• Journal of the Korean Society for Precision Engineering
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• v.26 no.12
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• pp.146-153
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• 2009

One of the most important issues in the wearable healthcare sensors is to minimize the motion artifacts in the vital signals for continuous monitoring. This paper presents a reflected type photoplethysmograph (PPG) sensor for monitoring heart rates at the artery of the wrist. Active noise cancellation algorithm was applied to compensate the distorted signals by motions with Least Mean Square (LMS) adaptive filter algorithms, using acceleration signals from a MEMS accelerometer. Experiments with a watch type PPG sensor were performed to validate the proposed algorithm during typical daily motions such as walking and running. The developed sensor is suitable for ubiquitous healthcare system and monitoring vital arterial signals during surgery.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.53 no.6
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• pp.446-452
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• 2004

Under the condition of a common narrowband target signal and interference signals from several directions, the linearly constrained minimum variance (LCMV) method using the generalized sidelobe canceller (GSC) for adaptive beamforming has been exploited successfully However, in the case of wideband signals, the length of the adaptive filter must be extended. As a result, the complexity of the beamformer increases, which makes real-time implementation difficult. In this paper, we improve the convergence characteristics of the adaptive filter using the transform domain normalized least mean square (NLMS) approach based on the subband GSC structure without the increase of complexity. Besides, the M-MAX algorithm, which is one of various selective coefficient updating methods, is employed in order to remarkably reduce the computational cost without decreasing the convergence quality. With the combination of these methods, we propose a computationally efficient wideband adaptive beamformer and verify its efficiency through a series of simulations.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.44 no.7 s.361
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• pp.35-40
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• 2007

As the mobile communication service is widely used and the demand for wireless repeaters is rapidly increasing because of the easiness of extending service areas. But a wireless repeater has a problem that the output of the transmit antenna is partially fed back to the receive antenna, which results in feedback interference. In this paper, we propose a new varable step-size LMS algorithm which utilizes correlation between reference and error signals to adjust the step sizes, for cancelling the feedback interference signals in the WCDMA repeater under time-varying multi-path channels. The proposed algorithm was evaluated through computer simualation by being applied to the feedback canceling filter of the WCDMA repeater. The simulation results demonstrated that the proposed one is superior to the conventional ones in terms of the cancelation perormance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.2A
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• pp.210-217
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• 2008

In this paper, the performance of equalization-based on-channel repeater for terrestrial DMB is analyzed. A primary concern in on-channel repeater is the performance degradation due to the echo and one of key component for on-channel repeater is the echo canceller, which usually employs LMS algorithm utilizing the repeater output as a reference for echo channel estimation and compensation. One problem using LMS algorithm is the tracking capability and there necessarily exists residual echo that has not been cancelled. To effectivelyremove the residual echo, we consider an equalization based on-channel repeater where the echo-cancellor is followed by an equalizer that performs channel estimation using pilot symbol and the channel inversion utilizing homomorphic decomposition. According to the simulation result, the performance degradation caused by the residual echo can be considerably alleviated by using the equalizer following the echo-canceller.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.1
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• pp.178-203
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• 2018

In this paper, authors have been evaluated the Frame Error Rate (FER) performance of IEEE 802.11 a/g/p standard 5 GHz frequency band WLAN over Rayleigh and Rician distributed fading channels in presence of Additive White Gaussian Noise (AWGN). Orthogonal Frequency Division Multiplexing (OFDM) based transceiver is implemented by using real-time signal processing frameworks (IEEE 802.11 Blocks) in GNU Radio Companion (GRC) and Ettus USRP N200 is used to process the symbol over the wireless radio channel. The FER is calculated for each sub-carrier conventional modulation schemes used by OFDM such as BPSK, QPSK, 16, 64-QAM with different punctuated coding rates. More precise SNR is computed by modifying the SNR calculation process of YANS and NIST error rate model to estimate more accurate FER. Here, real-time signal constellations, OFDM signal spectrums etc. are also observed to find the effect of multipath propagation of signals through flat and frequency selective fading channels. To reduce the error rate due to the multipath fading effect and Doppler shifting, channel estimation (CE) and equalization techniques such as Least Square (LS) and training based adaptive Least Mean Square (LMS) algorithm are applied in the receiver. The simulation work is practically verified at GRC by turning into a pair of Software Define Radio (SDR) as a simultaneous transceiver.

• The Journal of the Acoustical Society of Korea
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• v.34 no.2
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• pp.130-137
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• 2015

In this paper, we propose a new Modified Filtered-x Affine Projection Sign Algorithm(MFxAPSA) to improve the convergence speed of the conventional MFxAPSA which has been proposed for active control of impulsive noise. Under the impulsive noise environment, the adaptive algorithms based on the second order moment such as the Filtered-x Least Mean Square(FxLMS) show slow convergence speed or diverge because the noise source tends to have infinite variance. The MFxAPSA is the algorithm derived by applying the Affine Projection Sign Algorithm(APSA) to active noise control. The APSA has an advantage that it does not need the calculation for the inverse matrix, so it may be suitable for the active noise control that requires low computational burden. The proposed MFxAPSA also has APSA's advantage and furthermore, better performance than the conventional MFxAPSA. We carried out a performance comparison of the proposed MFxAPSA with the conventional MFxAPSA. It is shown that the proposed MFxAPSA has the faster convergence speed than the conventional MFxAPSA.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.38 no.2
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• pp.227-236
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• 2001

This paper proposes a frequency-domain partially adaptive algorithm, called a censoring algorithm, to reduce the computational complexity of the frequency domain adaptive array. The proposed censoring algorithm determines the existence of interferences in the frequency-domain at each frequency bin using a constant false alarm rate (CFAR) processor. The censoring algorithm adapts only those parts of the weights that correspond to the frequency bins expected to contain interferences. The censoring algorithm is also expanded to overcome the signal cancellation phenomenon caused by smart jammers. Accordingly, a censoring spatial smoothing, which combines the censoring algorithm with spatial smoothing, is proposed. Simulation results show that the proposed algorithms are effective in removing interferences with only part of the computational complexity of conventional algorithms yet with the same level of performance.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2008.11a
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• pp.497-502
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• 2008

In this paper, we studied on an attenuation effect of automobile exhaust noise according to the direction of secondary sound source in duct ANC system. Automobile exhaust noise was recorded at 800rpm. 3500rpm and 5000rpm of a diesel engine. Directions of loudspeaker(second sound source) can be exchanged to $30^{\circ}$, $90^{\circ}$ and $150^{\circ}$ against the primary noise flow by acrylic ducts to be made for experimentation. DSP board with TMS320C6416 chip of Texas Instrument Co used to control adaptive ANC system. This ANC system is based on the single-channel FxLMS algorithm. In experiment result, when the loud speaker direction was $150^{\circ}$, the attenuation effect showed largely. In case of $90^{\circ}$ duct, the noise was a little increased. In case of $30^{\circ}$ duct, the noise was a little increased or decreased according to the frequency range and the sound pressure(dB) of exhaust noise to comply with engine rpm.

• Progress in Superconductivity
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• v.4 no.1
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• pp.53-58
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• 2002

Noise from electronic instrumentation is invariably present in biomedical signals, although the art of instrumentation design is such that this noise source may be negligible. And sometimes signals of interest are contaminated or degraded by signals of similar type from another source. Biomedical signals are omni-presently contaminated by these background noises that span nearly all frequency bandwidths. In the magneto-cardiogram (MCG), several digital filters have been designed for the elimination of the power-line interference, broadband white noise, surrounding magnetic noise, and baseline wondering. In addition to the introduced FIR filter, notch, adaptive filter using the least mean square (LMS) algorithm, and recurrent neural network (RNN) filter, a new filtering method for effective noise canceling in MCG signals is proposed in this paper, which is realized by the wavelet packets. The experimental results show that the proposed filter using wavelet packet performs efficiently with respect to noise rejection. To verify this, two characteristics were analyzed and compared with LMS adaptive filter, SNR of filtered signal and attractor pattern using the nonlinear dynamics.

• The Journal of the Acoustical Society of Korea
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• v.37 no.4
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• pp.206-214
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• 2018

This paper proposes an impact point estimation algorithm of the rifle using microphone sensors. The proposed algorithm resolves the time synchronization problem by expanding the existing ToA (Time of Arrival) method to TDoA (Time Difference of Arrival) method and verifies the performance of the algorithm through the actual shooting experiments. By comparing analysis of the actual and the estimated impact points by the algorithm, it is confirmed that the proposed algorithm has excellent performance by estimating the impact point accurately within the tolerance range.