• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.10C
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• pp.984-994
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• 2003

This paper proposes a novel blind adaptive algorithm for computing the weight vector of an antenna array system. The new technique utilizes a Generalized On-Off algorithm to obtain the weight vector maximizing the SINR(Signal to Interference plus Noise Ratio) of the received signal. It is observed that the proposed algorithm generates a suboptimal weight vector with a linear computational load(O(6N+8)). From the various simulations, it is confirmed that, when the signal environment becomes adverse, e.g., low Processing Gain, and/or wide angular spread. the proposed algorithm outperforms the conventional one in terms of the communication capacity by about 3 times. Applying the proposed algorithm to satellite tracking systems as well as IS2000 1X mobile communication system, we have found that both communication capacity and communication quality are significantly improved.

• Journal of the Korea Society of Computer and Information
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• v.19 no.2
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• pp.101-107
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• 2014

This study is proposed the novel PPG sensor device and the signal processing method to replace the acceleration sensor that is used to reject motion artifacts contained in photoplethysmography(PPG). The proposed method is to reject motion artifacts by an adaptive filter based on the estimated motion artifact by using a blue LED light. To evaluate the performance of the proposed method experimentally, We did design a novel sensor consisted of blue/red LEDs and photo-sensor and implemented, and then rejected the motion artifacts by using an adaptive filter and the implemented sensor. In the results of the experiments, it is shown that the proposed sensor device and signal processing can reconstruct the PPG signal despite the occurrence of motion artifacts, and also that the SNR was 4.5 times of moving average filter. According to the experimental results, the proposed method can be applied to design a low-cost device.

• Proceedings of the KWS Conference
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• 1996.10a
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• pp.139-141
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• 1996

Among various welding parameters, the welding current which is inversely proportional to the tip-to-workpiece distance in GMAW is an essential parameter to monitor the GMAW process of horizontal fillet joints. For the case of weld defect such as overlap in horizontal fillet welding, therefore, the signal processing for process monitoring or automatic seam tracking should be modified by considering the weld pool surface geometry including the corresponding weld defect. In other words, the adequate signal processing algorithm is indispensible to improve the performance of the arc sensor. However, arc sensor algorithm already developed usually focus on weld seam tracing but do not considering the weld qualities. In this paper, various experiments were carried out to investigate the tendencies of the weld defects when weaving motion is added, and the experimental method based on 2$^n$ factorial design was proposed for deriving the mathematical model between the leg length and the various welding conditions. Moreover, a signal processing method based on the artificial neural network(Adaptive Resonance Theory) was proposed far discriminating the current signal of sound weld beads from that of weld beads with overlap. Finally, the algorithm for weld seam tracking combined with the mathematical modeling and the signal processing method was carried out to track the weld line in conjunction with the improvement of the weld qualities. The reliability of the proposed algorithms were evaluated through various experiments, which showed that the proposed algorithms could be effectively used for arc welding automation.

• Journal of the korean Society of Automotive Engineers
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• v.14 no.5
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• pp.95-106
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• 1992

Active control of acoustic noise is an application area of adaptive digital signal processing with increasingly interest along the last year. This work studies the implementation of the multichannel LMS filter and the application of this algorithm for the reduction of the noise inside a vechicle cabin using a number of 'secondary sources' drived by adaptive filtering of a reference noise source. Firstly, we propose the use of an adaptive method for the time-varient optimal convergence factor. Secondly, we propose the use of adaptive delayed inverse model to estimate the elastic-acoustic transfer function presented in vechicle cabin. The original, primary source is often periodic, with a known fundamental frequency. A suitably filtered reference signal can thus be used to drive the secondary sources. An algorithm is presented for adapting the coefficients of an FIR filter feeding such a secondary source in such a way as to minimize the output of a suitably placed microphone. In this algorithm, the coefficients of adaptive filter driving an array of secondary sources can be adapted to minimize the sum of the squares of the outputs of a number of error microphones. The multichannel LMS algorithm displays that such an algorithm is considered suitable to used for the global suppression of noise in vehicle cabin.

• Journal of Sensor Science and Technology
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• v.19 no.1
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• pp.43-51
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• 2010

In this study, wearable ECG measurement system was implemented for health monitoring during daily life. A wearable belt-type ECG electrode worn around the chest by measuring the real-time ECG is produced in order to minimize the inconvenience in wearing. The measured ECG signal is transmitted via an ultra low power consumption wireless data communications unit to personal computer using Zigbee-compatible wireless sensor node. The ECG monitoring program is developed at end user which is personal computer. The measured ECG contains many noises mainly due to motion artifacts. For ECG signal processing, adaptive filtering process is proposed which can reduce motion artifacts efficiently and accurately than digital filter. The experimental results show that a reliable performance with high quality ECG signal can be achieved using this wearable ECG monitoring system.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.1428-1431
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• 2002

It is general opinion that the future mobile multimedia networks will use different standards and a prospective solution to this problem will be software defined radio (SDR) techniques. SDR provides the flexibility to support multiple air interfaces and signal processing functions at the same time. Especially, digital signal processors and FPGAs are widely used for implementation of these adaptive and flexible functions of a baseband modem for SDR applications. Also, it is known that the modulation schemes of OCQPSK (Orthogonal Complex QPSK) and HPSK (Hybrid PSK) are used for IMT-2000 services of cdma2000 and WCDMA, respectively. Thus, in this paper, we design and implement an OCQPSK / HPSK modem using a DSP chip of Texas Instrument's TMS320C6701. One modulation scheme is operated by adaptive selection between the two schemes and 5 physical traffic channels differentiated by orthogonal codes are implemented in one DSP chip and each channel has 1Mbps data rates and 8Mcps chip rates.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.2
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• pp.211-224
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• 1994

In this paper, a multi-level adaptive intraframe method has been proposed to separate the luminance and chominance components in NTSC composite signal. The control signals are generated by detecting the vertical correlation and transition in the horizontal and diagonal directions. The chrominance component is adaptively processed through vertical and horizontal filters according to the control signals and the luminance component is processed by subtracting the chrominance component from the composite video signal. The several filters have been used at the sampling rate of four times the color subcarrier frequency and computer simulation and SVP(Serial Video Processing) system have been introduced to compare the performance of the conventional methods and that of proposed one.

• ETRI Journal
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• v.20 no.2
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• pp.133-148
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• 1998

The objective of this paper is to present an adaptive algorithm for computing the weight vector which provides a beam pattern having its maximum gain along the direction of the mobile target signal source in the presence of interfering signals within a cell. The conjugate gradient method (CGM) is modified in such a way that the suboptimal weight vector is produced with the computational load of O(16N), which has been found to be small enough for the real-time processing of signals in most land mobile communications with the digital signal processor (DSP) off the shelf, where N denotes the number of antenna elements of the array. The adaptive procedure proposed in this paper is applied to code division multiple access (CDMA) mobile communication system to show its excellent performance in terms of signal to interference plus noise ratio (SINR), bit error rate (BER), and capacity, which are enhanced by about 7 dB, ${\frac{1}{100}}$ times, and 7 times, respectively, when the number of antenna elements is 6 and the processing gain is 20 dB.

• Journal of KSNVE
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• v.6 no.2
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• pp.205-214
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• 1996

Active noise control(ANC) requires the full capability of a modern digital signal processing module. This paper describes the digital signal processing unit which is designed for ANC of noise fields in passenger car. System hardware is designed to allow software controlled versatility as well as fully qutomatic operation. The developed system is provided with the ability to be self-operated except the case of upload/download of data and program between the personal computer and the system memory. Experimental results are presented to demonstrate ANC performance of noise fields in lightly damped enclosure and passenger car.

• Proceedings of the KSPS conference
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• 2002.11a
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• pp.179-182
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• 2002

The development of noise robust speech processing systems is becoming increasingly important as speech technology is currently widely applied in real world applications. Recently, to resolve such a noise problem, adaptive noise canceller(ANC) is frequently used, which is based upon adaptive filters. The adaptive recursive filters perform better than adaptive non-recursive filters due to the added poles, but the stability may be severely threatened. But these problems of adaptive recursive filters was solved by ACHARF algorithm. This paper presents a method which combines speaker verification system with ANC(Adaptive Noise Canceller) using the ACHARF algorithm. In the front-end stage, ANC is adopted to suppress the additive noise imposed on the speech signal. The results show that the performance of speaker verification system becomes better than before.