• Journal of Korea Society of Industrial Information Systems
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• v.5 no.2
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• pp.32-38
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• 2000

One of the most popular algorithm in adaptive signal processing is the least mean square(LMS) algorithm. The majority of these papers examine the LMS algorithm with a constant step size. The choice of the step size reflects a tradeoff between misadjustment and the speed of adaptation. Subsequent works have discussed the issue of optimization of the step size or methods of varying the step size to improve performance. However there is as yet no detailed analysis of a variable step size algorithm that is capable of giving both the adaptation speed and the convergence. In this paper we propose a new variable step size algorithm where the step size adjustment is controlled by the gradient of error square. The proposed algorithm is performed in the Walsh-Hadamard domain in real-valued orthogonal transform because of fast convergence. The simulation results using the new algorithm for noise canceller system is described. They are compared to the results obtained by other algorithms. It is shown that the proposed algorithm produces good results compared with conventional algorithms.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.4
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• pp.669-678
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• 1994

This paper presents a new data predistortion technique with memory to compensate for the nonlinearities of high-power amplifiers (HPA`s) in digital radio systems employing QAM signal formats. In contrast with the conventional data predistortion technique which is designed to reduce nonlinearity of memoryless HPA`s, the proposed technique in this paper compensates not only for nonlinear warping of the signal constellation but also for clustering of the signal points caused by transmitter pulse sharping filter with memory. A practical implementation method which can reduce the size of memory at the predistortion stage is described by utilizing symmetry of QAM constellation format and Modulo-4 operation.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.10a
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• pp.471-473
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• 2009

In this paper, we propose a double talk and noise robust acoustic echo canceler for hands-free telephony applications. The proposed system includes a double-talk detection method that detects the double-talk durations, which uses covariance between microphone input signa and estimated microphone input signal. And proposed adaptive algorithm for estimating acoustic echo path, uses normalized auto-covariance matrix of input signal with multiplication of residual error power and projection order of AP(affine projeciton) algorithm. It is confirmed that the proposed algorithm shows better performance from acoustic interference cancellation (AIC) viewpoint in double talk and noisy environments.

• The Journal of the Korea institute of electronic communication sciences
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• v.13 no.5
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• pp.979-986
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• 2018

This paper deals with acoustic echo cancellation using blind signal separation method. This method does not degrade the echo cancellation performance even during double-talk. In the closed echo environment, the mixing model of acoustic signals is multi-channel, so the convolutive blind signal separation method is applied and the mixing coefficients are calculated by using the feedback model without directly calculating the separation coefficients for signal separation. The coefficient update is performed by iterative calculations based on the second-order statistical properties, thus estimates the near-end speech. A number of simulations have been performed to verify the performance of the proposed blind signal separation method. The simulation results show that the acoustic echo canceller using this method operates safely regardless of the presence of double-talk, and the PESQ is improved by 0.6 point compared with the general adaptive FIR filter structure.

• The Journal of the Korea institute of electronic communication sciences
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• v.17 no.3
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• pp.409-416
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• 2022

This paper relates to the blind noise separation method of time-delayed convolutive mixed signals. Since the mixed model of acoustic signals in a closed space is multi-channel, a convolutive blind signal separation method is applied and time-delayed data samples of the two microphone input signals is used. For signal separation, the mixing coefficient is calculated using an inverse model rather than directly calculating the separation coefficient, and the coefficient update is performed by repeated calculations based on secondary statistical properties to estimate the speech signal. Many simulations were performed to verify the performance of the proposed blind signal separation. As a result of the simulation, noise separation using this method operates safely regardless of convolutive mixing, and PESQ is improved by 0.3 points compared to the general adaptive FIR filter structure.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.18 no.1
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• pp.79-84
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• 2018

This paper relates with the performance improvement of SE-MMA (Signed Error-Multiple Modulus Algorithm) adaptive equalization algorithm that is used for the reduction of the intersymbol interference due to the distortion which occurs in the communication channel for the transmission of 16-QAM nonconstant modulus signal.. In the conventional MMA, the fixed modulus value that is second order statistics of transmitting signal were used, and the SE-MMA was introduced in order to the simplification of the algorithm's arithmetic operation. The SE-MMA have a fast convergence speed than MMA, but it has a problem of degradation of equalization performance in the steady state due to the arithmetic simplification. In this paper, we propose the new algorithm AV-SE-MMA (Adaptively Varying-SE-MMA) that uses the adaptive varying modulus in order to obtain the error signal for updating the adaptive equalizer coefficient, and its equalization performance were confirmed by simulation. In this paper, the performance of SE-MMA and proposed algorithm were compared, and the equalizer output signal constellation, residual isi, MSE and SER in order to confirm the robustness of noise were used as performace index. As a result of performance comparison, the AV-SE-MMA has better performance in output signal constellation, residual isi and MD compared to the SE-MMA, but it was confirmed that the AV-SE-MMA has similar in the SER performance that means the robustness to the noise.

• Journal of the Korean Institute of Telematics and Electronics
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• v.26 no.1
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• pp.129-139
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• 1989

During a signal is being transmitted, an interference signal can be introduced through an unknown channel. In these cases, an adaptive system, so called adaptive noise canceller, can restore the original signal from the corrupted signal by first identifying the unknown interference channel on the minimum mean square error criteron, and then by cancelling the interference signal using the identified interference channel. Whereas this method is quite effective when the a priori knowledges about the characteristics of the interference signal and of the intrference channel are unknown or time-varyng, but has a drawback that the presence of the original signal has a severe effect on the optimum value of the interference channel to be identified on the miniumum mean square eror criterion In this paper an adaptive noise canceller and its algorithms are introduced that can restore the original signal more accurately especially when the correlatedness between the original signal and the interference signal is small.

• Journal of the Institute of Convergence Signal Processing
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• v.9 no.2
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• pp.104-111
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• 2008

The ECG is biomedical electrical signal occurring on the surface of the body due to the contraction and relaxation of the heart. This signal represents an extremely important measure for health monitoring, as it provides vital information about a patient's cardiac condition and general health. ECG signals are contaminated with high frequency noise such as power line interference, muscle artifact and low frequency nose such as motion artifact. But it is difficult to filter nose from ECG signal, and errors resulting from filtering can distort a ECG signal. The present study implemented a small-size and low-power ECG measurement system that can remove motion artifact for convenient health monitoring during daily life. The implemented ECG monitoring system consists of ECG amplifier, a low power microprocessor, bluetooth module and monitoring program. Amplifier was designed and implemented using low power instrumentation amplifier, and microprocessor was interfaced to the ECG amplifier to collect the data, process, store and feed to a transmitter. And bluetooth module used to wirelessly transmit and receive the vital sign data from the microprocessor to an PC at the receiving site. In order to evaluate the performance of the implemented system, we assessed motion artifact rejection performance in each situation with artificially set condition using adaptive filter.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.3A
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• pp.327-334
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• 2008

Conventional on-channel-repeaters (OCRs) have a crucial problem that the power of a re-transmitted signal is highly limited by a feedback signal due to antenna coupling. The power limitation problem in OCRs has been solved by incorporating a demodulation-type feedback canceller which eliminates unwanted feedback signals by estimating a feedback channel. In applying the demodulation-type feedback canceller to T-DMB repeaters, there is a troublesome problem of unfrequent known pilot symbols, resulting in poor convergence performance of channel estimation. To solve this problem and enhance the accuracy of estimation, we propose a partitioning method of the Phase Reference Symbol (PRS) transformed in time domain. Since filter coefficients are updated every one partitioned subgroup, the number of updates is increased by the number of partitioned subgroups and thus the convergence speed is enhanced. The improved performance of feedback-channel estimation is directly connected with the feedback-cancellation performance. Simulation result shows that the feedback canceller incorporating the proposed partitioning method has a good performance in terms of residual feedback power.

• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.49 no.11
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• pp.933-940
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• 2021

The space-borne SAR(Synthetic Aperture Radar) system radiates the microwave signal and receives the backscattered signal. The received signal is converted to digital at the Digital Receiver, which is implemented at the end of the SAR sensor receiving chain. The converted signal is formated after signal processing such as filtering and data compression. Two quantization are conducted in the Digital Receiver. One quantization is an analog to digital conversion at ADC(Analog-Digital Converter). Another quantization is the BAQ(Block Adaptive Quantization) for data compression. The quantization process is a conversion from a continuous or higher bit precision to a discrete or lower bit precision. As a result, a quantization noise is inevitably occurred. In this paper, the impact of two quantization processes are analyzed in a view of SNR degradation.