• Proceedings of the Korean Institute of Navigation and Port Research Conference
• /
• 2004.04a
• /
• pp.55-60
• /
• 2004

The paper describes measurement techniques for an acoustic transfer characteristic of ship's bridge stimulated by a whistle sound. The response sounds, according to the opening-shutting conditions of bridge doors for T／S ‘SAENURI’, are measured by B＆K 2260D equipment, and then extracted the frequency responses by B＆K 7830 software. To evaluate the measured transfer characteristic, the 128th order FIR (Finite Impulse Response) filters, containing the frequency characteristic, are constructed based on the frequency sampling-based design method. Using evaluation indexes with six scales, psychological assessments by five subjects are carried out with test sounds which are convolved source sound and FIR filters. As results of tests, the test sounds gives A／sub s／=3.3∼4.7 which means the psychological sense of 'it is almost similar sound as original ones in a real world', and thus it is cleary seen tint the proposed method can be suit for the measurement of an acoustic transfer characteristic of ship's bridg.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.25 no.1A
• /
• pp.167-177
• /
• 2000

In this paper, a new algorithm to design a time-domain equalizer (TEQ) for an xDSL system employing the discrete multitone (DMT) modulation is proposed. The proposed algorithm, derived by neglecting the terms whichdo not affect the performance of a DMT system in ARMA modeling, is shown to have similar performance tothe previous TEQ algorithms such as matrix inverse algorithm, fast algorithm, iterative algorithm, and inversepower method, even with the significantly lower computational complexity. In addition, since the proposedalgorithm requires only the received signal, the information on the channel impulse response or training sequenceis not needed. It is also shown that for the case where bridged tap is not included, the number of TEQ tapsrequired can be reduced to half(from 16 to 8) without affecting the overall performance. The performances of theproposed and previous TEQ algorithms are compared by applying them to ADSL environment.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.34 no.2C
• /
• pp.145-152
• /
• 2009

Ultra-wideband communication systems for impulse radio have merits that are possible for either high resolution ranging system or radio determination. Conventionally, in order to accomplish these functions, the rapid analog to digital convertor (ADC) is necessary to apply radio determination system operating in time domain. However, considering that low rate - wireless personal area network (LR-WPAN) aims to low-cost hardware implementation, the expensive ADC converting GHz sampling per second is not appropriate. So, this paper introduces the high resolution ranging system operating in frequency domain with using low sampling rate ADC, and a new non-coherent ranging scheme utilizing analog Frequency Modulation (FM) mode for the frequency domain transformation. To verify the superiority of the proposed ranging algorithm working in frequency domain, the suggested IEEE 802.15.4a TG channel model is used to exploit affirmative features of the proposed algorithm with conducting the simulation results.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.39B no.10
• /
• pp.715-721
• /
• 2014

In decades, many researchers have studied the time delay estimation (TDE) method for the signals in the two different receivers. The channel estimation based TDE is one of the typical TDE methods. The channel estimation based TDE models the time delay between two receiving signals as an impulse response in a channel between two receivers. In general the impulse response becomes sparse. However, most conventional TDE algorithms cannot have utilized the sparsity. In this paper, we propose a TDE method taking the sparsity into consideration. The performance comparison shows that the proposed algorithm improves the estimation accuracy by 10 dB in the white gaussian source. In addition, even in the colored source, the proposed algorithm doesn't show the estimation threshold effect.

• The Journal of the Acoustical Society of Korea
• /
• v.32 no.2
• /
• pp.124-130
• /
• 2013

The performance of underwater acoustic communication system is sensitive to the ISI(Inter-Symbol Interference) due to delay spread develop of multipath signal propagation. The equalizer is used to combat the ISI. In this paper, the performances of underwater acoustic communication with turbo equalizer were evaluated by real data collected in Korean littoral sea. As a result, when one iterative decoding using turbo equalizer is applied, the performance was improved 1.5 dB than the case of the non-iterative equalizer at BER $10^{-4}$. In the case of two or three iterations the performance was enhanced about 3.5 dB, but the performance wasn't improved any more in the case of more than three times.

• Journal of the Institute of Convergence Signal Processing
• /
• v.10 no.3
• /
• pp.198-206
• /
• 2009

This paper presents the reduction of the positional uncertainty of an ultrasonic sensor ring with overlapped beam pattern for the efficient obstacle detection of a mobile robot. Basically, it is assumed that a relatively small number of inexpensive low directivity ultrasonic sensors are installed at regular spacings along the side of a circular mobile robot with their beams overlapped. First, for both single and double obstacles, we show that the positional uncertainty inherent to an ultrasonic sensor can be reduced using the overlapped beam pattern, and also quantify the relative improvement in positional uncertainty. Second, given measured distance data from one or two ultrasonic sensors, we devise the geometric method to determine the position of an obstacle with respect to the center of a mobile robot. Third, we examine and compare existing ultrasonic sensor models, including Gaussian distribution, parabolic distribution, uniform distribution, and impulse, and then build the sensor model of overlapped ultrasonic sensors, adequate for obstacle detection in terms of positional uncertainty and computational requirement. Finally, through experiments using our prototype ultrasonic sensor ring, the validity of overlapped beam pattern for reduced positional uncertainty and efficient obstacle detection is demonstrated.

• Journal of Institute of Control, Robotics and Systems
• /
• v.18 no.10
• /
• pp.970-976
• /
• 2012

In this paper, a single LRF has been used to produce a 3D map for the mobile robot navigation. The 2D laser scanners are used for mobile robots navigation, where the laser scanner is applied to detect a certain level of area by the straight beam. Therefore it is limited to the usages of 2D obstacle detection and avoidance. In this research, it is designed to complement a mobile robot system to move up and down a single LRF along the yaw axis. During the up and down motion, the 2D data are stacked and manipulated to build a 3D map. Often a single LRF data are mixed with Gaussian and impulse noises. The impulse noises are removed out by the hybrid median filter designed in this research. The 2D data which are improved by deleting the impulse noises are layered to build the 3D map. Removing impulse noises while preserving the boundary is a main advantages of the hybrid median filter which has been used widely to improve the quality of images. The effectiveness of this hybrid median filter for rejecting the impulse noises has been verified through the real experiments. The performance of the hybrid median filter is evaluated in terms of PSNR (Peak Signal to Noise Ratio) and the processing time.

• Journal of the Korean Institute of Telematics and Electronics S
• /
• v.35S no.10
• /
• pp.11-18
• /
• 1998

A binary line code, called MB46d, is designed by use of the BUDA(Binary Unit DSV and ASV) cell concept to retain the property of being runlength limited, DC tree, and with a power spectral null at the Nyquist frequency. This new code is a stateless line code with a simple encoding and a decoding rule and enables efficient error monitoring. The power spectrum and the eye pattern of the new line code are simulated for a minimum-bandwidth digital transmission system where the sinc function is used as a basic pulse. The obtained power null at the Nyquist frequency is wide enough to enable easy band-limiting as well as secure insertion of a clock pilot where necessary. The eye is also substantially wide to tolerate a fair amount of timing jitter in the receiver.

• Journal of the Korean Institute of Telematics and Electronics S
• /
• v.35S no.10
• /
• pp.175-185
• /
• 1998

In this paper, we studied some design methodologies of typical FIR filters based on the peak-error constrained least square criterion which was first introducedd by Adams in 1991. This method is a mixed type of the classical least squared error method(LSM) and the so-called min-max error method (MMM). And by considering both the least squared error as well as the maximum error, the solution, i.e. the impulse response of the filter, can be found only when the restrictions on maximum gain, transition bandwidth, and the squared error are satisfied simultaneously under some trade-off conditions. We used the multiple exchange algorithms for optimization procedure and applied the design methodology to the cases of the multiband filter, the differentiator, and the Hilbert transformer by taking the balance of two design criteria into account. The results show that the peak-error constrained least weighted square error design method(PLEM) is superior in performance to the existing LSM and MMM from both the squared error and the maximum error standpoints. And it is verified that PLEM can be applied to not only the case of simple low pass filter, but also to various types of FIR filters.

• Journal of the Institute of Electronics and Information Engineers
• /
• v.50 no.5
• /
• pp.231-241
• /
• 2013

In this paper, we propose an average approximation filter which can effectively remove the noises of the images. The noises include impulse noises, gaussian noises and mixed noises. The algorithm is as follows. First, as a step of noise detection, we find whether the difference between the pixel value and the average value is greater than the threshold value or not after getting the average value that removed the minimum and maximum values in the applied mask. If the pixel value is greater than the threshold value, the pixel value is processed as noise. If it is less than or equal to the threshold value, it is processed as non-noise. Next, as the noise reduction step, we output the approximate value in mask as the pixel value and the average value except the minimum and maximum values of the pixel including the noise. As the result of applying this average approximation filter to the mixed noise images, the approximation filter can reduce the noises effectively more than 0.4[dB] as compared with applying the median filter and the average filter, respectively.