• Proceedings of the Korean Society of Computer Information Conference
• /
• 2021.07a
• /
• pp.201-202
• /
• 2021

본 논문에서는 다중 엑세스 포인트 환경에서 전송전력, 수신필터, 엑세스 포인트 선정 최적화 알고리즘을 제안한다. 최종목적은 신호대간섭비를 유지하면서, 전송전력의 총합을 최소화하는 것이다. 증명을 통해서 제안하는 알고리즘은 최소전력에 수렴함을 보인다. 제안하는 알고리즘이 기존에 제안되었던 두 개의 알고리즘인 1)전송전력과 최소제곱평균오차(MMSE) 수신필터 최적화 알고리즘, 2) 전송전력 최적화 알고리즘보다 전송전력 소모량에서 성능이 우수함을 실험을 통해서 확인하였다.

• Journal of Satellite, Information and Communications
• /
• v.10 no.2
• /
• pp.19-23
• /
• 2015

In this paper, we present the return link ACM method to improve the link availability and system throughput for satellite communication service. Also, we describe the optimization of an algorithm for channel prediction using the LMS (Least Mean Square) adaptive filter and the MODCOD (Modulation & Code rate) decision. The simulation results show that the optimized filter taps and step-size of adaptive filter are 2 and 0.00026, respectively. And also confirms the required SNR margin for minimization of MODCOD decision error is 0.3dB.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.33 no.5C
• /
• pp.403-412
• /
• 2008

This paper presents the methods which obtain the solution of Wiener-Hopf equation by LMS algorithm and get the coefficient of TDL filter in lattice filter directly. For this result, we apply an orthogonal input signal generated by lattice filter into an equivalent Wiener-Hopf equation and shows the scheme that can obtain the solution by using the MMSE algorithm. Conventionally, the method like aforementioned scheme can get an error and regression coefficient recursively. However, in this paper, we can obtain an error and the coefficients of TDL filter recursively. And, we make an theoretical analysis on the convergence characteristics of the proposed algorithm. Then we can see that the result is similar to conventional analysis. Also, by computer simulation, we can make sure that the proposed algorithm has an excellent performance.

• Journal of Korea Multimedia Society
• /
• v.9 no.8
• /
• pp.1000-1009
• /
• 2006

We propose an adaptive linear interpolation locating sub-pixels. We utilize a pixel-based parameter in the conventional linear interpolation. To optimally obtain the parameter, we propose a generic interpolation structure including a low pass filter and minimum mean square error. We also propose a simple version of the generic interpolation method, which obtain a closed-form solution. Simulation results show that the proposed method is superior to the state-of-the-art methods such as warped distance linear interpolation and shifted linear interpolation, as well as the conventional method such as the linear interpolation and the cubic convolution interpolation in terms of the subjective and objective image quality.

• ETRI Journal
• /
• v.31 no.2
• /
• pp.162-172
• /
• 2009

A linear-prediction-based blind equalization algorithm for single-input single-output (SISO) finite impulse response/infinite impulse response (FIR/IIR) channels is proposed. The new algorithm is based on second-order statistics, and it does not require channel order estimation. By oversampling the channel output, the SISO channel model is converted to a special single-input multiple-output (SIMO) model. Two forward linear predictors with consecutive prediction delays are applied to the subchannel outputs of the SIMO model. It is demonstrated that the partial parameters of the SIMO model can be estimated from the difference between the prediction errors when the length of the predictors is sufficiently large. The sufficient filter length for achieving the optimal prediction is also derived. Based on the estimated parameters, both batch and adaptive minimum-mean-square-error equalizers are developed. The performance of the proposed equalizers is evaluated by computer simulations and compared with existing algorithms.

• Proceedings of the KSPS conference
• /
• 2005.11a
• /
• pp.147-150
• /
• 2005

The performance of speech recognition in car environment is severely degraded when there is music or news coming from a radio or a CD player. Since reference signals are available from the audio unit in the car, it is possible to remove them with an adaptive filter. In this paper, we present experimental results of speech recognition in car environment using the echo canceller. For this, we generate test speech signals by adding music or news to the car noisy speech from Aurora2 DB. The HTK-based continuous HMT system is constructed for a recognition system. In addition, the MMSE-STSA method is used to the output of the echo canceller to remove the residual noise more.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.30 no.9A
• /
• pp.830-837
• /
• 2005

ISI is caused by delay spread in the multipath channel environment. There are two kinds of channel equalizer: Linear and Non-Linear type according to the structures. In this paper, we propose an improved adaptive linear equalizer to mitigate ISI. The proposed adaptive equalizer is constructed by using asymmetrical Dsmvenu filter based on USE sub-optimal receiver. Asymmetrical structure of the transversal filter is realized by moving the main tap position from center to side. If this structure is used, we can divide ISI to precusor and postcusor. As a result the proposed equalizer has a larger extended compensation range than conventional adaptive linear equalizer. In computer simulation, we compare the bit error rate performance of the proposed linear equalizer with the conventional one on the S-V channel which is modeled for WB systems.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.6 no.5
• /
• pp.1316-1332
• /
• 2012

Formation flying is an important technology that enables high cost-effective organization of outer space aircrafts. The ad-hoc wireless network based on direct-sequence ultra-wideband (DS-UWB) techniques is seen as an effective means of establishing wireless communication links between aircrafts. In this paper, based on the theory of matched filter and error bits correction, a hybrid detection algorithm is proposed for realizing multiuser detection (MUD) when the DS-UWB technique is used in the ad-hoc wireless network. The matched filter is used to generate a candidate code set which may contain several error bits. The error bits are then recognized and corrected by an novel error-bit corrector, which consists of two steps: code mapping and clustering. In the former step, based on the modified optimum MUD decision function, a novel mapping function is presented that maps the output candidate codes into a feature space for differentiating the right and wrong codes. In the latter step, the codes are clustered into the right and wrong sets by using the K-means clustering approach. Additionally, in order to prevent some right codes being wrongly classified, a sign judgment method is proposed that reduces the bit error rate (BER) of the system. Compared with the traditional detection approaches, e.g., matched filter, minimum mean square error (MMSE) and decorrelation receiver (DEC), the proposed algorithm can considerably improve the BER performance of the system because of its high probability of recognizing wrong codes. Simulation results show that the proposed algorithm can almost achieve the BER performance of the optimum MUD (OMD). Furthermore, compared with OMD, the proposed algorithm has lower computational complexity, and its BER performance is less sensitive to the number of users.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.4 no.3
• /
• pp.661-669
• /
• 2000

DS-CDMA downlink distinguishes actual user by orthogonal spreading codes ,but its orthogonality may be lost by the multiple access interference(MAI) caused by the multipath channel. Therefore in this paper, we proposed the single-user receiver, which use linear channel equalizer to eliminate the interference due to multipath channel and to recover orthogonality and then use code-matched filter to detect transmitted data. Unlike existing research, which mainly assumed ideal channel information, we use pilot channel assisted methods that is a kind of transmission of a parallel reference method to estimate the channel coefficients. Especially we use guard symbols which are inserted periodically to estimate channel coefficients exactly without interference from user signal. The results show that we accepted an approximately ideal channel information and achieved excellent performance improvement using proposed receiver compared with the conventional receiver especially user populations are high.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.24 no.12B
• /
• pp.2369-2381
• /
• 1999

In this paper, we propose an iterative algorithm for reducing the blocking artifact in block transform-coded images by using a wavelet transform. In the proposed method, an image is considered as a set of one-dimensional horizontal and vertical signals and one-dimensional wavelet transform is utilized in which the mother wavelet is the first order derivative of a Gaussian like function. The blocking artifact is reduced by removing the blocking component, that causes the variance at the block boundary position in the first scale wavelet domain to be abnormally higher than those at the other positions, using a minimum mean square error (MMSE) filter in the wavelet domain. This filter minimizes the MSE between the ideal blocking component-free signal and the restored signal in the neighborhood of block boundaries in the wavelet domain. It also uses local variance in the wavelet domain for pixel adaptive processing. The filtering and the projection onto a convex set of quantization constraint are iteratively performed in alternating fashion. Experimental results show that the proposed method yields not only a PSNR improvement of about 0.56-1.07 dB, but also subjective quality nearly free of the blocking artifact and edge blur.