• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.285-288
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• 2008

In ubiquitous computing environment the biological signal ECG (Electrocardiogram signal) is usually recorded with noise components. Adaptive interference (or noise) canceller do adaptive filtering of the noise reference input to maximally match and subtract out noise or interference from the primary (signal plus noise) input thereby adaptively eliminate unwanted interference from the ECG signal. Measured Stress ECG (or exercise ECG signal) signal have three major noisy component like baseline wander noise, motion artifact noise and EMG (Electro-mayo-cardiogram) noise. These noises are not only distorted signal but also root of incorrect diagnosis while ECG data are analyzed. Motion artifact and EMG noises behave like wide band spectrum signals, and they considerably do overlapping with the ECG spectrum. Here the multidimensional adaptive method used for filtering which is more effective to improve signal to noise ratio.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.18 no.7
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• pp.839-846
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• 2007

In this paper, the first attempt to design a novel structure of dual-band feedforward linear power amplifier(FFW LPA) was presented. Up to now, primary technical difficulty has been the extension of the conventional signal canceller to the dual-band operation. Therefore, we propose the design technique of the dual-band equal group delayed carrier canceller, the dual-band equal group delayed intermodulation distortion(IMD) canceller and the dual-band FFW LPA. The operation frequency bands of the implemented dual-band FFW LPA are digital cellular($f_0=880$ MHz) and IMT-2000($f_0=2.14$ GHz) band, which are separated about 1.26 GHz. With the high power amplifier of 120 W PEP for commercial base-station application, IMD cancellation loop shows 20.45 dB and 25.04 dB loop suppression at each band of operation for 100 MHz. From the adjacent channel leakage ratio(ACLR) measurement with CDMA IS-95A 4FA and WCDMA 4FA signal, we obtained 16.52 dB improvement at the average output power of 41.5 dBm for digital cellular band, and 18.59 dB improvement at the average output power of 40 dBm for IMT-2000 band simultaneously.

• The Journal of the Acoustical Society of Korea
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• v.34 no.5
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• pp.377-384
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• 2015

An active noise control (ANC) algorithm for noise canceling headphone is proposed. In this study, the feedback ANC operated with the filtered-x least mean square algorithm (FxLMS) algorithm is used to attenuate the undesired noise. Also an adaptive residual music canceller (RMC) is proposed for enhancing the accuracy of the reference signal of the feedback ANC. Simulation results show that a high quality of music sound can be consistently achieved in a time-varying secondary path situation.

• The Journal of the Acoustical Society of Korea
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• v.21 no.3
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• pp.264-273
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• 2002

Adaptive filters cannot exactly estimate the echo path of the receiving room because of the cross-correlation of stereo signals. In this paper, a new pre-processing method reducing the cross-correlation without degradation of stereophony is proposed to enhance the performance of stereophonic acoustic echo canceller. To reduce the cross-correlation, absolutes of two orthogonal signals derived from each channel signals are added to original channel signals. Assuming that the power of each channel signal is larger than that of the cross-correlation, the computation of pre-processing can be reduced. As results of simulations, it is shown that the performance of stereo acoustic echo canceller with the proposed pre-processing method is better than that of conventional ones.

• Journal of Korea Multimedia Society
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• v.19 no.7
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• pp.1127-1136
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• 2016

Speech enhancement algorithm plays an important role in numerous speech signal processing applications. Over the last few decades, many algorithms have been studied for speech enhancement. The algorithms are based on spectral subtraction, Wiener filter, and subspace method etc. They have good performance of speech enhancement, but the performance can be deteriorated in specific noises or low SNR environment. In this paper, a new speech enhancement algorithms are proposed based on adaptive noise canceller. And the proposed algorithm improved performance of adaptive noise cancelling using 2-D binary mask. From objective experimental index, it is confirmed that the proposed algorithm is useful and has better performance than recently proposed speech enhancement algorithms.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.2C
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• pp.222-228
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• 2004

This paper describes a new adaptive algorithm which can reduce the required computation quantities in the adaptive filter. The proposed adaptive algorithm uses only the signs of the normalized input signal rather than the input signals when coefficients of the filter are adapted. By doing so, there is no need for the multiplications and divisions which are mostly responsible for the computation quantities. To analyze the convergence characteristics of the proposed algorithm, the condition and speed of the convergence are derived mathematically. Also, we simulate an echo canceller adopting this algorithm and compare the performances of convergence for this algorithm with the ones for the other algorithm. As the results of simulations, it is proved that the echo canceller adopting this algorithm shows almost the same performances of convergence as the echo canceller adopting the SIA algorithm.

• The Journal of the Korea institute of electronic communication sciences
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• v.4 no.1
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• pp.66-69
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• 2009

An adaptive echo canceller is necessary for an application such as a speakerphone, 3G image telephony and VoIP service system. These echo cancellers need to have many taps for filtering echo signals. Many taps cause computation data to increase and convergence speed to be low. To overcome these problems, An adaptive echo canceller with the advanced convergence speed is proposed in this paper. To improve the speed, we divide an echo band into subbands and place a subband filter to be adaptive for each subband. Each subband filter recognizes the echo signal as subband echo signals. So, dynamic range of subband is small, the convergence speed is fast. Moreover, as the number of Tap and weight update are estimated in each subband, the implementation complex of a adaptive filter is low.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.1C
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• pp.54-60
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• 2005

In this Paper, an adaptive microphone array system with self-delay estimator is proposed. By showing that the adaptive blocking matrix (ABM) of the generalized sidelobe canceller (GSC) can estimate the relative time delay between each sensor, the proposed system utilizes the ABM not only for blocking target components in the blocked signal path, but also for estimating the relative time delay. Therefore, the proposed system requires only the GSC structure while maintaining the system performance similar to the conventional system using an additional time delay estimator as a preprocessor. Simulation results show that the performance of the proposed system is identical to the conventional system that uses an additional time delay estimation module.

• ETRI Journal
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• v.33 no.5
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• pp.670-678
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• 2011

This paper investigates the design and performance of a digital on-channel repeater (DOCR) for use in Advanced Television Systems Committee (ATSC) digital television (DTV) broadcasting. The main drawback of a DOCR is the echo interference caused by coupling between transmitter and receiver antennas, which induces system instability and performance degradation. In order to overcome this problem, an echo canceller based on the adaptive echo channel estimation (ECE) technique has been researched and applied for a DOCR. However, in the case of ATSC, the pilot signal, which is used for carrier synchronization, may cause a DC offset error and reduce the isolation performance of the echo canceller for a DOCR in an ATSC network. Moreover, since the multipath fading effect of a radio channel usually occurs in a real environment, it should be minimized to improve the overall performance of a DOCR. Therefore, due to the limited isolation performance of echo canceller and the multipath fading effect, an interference cancellation system (ICS) is proposed for a DOCR in an ATSC network. The performance of the proposed DOCR with an ICS is evaluated by software simulation and hardware test results.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.12A
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• pp.958-963
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• 2003

Increasing throughput using high order data modulations in the WCDMA downlink requires the elimination of multipath interferences. For this reason, Kenichi Higuchi proposed a multipath interference canceller in WCDMA downlink. However, this paper considers the downlink signal models that only contains code channels for which code information is known to the receiver. WCDMA systems contain code channels for other active users whose code information is unknown to the receiver and these code channels degrade the performance of the multipath interference canceller. Thus, this paper proposes a multipath interference mitigation scheme which removes multipath interference induced by all the code channels within a cell. Simulation results show that the proposed scheme outperforms both the RAKE receiver and the multipath interference canceller in the WCDMA downlink.