notch was detectable well when the notch depth was 10% of the thickness or greater.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2008.04a
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• pp.727-734
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• 2008

This research presents a laboratory study about difference of subjective response between with and without pictures. A main source is impulsive sound caused by leisure shooting. The sources are sampled from outdoor noise and their levels range from 40 to 75 dB at the interval of 5dB. The noise unit is based on A-weighted sound exposure level (ASEL; $L_{AE}$). To make equal ASEL of outdoor noise, finite impulse response (FIR) filter is applied to the originally sampled source to include the effect of distance attenuation. The evaluation method of the jury test adopted a Semantic Difference(SD) Method. The intersection point which two lines crossed was used as reference point. The intersecting point of mean response rating between with and without pictures was approximately 44ASEL and that of %HA was about 60ASEL. In the result of the test, the negative effect of pictures was given at a lower levels than intersection point while the positive effect was given at a higher levels than that.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.18 no.7
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• pp.746-755
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• 2008

This research presents a laboratory study on subjective response of impulsive sound caused by construction site (breaker and blasting). The sources are sampled from outdoor noise and their levels range from 40 to 75 dBA at the interval of 5 dBA. The noise unit is based on A-weighted sound exposure level(ASEL; $L_{AE}$). To make equal ASEL of listening level, finite impulse response(FIR) filter is applied to the originally sampled source to include the effect of propagation attenuation. Sixty-three subjects, forty-two males and twenty-one female, between 18 and 29 years of age, participated in the experiment. The evaluation method of jury test adopted a semantic difference method(SDM). In the test results for impulsive noise, the subjective response of blasting noise was higher than that of breaker noise. The result of %HA that has been combined responses of the three methods except for pink-noise was executed by regression analysis and was shown as the following equation.: $%HA=746.53/(1+{\exp} (L_{AE}-93.3))+0.34$.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2003.11a
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• pp.643-648
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• 2003

The problem considered in this paper is that the tip position of a flexible cantilever beam is controlled to follow a command signal, using a pair of piezoelectric actuators at the clamped end. The beam is lightly damped and so the natural transient response is rather long, and also since the sensor and actuator are not collocated, the plant response is non-minimum phase. Two control strategies were investigated. The first involved conventional PID control in which the feedback gains were adjusted to give the fastest closed-loop response to a step input. The second control strategy was based on an internal model control (IMC) architecture. The control filter in the IMC controller was a digital FIR device designed to minimize the expectation of the mean square tracking error. The IMC controller designed fur the beam was found to have very much reduced settling times to a step input compared with those of the PID controller.

• Journal of the Institute of Convergence Signal Processing
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• v.9 no.3
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• pp.224-229
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• 2008

In the paper, a new Constraint algorithm is proposed to solve the fan-in problem occurred in the encoding circuitry of an ADC. The Flash ADC architecture uses a Double-Base Number System(DBNS). The DBNS has been known to represent the Multidimensional Logarithmic Number System (MDLNS) used for implementing the multiplier accumulator architecture of FIR filter in Digital Signal Processing (DSP) applications. The authors use the DBNS with the base 2 and 3 in designing ADC encoder circuits, which is called as Double Base Integer Encoder(DBIE). A symmetric map is analyzed first, and then asymmetric map is followed to provide addition ready DBNS for DSP circuitry. The simulation results of the DBIE circuits in 6-bit and 8-bit ADC show the effectiveness of the Constraint algorithm with $0.18{\mu}m$ CMOS technology. The DBIE yields faster processing speed compared to the speed of Fat Tree Encoder (FAT) circuits by 17% at more power consumption by 39%.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.7C
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• pp.704-711
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• 2002

This paper proposes a DFE (Decision Feedback Equalizer) equalizer ASIC using the Multi-Modulus Algorithm (MMA) for cable modem applications. We believe that it is the first effort to combine the DFE structure and the MMA algorithm. The proposed equalizer has been designed for 64/256 QAM modems. The existing MMA equalizer uses two transversal filters and updates two tap weights while the proposed equalizer uses two DFE filter banks to improve the channel adaptive performance and to reduce the number of taps and updates only one tap weights. We have used the 0.35 $\mu\textrm{m}$ standard cell library. The implemented equalizer ASIC operates at 8 MHz and provides 64 Mbps which is higher than existing equalizers. The total number of gates are about 160,000.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.25 no.8
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• pp.30-38
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• 2011

This paper proposes a new intelligent diagnosis equipment for the partial discharge, which keeps deteriorating the insulating materials inside electric machineries, ultimately leading to electrical breakdown. In order to simulate experimentally the partial discharge inside the electric machinery, the tip-to-plate, the sphere-to-plate, the sphere-to-sphere and the plate-to-plate electrodes are used respectively, of which the gaps are 1[mm], 3[mm] or 5[mm] and the applied voltages are 3[kV], 5[kV] or 7[kV]. Ceramic coupler sensor and FIR digital filter are used to measure the partial discharge and the artificial neural network is used for the deterioration diagnosis of the electric machinery. The microprocessor of PD diagnosis equipment is DSP (TMS320C6713) with FPGA (Cyclone II). The results of the real-time and on-line experiments performed with the developed equipment are also explained.

• Proceedings of the KOSOMBE Conference
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• v.1996 no.05
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• pp.320-324
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• 1996

Digital fetal monitoring system based on the personal computer combined with the digital signal processing board was implemented. The DSP board acquires and digitally processes ultrasound fetal Doppler signal for digital rectification, FIR filtering, autocorrelation function calculation, its peak detection and MEDIAN filtering. The personal computer interfaced with the DSP board is in charge of graphic display, hardcopy, data transmission and on-line analysis of fetal heart rate change including and variability. I used a recursive technique for autocorrelation function computation method and MEDIAN filter which can greatly reduce the amount of calculation and accuracy. I also implemented analysis algorithm of fetal heart rate change based on normal fetal sample data in order to exact diagnosis.

• The Journal of the Acoustical Society of Korea
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• v.15 no.6
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• pp.81-89
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• 1996

It is very important for ultrasonic test method to evaluate the integrity of the class I components in nuclear power plants. However, as the rltrasonic test is affected by internal structures and configurations of test materials, backscattering, that is, time invariant noise is generated in large grain size materials. Due to the above reason, the received signal results in low signal to noise(S/N) ratio. Split spectrum processing(SSP) technique is effective to suppress the grain noise. The conventional SSP technique. however, has been applied to unique algorithm. This paper shows that MPO(minimization and polarity threshold) algorithm which two algorithms are applied simulatancously, was utilized, the signal processing time was shorten by using the new constant-Q SSP with the FIR filter which frequency to bandwidth ratio is constant and the optimum parameters were analysed for the signal processing to longitudinal wave and shear wave with the same requirements of inspection on nuclear power plant site. Moreover, the new ultrasonic test instrument, the reference block of the same product form and material specification, stainless stell test specimens and copper test specimens block of the same fabricated for the application of new SSP technique. As the result of experimental test with new ultrasonic test instrument and test specimens, the signal to noise ratio was improved by appying the new SSP technique.

• The Journal of the Acoustical Society of Korea
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• v.14 no.2
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• pp.50-61
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• 1995

A time scale modification method yielding rate-modified speech while conserving the characteristic of speech was implemented in real-time using a goneral purpose digital signal processor. Time scale modification changed pronunciation speed only, producing a time difference between the input signal and the modified signal, making it impossible to implement it in real-time. In this thesis, a system was implemented to remove the time difference between the input and modified signals. Speech signals slowed down or speeded up by a physical time scale modification method, such as adjusting the motor speed of the cassett tape recorder, was used as the input signal. Physical modification that controled only the inter speed of the cassette tape player distorted the pitch period of the original speech. In this study, a real-time system was implemented so that the pitch-distorted speech was reconstructed back to the original by fractional sampling pitch shifting using an FIR filter, and this signal was time scale modified to match the cassette tape recorder motor speed using SOLA time-scale medification. In experiments using speech signals medifiedby the proposed method, results obtained using a 16-bit resolution ADSP2101 processor and using computer simulations employing floating point operations showed about the same average frame signal-to-noise ratio of about 20 dB.