A Study on the Electronic Traffic Control Signal System

기계식 교통신호시스템에 비해 전자식 교통신호시스템은 여러가지 장점을 가지고 있다. 따라서, 인공지능 교통신호시스템은 전자식 교통신호시스템이 기본이 된다. 본 논문에서는 현재 국내에서 쓰이고 있는 교통신호시스템을 개선하여 입출력 등의 조작이 간단하고, 프로그램이 용이한 전자식 교통신호시스템을 설계하였다.Electronic traffic signal systems have more advantages than mechnical traffic signal systems, so that the intellignet transportation system (ITS) is based on the electronic traffic signal system. In this paper, the electronic traffic signal system, but not easy to deal with, is improved, that is, the input/output operation is simpler and the reprogramming is easier.Electronic traffic signal systems have more advantages than mechnical traffic signal systems, so that the intellignet transportation system (ITS) is based on the electronic traffic signal system. In this paper, the electronic traffic signal system, but not easy to deal with, is improved, that is, the input/output operation is simpler and the reprogramming is easier

Design of the Adaptive Noise Canceler using Neural Network with Backpropagation Algorithm

본 논문에서는 다층 신경회로망의 구조를 가지며, 백프로퍼게이션 학습 알고리즘을 이용한 적응신호처리 시스템을 구현하였다. 최소자승 알고리즘을 이용한 적응 잡음 제거기는 기준 신호와 잡음과의 상관도에 영향을 많이 받고, 신호가 잡음에 비하여 상대적으로 작은 경우에 한계를 보이고 있다. 이와 같은 잡음에 대하여 본 논문에서 제안된 시스템은 좋은 성능을 보인다. 또한, 은닉층의 수와 노드 수를 다르게 구성했을 경우에 시스템의 출력에 미치는 결과에 대하여 분석하였다. 제안된 적응 신호처리 시스템의 장점을 알아보기 위하여 성능 평가의 기준이 되는 최소자승 알고리즘을 이용한 시스템과 비교하였다.In this paper, the adaptive noise canceler using neural network with backpropagation is designed. The adaptive noise canceler using the least mean square algorithm has the large correlativity of the reference signal and shows the limitation when the signal is relatively small to the noise. The system proposed in this paper plays an important role in denoising these signals. In addition, the experiments are carried out to analyze the effects of the number of hidden layers and nodes about the system. The performance of the proposed adaptive noise canceler is compared with that of the system which is used the least mean square algorithm.In this paper, the adaptive noise canceler using neural network with backpropagation is designed. The adaptive noise canceler using the least mean square algorithm has the large correlativity of the reference signal and shows the limitation when the signal is relatively small to the noise. The system proposed in this paper plays an important role in denoising these signals. In addition, the experiments are carried out to analyze the effects of the number of hidden layers and nodes about the system. The performance of the proposed adaptive noise canceler is compared with that of the system which is used the least mean square algorithm

A Study on the Identification of Acoustic Signals Using PCRNN

파워스펙트럼은 디지탈신호처리 분야에서 널리 이용되어 왔으나 해석하고자 하는 신호의 위상에 관한 정보를 잃어 버리는 단점이 있어서, 위상에 관한 정보를 필요로 하는 경우에는 파워스펙트럼 해석법은 그 이용에 제약을 받는다. 바이스펙트럼은 그 계산에 시간이 많이 걸리는단점은 있으나 반면에 위상에 관한 정보를 잃지 않는 장점이 있다. 본 논문에서는 여러가지 자동차들의 엔진 음향을 녹음하여 바이스펙트럼을 구한 후, 부분결합 회귀신경회로망을 이용하여 바이스펙트럼 패턴을 식별하고 결과적으로 신호원을 식별하는 방법에 관하여 연구하였다.Power spectrum has been widely utilized in the digital signal processing area. However, there is no information about the phase of a signal, the power spectral analysis technique can not be used to interpret the phase coherency of the signal produced by some nonlinear process. In this case, the third-order higher-order spectrum, the so called bispectrum, is very useful in analyzing such signals. In this paper, the bispectra of the measured acoustic signals of vehicles are computed and then the PCRNN(partially connected recurrent neural netwok) is used in order to identify the acoustic signal sources using the bispectrum patterns.Power spectrum has been widely utilized in the digital signal processing area. However, there is no information about the phase of a signal, the power spectral analysis technique can not be used to interpret the phase coherency of the signal produced by some nonlinear process. In this case, the third-order higher-order spectrum, the so called bispectrum, is very useful in analyzing such signals. In this paper, the bispectra of the measured acoustic signals of vehicles are computed and then the PCRNN(partially connected recurrent neural netwok) is used in order to identify the acoustic signal sources using the bispectrum patterns

A study on the improvement of wavelet packet algorithm for image compression

본 논문에서는 코딩성능을 향상시키는 base 선택 알고리듬을 제안하였다. 제안한 알고리듬은 실험을 통하여 정해진 기준값에 의하여 알고리듬의 계산량이 제한되고, top-down tree search를 이용하며 주어진 영상에 따라서 자식 서브밴드에서 상대적으로 에너지 분포가 많은 suboptimal base를 선택한다. 본 연구의 목적은 웨이브릿 패킷 변환의 성능을 알아보는 것이므로 양자화기와 코딩 방법의 최적화는 고려되지 않았고, 제안한 알고리듬의 실험을 위하여 쿼드트리를 이용한 코팅을 적용하여 비트율과 PSNR을 조사하였으며, 비적응적 웨이브릿 변환과 비교하여 상대적으로 적은 계산량의 증가와 1.1㏈의 코딩 효율의 향상을 보였다.In this paper, the base selection algorithm, which improves the performance of the coding gains, is proposed. The resolved threshold value restrict the computational complexity and the proposed algorithm decompose the child subband with the top-down tree search and the relative energy between the parent and child subband. The object of this paper is the performance of wavelet packet transform and optimization of quantisers or coders is not considered. For the present experiments, we used the quadtree coder scheme and showed the bit-rate, peak signal-to-noise ratio (PSNR) distortion measure, and the reconstructed image. Compared to discrete wavelet transform, PSNR improvements up to 1.1㏈ are achieved for a set of standard test images.In this paper, the base selection algorithm, which improves the performance of the coding gains, is proposed. The resolved threshold value restrict the computational complexity and the proposed algorithm decompose the child subband with the top-down tree search and the relative energy between the parent and child subband. The object of this paper is the performance of wavelet packet transform and optimization of quantisers or coders is not considered. For the present experiments, we used the quadtree coder scheme and showed the bit-rate, peak signal-to-noise ratio (PSNR) distortion measure, and the reconstructed image. Compared to discrete wavelet transform, PSNR improvements up to 1.1㏈ are achieved for a set of standard test images

Traffic signal Control System using Fuzzy Theory

교통신호시스템을 퍼지 이론을 이용하여 최적화 시키는 알고리즘에 관하여 연구하고, 간단한 교차로에 적용시켜 보았다. 퍼지이론을 이용한 교통신호시스템을 이용할 경우 신호등에서의 대기시간이 감소함을 알 수 있었다.In this paper, a fuzzy algorithm, which is optimized the traffic control system, is studied, and this algorithm is applied to a simple crossroad. the delay time (i.e, waiting time) of the fuzzy controlled traffic signal system is less than that of the conventional system.In this paper, a fuzzy algorithm, which is optimized the traffic control system, is studied, and this algorithm is applied to a simple crossroad. the delay time (i.e, waiting time) of the fuzzy controlled traffic signal system is less than that of the conventional system

The Design of Digital Single Side-band Modulator

Weaver's analog단측파대 변조기의 원리와 Digital신호처리 기법을 적용하여 Digital단측파대 변조기의 특성을 computer simulation에 의하여 구하였다. 이러한 Digital단측파대 변조기는 주파수 분할 다중통신(FDM)시스템과 시분할 다중통신(TDM)시스템의 다중변환 시스템으로 사용할 수 있으며 Group band(60~108KHz)다중통신을 예로서 상호변환을 설명하고 시스템의 신호 대 잡음비(SNR)를 구하였다.The digital Weaver's single side band(SSB) modulator is simulated, using its corresponding theory and digital processing techniques. The simulated digital Weaver's modulator is used as a main part of the transmultiplexer which converts time division multiplexer(TDM) system to frequency division multiplexer(FDM) system. The digital filters are designed for this purpose and the SNR of the overall system is calculated.The digital Weaver's single side band(SSB) modulator is simulated, using its corresponding theory and digital processing techniques. The simulated digital Weaver's modulator is used as a main part of the transmultiplexer which converts time division multiplexer(TDM) system to frequency division multiplexer(FDM) system. The digital filters are designed for this purpose and the SNR of the overall system is calculated

Recognition of Acoustic Signals Using Bispectrum Technique and Neural Network

어떤 신호를 주파수영역에서 해석하고자 할 때 바이스펙트럼 기법은 파워스펙트럼 기법에 비하여 상대적으로 계산 시간이 많이 걸린다는 단점이 있으나 신호의 크기에 관한 정보 뿐 만 아니라 위상에 관한 정보도 찾아 낼 수 있다는 장점이 있다. 한편 바이스펙트럼은 1차원 함수인 파워스펙트럼과 달리 2차원 함수이므로 등고선 그림으로 표시된 바이스펙트럼은 하나의 무늬로 볼 수 있다. 측정된 음향신호의 바이스펙트럼을 무늬로 표시하면 각 음향신호에 대한 바이스펙트럼 무늬는 서로 다르다. 따라서 음향 신호원을 식별하기 위한 한가지 방법으로 각 음향신호의 바이스펙트럼 무늬를 식별하여 궁극적으로는 음향신호원을 식별하는 방법을 제안하였다. 본 논문에서는 여러 종류의 무한궤도차량(無限軌道車輛)의 음향신호를 측정하여 음향신호의 바이스펙트럼을 각각 구한 후, 신경회로망을 이용하여 각 바이스펙트럼을 식별하여 결과적으로 음향신호원을 식별하는 방법을 보였다.Bispectrum has the magnitude information of a signal in the frequency domain as well as the phase information of the signal. However, it takes longer time to compute the bispectrum of the signal than to compute the power spectrum which has only the magnitude information of the signal. Although the power spectrum is a 1-dimensional function, the bispectrum is a 2-dimensional function so that the contour plot of the bispectrum can be considered as a pattern. Since the patterns of the bispectra of the measured acoustic signals are different each other, a pattern recognition technique is proposed to recognizing the bispectra of the acoustic signals as a method to identify the sources of the acoustic signals. In this paper, the bispectra of the acoustic signals which are measured from several kinds of caterpillar vehicles are computed, then a neural network is used as the identifier of the bispectrum patterns of the caterpillar vehicles, as a result, the bispectrum pattrns are used to identify the sources of the acoustic signals.Bispectrum has the magnitude information of a signal in the frequency domain as well as the phase information of the signal. However, it takes longer time to compute the bispectrum of the signal than to compute the power spectrum which has only the magnitude information of the signal. Although the power spectrum is a 1-dimensional function, the bispectrum is a 2-dimensional function so that the contour plot of the bispectrum can be considered as a pattern. Since the patterns of the bispectra of the measured acoustic signals are different each other, a pattern recognition technique is proposed to recognizing the bispectra of the acoustic signals as a method to identify the sources of the acoustic signals. In this paper, the bispectra of the acoustic signals which are measured from several kinds of caterpillar vehicles are computed, then a neural network is used as the identifier of the bispectrum patterns of the caterpillar vehicles, as a result, the bispectrum pattrns are used to identify the sources of the acoustic signals

Design of Duplexer Filter Using Dielectric Resonator for Mobile Communication Handy Set

본 논문에서는 고유전율을 갖는 유전체 세라믹 재료를 이용하여 유전체 공진기 듀플렉서 필터를 설계 하였다. 기존의 듀플렉서 필터는 송신 3차, 수신 4차로 각 공진기를 횡으로 배열하는 구조를 갖는다. 본 논문에서는 송수신단 공진기를 수직으로 배열한 구조로 설계 하였다. 송신단과 수신단을 동일한 4차 필터로 구현하여 송신단에서 충분한 감쇠를 얻을 수 있었으며, 송·수신단에서의 통과대역은 각각 1.750-1.780GHz, 1.840-1.870GHz로 30MHz의 대역폭을 가지며 각 단에서의 신호의 분리도는 약 34dB 이상이며, 부피는 16㎜ ×6.77㎜ ×4㎜로 0.433㏄로 상용화된 필터의 약 90.2%로 설계 하였다. 각각의 유전체 공진기는 원형 동축형을 사용 하였으며, 제작공정의 단순화를 위해 각 공진기의 내·외경을 동일하게 하고 각 공진기의 공진 주파수는 공진기의 길이를 조절해 휴대전화 대역에 적합한 구조와 특성을 갖도록 설계 하였다.In this paper, a duplexer filter for mobile communication is designed using dielectric resonators with high dielectric constant ceramic materials. A commercial duplexer filter for mobile communication handy set is consist of horizontal array, and each sections of duplexer have 3 pole for Tx and 4 pole for Rx. In this paper, the designed filter have the structure of vertical array, and each sections of Tx and Rx have 4 pole. As a results, Tx sections have enough attenuations at Tx stopband frequencies. The passband of each sections; Tx, Rx, are 1.750-1.780GHz, 1.840-1.870GHz, so have 30MHz bandwidth, and the isolations between Tx and Rx bands are more than about 34dB. The physical size of designed duplexer filter is 16㎜ ×6.77㎜ ×4㎜, equals 0.433㏄, and it is about 90.2% of commercial filter size. Each dielectric resonators are designed by stepped impedance resonators with equal size of inside and outside diameters for simple fabrications. The resonant frequencies of each resonators are tuned by each resonator length, and the structure and characteristics are well suited for PCS(Personal Communication Systems) service.In this paper, a duplexer filter for mobile communication is designed using dielectric resonators with high dielectric constant ceramic materials. A commercial duplexer filter for mobile communication handy set is consist of horizontal array, and each sections of duplexer have 3 pole for Tx and 4 pole for Rx. In this paper, the designed filter have the structure of vertical array, and each sections of Tx and Rx have 4 pole. As a results, Tx sections have enough attenuations at Tx stopband frequencies. The passband of each sections; Tx, Rx, are 1.750-1.780GHz, 1.840-1.870GHz, so have 30MHz bandwidth, and the isolations between Tx and Rx bands are more than about 34dB. The physical size of designed duplexer filter is 16㎜ ×6.77㎜ ×4㎜, equals 0.433㏄, and it is about 90.2% of commercial filter size. Each dielectric resonators are designed by stepped impedance resonators with equal size of inside and outside diameters for simple fabrications. The resonant frequencies of each resonators are tuned by each resonator length, and the structure and characteristics are well suited for PCS(Personal Communication Systems) service

Acquisition behavior of a class of digital phase-locked loops

학위논문 (석사) - 한국과학기술원 : 전기 및 전자공학과, 1979.2, [ vi, 127 p. ]In this thesis the acquisition behaviors of a class of first-and second-order digital phase-locked loops (DPLL) originally proposed by Reddy and Gupta have been studied in the absence of noise. Specifically, the acquisition time, the lock range, and the effects of quantization on acquisition behaviors have been investigated. It has been found that the number of quantization levels L and the number of phase error states N in modulo 2$\pi$ play important roles in acquisition. For a given L-level quantizer, as N increases, the acquisition time increases, but the lock range decreases. Also, the deviation of the steady state phase error decreases in this case. When the number of quantization levels L increases, the acquisition time decreases, and the lock range increases. However, variation of L affects little for the steady state phase error. The effects of loop filter characteristics on acquisition have also been considered. One can get a smaller acquisition time and a larger lock range as the filter parameter value becomes larger. However, the deviation of the steady state phase error increases in that case. In addition, the possibility of using a nonuniform quantizer for fast acquisition has also been studied. Analytical results have been verified by computer simulation and experiments.한국과학기술원 : 전기 및 전자공학과