• Journal of Communications and Networks
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• 제11권2호
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• pp.166-174
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• 2009

In this paper, a distributed power and end-to-end rate control algorithm is proposed in the presence of licensed users. By Lagrangian duality theory, the optimal power and rate control solution is given for the unlicensed users while satisfying the interference temperature limits to licensed users. It is obtained that transmitting with either 0 or the maximum node power is the optimal scheme. The synchronous and asynchronous distributed algorithms are proposed to be implemented at the nodes and links. The convergence of the proposed algorithms are proved. Finally, further discussion on the utility-based fairness is provided for the proposed algorithms. Numerical results show that the proposed algorithm can limit the interference to licensed user under a predefined threshold.

• ETRI Journal
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• 제35권4호
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• pp.697-705
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• 2013

Emerging trends in the area of digital very large scale integration (VLSI) signal processing can lead to a reduction in the cost of the cochlear implant. Digital signal processing algorithms are repetitively used in speech processors for filtering and encoding operations. The critical paths in these algorithms limit the performance of the speech processors. These algorithms must be transformed to accommodate processors designed to be high speed and have less area and low power. This can be realized by basing the design of the auditory filter banks for the processors on digital VLSI signal processing concepts. By applying a folding algorithm to the second-order digital gammatone filter (GTF), the number of multipliers is reduced from five to one and the number of adders is reduced from three to one, without changing the characteristics of the filter. Folded second-order filter sections are cascaded with three similar structures to realize the eighth-order digital GTF whose response is a close match to the human cochlea response. The silicon area is reduced from twenty to four multipliers and from twelve to four adders by using the folding architecture.

• Journal of Information Processing Systems
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• 제16권5호
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• pp.1001-1007
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• 2020

In recent years, machine learning algorithms are continuously being used and expanded in various fields, such as facial recognition, signal processing, personal authentication, and stock prediction. In particular, various algorithms, such as deep learning, reinforcement learning, and Q-learning, are continuously being improved. Among these algorithms, the expansion of deep learning is rapidly changing. Nevertheless, machine learning algorithms have not yet been applied in several fields, such as personal authentication technology. This technology is an essential tool in the digital information era, walking recognition technology as promising biometrics, and technology for solving state-space problems. Therefore, algorithm technologies of deep learning, reinforcement learning, and Q-learning, which are typical machine learning algorithms in various fields, such as agricultural technology, personal authentication, wireless network, game, biometric recognition, and image recognition, are being improved and expanded in this paper.

• 한국항해항만학회:학술대회논문집
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• 한국항해항만학회 2006년도 International Symposium on GPS/GNSS Vol.2
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• pp.9-14
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• 2006

Galileo is the European Global Navigation Satellite System, under civilian control, and consists on a constellation of medium Earth orbit satellites and its associated ground infrastructure. Galileo will provide to their users highly accurate global positioning services and their associated integrity information. The elements in charge of the computation of Galileo navigation and integrity information are the OSPF (Orbit Synchronization Processing Facility) and IPF (Integrity Processing Facility), within the Galileo Ground Mission Segment (GMS). Navigation algorithms play a key role in the provision of the Galileo Mission, since they are responsible for computing the essential information the users need to calculate their position: the satellite ephemeris and clock offsets. Such information is generated in the Galileo Ground Mission Segment and broadcast by the satellites within the navigation signal, together with the expected a-priori accuracy (SISA: Signal-In-Space Accuracy), which is the parameter that in fault-free conditions makes the overbounding the predicted ephemeris and clock model errors for the Worst User Location. In parallel, the integrity algorithms of the GMS are responsible of providing a real-time monitoring of the satellite status with timely alarm messages in case of failures. The accuracy of the integrity monitoring system is characterized by the SISMA (Signal In Space Monitoring Accuracy), which is also broadcast to the users through the integrity message.

• 융합신호처리학회논문지
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• 제7권4호
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• pp.178-183
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• 2006

As a lot of researches on the speech signal processing are performed due to the recent rapid development of the information-communication technology. the pitch period is used as an important element to various speech signal application fields such as the speech recognition. speaker identification. speech analysis. or speech synthesis. A variety of algorithms for the time and the frequency domains related with such pitch period detection have been suggested. One of the pitch detection algorithms for the time domain. AMDF (average magnitude difference function) uses distance between two valley points as the calculated pitch period. However, it has a problem that the algorithm becomes complex in selecting the valley points for the pitch period detection. Therefore, in this paper we proposed the modified AMDF(M-AMDF) algorithm which recognizes the entire minimum valley points as the pitch period of the speech signal by using the rotation transform of AMDF. In addition, a threshold is set to the beginning portion of speech so that it can be used as the selection criteria for the pitch period. Moreover the proposed algorithm is compared with the conventional ones by means of the simulation, and presents better properties than others.

• 융합신호처리학회논문지
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• 제7권4호
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• pp.201-206
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• 2006

유전자 알고리즘은 공학 분야에서 필요한 여러 가지 최적화 문제에 대하여 최적에 가까운 해를 제공해주는 반복적 알고리즘으로 알려져 있다. 본 논문에서는 특정 교배방법에서 유전자의 배열순서가 적합도가 높은 스키마의 길이에 미치는 영향을 고찰하였다. 또한 이에 따른 유전자 알고리즘의 성능 변화를 두 개의 예제를 이용한 실험을 통하여 관찰하였다. 예제로 사용된 그래프 분할과 knapsack 문제를 위해 몇 가지 유전자 재배열 방법을 제시하였다. 실험결과에 따르면 유전자 재배열 방법마다 서로 다른 유전자 알고리즘 성능을 보여주었으며, 적합도가 높은 스키마의 길이를 고려한 재배열 방법이 재배열을 하지 않았을 때 보다 유전자 알고리즘의 성능을 향상시켜 주는 것을 관찰하였다. 따라서 주어진 문제에 적합한 유전자 재배열 방법을 찾는 것이 대단히 중요함을 확인하였다.

• Structural Engineering and Mechanics
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• 제71권1호
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• pp.57-63
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• 2019

During the last two decades, much joint research regarding vibration based methods has been done, leading to developing various algorithms and techniques. These algorithms and techniques can be divided into modal methods and signal methods. Although modal methods have been widely used for health monitoring and damage detection, signal methods due to higher efficiency have received considerable attention in various fields, including aerospace, mechanical and civil engineering. Signal-based methods are derived directly from the recorded responses through signal processing algorithms to detect damage. According to different signal processing techniques, signal-based methods can be divided into three categories including time domain methods, frequency domain methods, and time-frequency domain methods. The frequency domain methods are well-known and interest in using them has increased in recent years. To determine dynamic behaviours, to identify systems and to detect damages of bridges, different methods and algorithms have been proposed by researchers. In this study, a new algorithm to detect seismic damage in the bridge's piers is suggested. To evaluate the algorithm, an analytical model of a bridge with simple spans is used. Based on the algorithm, before and after damage, the bridge is excited by a sine force, and the piers' responses are measured. The dynamic specifications of the bridge are extracted by Power Spectral Density function. In addition, the Least Square Method is used to detect damage in the bridge's piers. The results indicate that the proposed algorithm can identify the seismic damage effectively. The algorithm is output-only method and measuring the excitation force is not needed. Moreover, the proposed approach does not need numerical models.

• 융합신호처리학회논문지
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• 제12권2호
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• pp.128-132
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• 2011

LMS(Least mean square) 적응 알고리즘은 radar, sonar, 음성처리, 이동통신 분야 등에서 중요한 역할을 하고 있다. 이동통신 분야에서는 LMS 적응 알고리즘의 빠른 수렴속도가 더욱 중요하다. 하지만 LMS 알고리즘은 수렴속도가 느리고 일정치 않은 수렴을 하는 문제점을 가지고 있다. 이러한 문제점을 해결하기 위하여 다양한 가변 스텝 LMS 적응 알고리즘들이 최근에 많이 연구되어왔다. 연구된 많은 LMS 알고리즘들은 빠른 수렴속도를 얻기 위하여 복잡한 가변스텝방식을 사용하는데 이는 많은 계산량을 필요로 한다. 따라서 LMS 알고리즘의 최대 장점인 단순성과 강인성을 약화시킨다. 제안하는 결정함수 가변스텝 LMS 알고리즘은 스텝 값을 간단한 결정함수에 따라 결정하므로 단순성을 최대한 강화하면서 빠른 수렴속도를 얻도록 한다.

• The Journal of the Acoustical Society of Korea
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• 제15권3E호
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• pp.78-85
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• 1996

The set of admissible time-variations in the input signal can be separated into two categories : slow parameter changes and large parameter changes which occur infrequently. A common approach used in the tracking of slowly time-varying parameters is the exponential recursive least-squares(RLS) algorithm. There have been a variety of research works on the error analysis of the exponential RLS algorithm for the slowly time-varying parameters. In this paper, the focus has been given to the error analysis of exponential RLS algorithms for the input data with abrupt property changes. The voiced speech signal is chosen as the principal application. In order to analyze the error performance of the exponential RLS algorithm, deterministic properties of the exponential RLS algorithms is first analyzed for the case of abrupt parameter changes, the impulsive input(or error variance) synchronous to the abrupt change of parameter vectors actually enhances the convergence of the exponential RLS algorithm. The analysis has also been verified through simulations on the synthetic speech signal.

• 융합신호처리학회논문지
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• 제7권2호
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• pp.65-72
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• 2006

디지털신호처리에 있어서 신호의 다상분해(polyphase decomposition)는 멀티레이트(multirate) 시스템 구현에 있어서 다양한 용도로 사용된다. 특히 디지털 필터설계 등에 있어서 다채널 신호를 제공함으로써 병열연산을 가능하게 하여 알고리즘의 성능을 개선하는데 특히 유용하다. 일반적으로 다상분해신호는 원신호를 다운샘플링을 위해 데시메이션(decimation)하기 때문에 원신호에 비해 많은 고주파 성분을 포함하여 주파수 대역이 확장되는 경향이 있다. 이러한 성질은 다상분해신호의 활용, 즉 디지털 신호처리 시스템의 구조나 성능에 상당한 제약을 초래한다. 따라서 본 연구에서는 다상분해신호에 대한 완전 대역압축 및 복원 방법을 이론적으로 제시하고 Matlab 시뮬레이션을 통하여 그 유효성을 확인한다.