• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.2
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• pp.114-119
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• 2009

Sound source localization systems in service robot applications estimate the direction of a human voice. Time delay information obtained from a few separate microphones is widely used for the estimation of the sound direction. Correlation is computed in order to calculate the time delay between two signals. Inverse cosine is used when the position of the maximum correlation value is converted to an angle. Because of nonlinear characteristic of inverse cosine, the accuracy of the computed angle is varied depending on the position of the specific sound source. In this paper, we propose an efficient sound source localization system using angle division. By the proposed approach, the region from $0^{\circ}$ to $180^{\circ}$ is divided into three regions and we consider only one of the three regions. Thus considerable amount of computation time is saved. Also, the accuracy of the computed angle is improved since the selected region corresponds to the linear part of the inverse cosine function. By simulations, it is shown that the error of the proposed algorithm is only 31% of that of the conventional a roach.

• IE interfaces
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• v.9 no.3
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• pp.180-193
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• 1996

본 논문에서는 시뮬레이션 모델을 통해 호유형뿐만 아니라 개인통신서비스에서 필수적인 핸드오버 및 위치등록 등의 모든 트래픽 유형을 고래했을 때 개인통신교환기(PCX : Personal Commuincations Exchange)에서의 지연시간을 분석하며, 지연시간에 대한 결과와 프로세서의 이용율로부터 개인통신교환기의 호처리 및 이동성처리용량을 분석하고 그에 따른 가입자의 수용능력을 추정한다. IPC(Inter-Processor Communications) 메시지의 송수신시간 및 메시지의 처리시간등 교환기 제어계의 성능분석에 필요한 입력 파라미터 값은 기존 ISDN(Integrated Services Digital Network) 및 CMS-MX(CDMA Mobile System-Mobile Exchange) 교환기의 측정자료를 이용하였다. 시뮬레이션 분석결과 PCX 교환기의 호처리 성능은 주로 번호번역 기능을 담당하는 프로세서인 NTP(Number Translation Processor)의 용량에 의해 결정되며, 가입자 밀도가 1,500명$/km^2$인 경우 호처리용량은 약 42만 BHC(Busy Hour Call Completion)로 추정되었고, 이에 상응하는 핸드오버 및 위치등록 처리용량은 각각 시간당 약 2만 6천히 및 40만 6천회로 나타났다. 이것은 가입자당 호처리 부하가 1.6BHC일 경우, 약 15만 7천 가입자를 수용할 수 있는 용량에 해당한다.

• Proceedings of the Korean Society of Precision Engineering Conference
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• 2005.06a
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• pp.1120-1124
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• 2005

We reviewed controllers with a time delay estimation in this paper. Time delay control (TDC) and sliding mode control (SMC) are well known robust control schemes. Basically, the TDC has a main characteristic called a time delay estimation from which we can estimate the total uncertainty of a system. . The TDC causes the stick-slip in the case of systems with a friction. The so-called TDCSA which are short for TDC with switching action was developed to reduce the stick-slip. The TDC has the additional switching action term in the TDC structure. In the other hand, the SMC dose not have a time delay estimation but instead it can estimate the system uncertainty through the switching action. The SMC has a difficulty to estimate the total uncertainty of a system because it does not have a time delay estimation. In order to solve the difficulty, some control schemes were developed. Among them, we need to focus our attention on two control schemes: SMCPE and SMCTE, which are short for sliding mode control with a perturbation estimation and sliding mode control with a time delay estimation, respectively. In this paper, we analyzed and compared the characteristic of above three controllers. Even though the motives for the development of three control schemes are different, three control schemes have much in common in terms of their controller structures.

• The KIPS Transactions:PartB
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• v.9B no.4
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• pp.421-428
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• 2002

The local prediction method utilizing embedding vector loses the prediction power when the parameter r estimation is not exact for predicting the chaos time series induced from the high order differential equation. In spite of the fact that there have been a lot of suggestions regarding how to estimate the delay time ($\tau$), no specific method is proposed to apply to any time series. The inclinded linear model, which utilizes inclinded netter, yields satisfying degree of prediction power without estimating exact delay time ($\tau$). The usefulness of this approach has been indicated not only theoretically but also in practical situation when the method w8s applied to economical time series analysis.

• The Journal of the Acoustical Society of Korea
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• v.20 no.3
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• pp.34-41
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• 2001

In an IMBE (Improved Multi-band Excitation) speech coder, initial pitch estimation occupies most of the total computing time for the coder due to complex cost function and exhaustive search over candidate pitches. Future frames in initial pitch estimation cause inevitable time delay. Therefore, it is difficult to implement a real-time coder. Furthermore, unvoiced frames use the unnecessary pitch estimation as in the voiced frames. In this paper, each frame is determined voiced or unvoiced by Dyadic Wavelet Transform (DyWT) and, then, initial pitch estimation is performed only for voiced frame. Therefore different pitch estimation algorithms are employed between voiced and unvoiced frames incurring reduced time delay at transmitter and receiver. Simulation result show that the relative complexity of initial pitch estimation is reduced by 23％, and the processing time decreases down to 1/10 ∼ 1/1l of the IMBE coder while speech quality is almost maintained.

• The Journal of the Korea institute of electronic communication sciences
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• v.12 no.3
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• pp.493-498
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• 2017

Recently, traffic data is analyzed for efficiency of bus operation, D2D(: Door to Door) service, and self-driving of public transportation. However, various studies have been carried out to predict the delay time of public transportation, especially buses, but the research to date has been insufficient due to limitations of simple analysis and data acquisition. In this study, delay time estimation is performed by collecting and processing data such as day of the week, weather, and time of day based on bus operation information. The proposed method in this paper can be applied to autonomous public transport and public traffic control system by improving the accuracy by adding variables in the future.

• The Journal of the Acoustical Society of Korea
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• v.43 no.2
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• pp.178-183
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• 2024

This study introduces an enhanced sound source localization technique, bolstered by a data-driven deep learning approach, to improve the precision and accuracy of direction of arrival estimation. Focused on refining Time Difference Of Arrival (TDOA) based sound source localization, the research hinges on accurately estimating TDOA from cross-correlation functions. Accurately estimating the TDOA still remains a limitation in this research field because the measured value from actual microphones are mixed with a lot of noise. Additionally, the digitization process of acoustic signals introduces quantization errors, associated with the sampling frequency of the measurement system, that limit the precision of TDOA estimation. A deep learning-based approach is designed to overcome these limitations in TDOA accuracy and precision. To validate the method, we conduct comprehensive evaluations using both two and three-microphone array configurations. Moreover, the feasibility and real-world applicability of the suggested method are further substantiated through experiments conducted in an anechoic chamber.

• The KIPS Transactions:PartC
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• v.11C no.2
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• pp.263-268
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• 2004

The end-to-end characteristic Is an important factor for QoS support. Since network users and required bandwidths for applications increase, the efficient usage of networks has been intensively investigated for the better utilization of network resources. The distributed adaptive routing is the typical routing algorithm that is used in the current Internet. The DCLC(Delay Constrained 1.east Cost) path problem has been shown to be NP-hard problem. The path cost of LD path is relatively more expensive than that of LC path, and the path delay of LC path is relatively higher than that of LD path in DCLC problem. In this paper, we investigate the performance of heuristic algorithm for the DCLC problem with new factor which is probabilistic combination of cost and delay. Recently Dr. Salama proposed a polynomial time algorithm called DCUR. The algorithm always computes a path, where the cost of the path is always within 10% from the optimal CBF. Our evaluation showed that heuristic we propose is more than 38% better than DCUR with cost when number of nodes is more than 200. The new factor takes in account both cost and delay at the same time.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.12
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• pp.2376-2385
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• 1994

In receiving the digital TV signal, both decoding delay and the channel hopping delay are very critical factors in applications. The intra-slice coding in the MPEG-2 SIMPLE PROFLE of No B-picture is one of the primary methods for short delay time in video decoding. It has the advantage of short decoding delay, but has the drawback of long channel hopping delay time. In this paper, we propose a method to reduce the channel delay with negligible loss in SNR performance. It is shown that dividing pictures into several regions of slices and adding some restriction in motion vector search for inter-frame coding. hence the random acess points are effectively increased. and the channel hopping delay is reduced.

• The Journal of the Acoustical Society of Korea
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• v.14 no.3
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• pp.28-36
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• 1995

In this paper, a double loop control model using leaky delay LMS algorithm are proposed for active noise control. The proposed double loop control model estimates the loudspeaker characteristic and the error path transfer function with on-line using only gain and acoustic time delay to reduce computation burden. The control of error signal through double loop control scheme makes the more robust cntrol system. The input signal of filter to estimate acoustic time delay is used difference between input signal of input microphone and adaptive filter output. And also, in nonstationary environments, the leaky delay LMS algorithm is employed to counteract parameter drift of delay LMS algorithm. For practical noise signal, the proposed double loop control model reduces noise level about 12.9 dB.